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<title>Re: How to do an SPA-3000 setup like mine... in VOIP Tech Chat</title>
<link>http://www.dslreports.com/forum/r14182197</link>
<description></description>
<language>en</language>
<pubDate>Sun, 29 Nov 2009 00:07:15 EDT</pubDate>
<lastBuildDate>Sun, 29 Nov 2009 00:07:15 EDT</lastBuildDate>

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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,17588462</link>
<description><![CDATA[<A HREF="/useremail/u/1428453"><b>nor500</b></A> : I am still having blocked inbound packets problem with sipura spa3000 and linksys wireless router result one way audio. Eventhough I forwarded the 5060-5061 udp ports and 16384-16482 to my sipura and I also enabled Nat mapping enabled and Nat keep alive.<br><br>If you can, please help me to solve this problem,<br><br>Best regards,<br><br>Nor]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,17588462</guid>
<pubDate>Sat, 06 Jan 2007 02:54:06 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,17563251</link>
<description><![CDATA[<A HREF="/useremail/u/1384473"><b>joneill</b></A> : I have a linksys PAP2T Unlocked(Firmware Version: 3.1.9(LSc)) I need to forward line two to ring on line one for incoming calls can anyone advise how I can do this by adding something to the dialplan.<br>I am in the uk.<br><br>Thankyou]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,17563251</guid>
<pubDate>Tue, 02 Jan 2007 07:40:00 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,17093677</link>
<description><![CDATA[<A HREF="/useremail/u/1201620"><b>navjun</b></A> : PLease help!!!!<br><br>I have a pap2 that I have put a sipura Firmware on ( so that I can use both lines using one phone) and it has been working perfectly for a few months now, I have set up line one with voicestic so I can get incoming calls and calls and since I live overseas and Voicestics rates are cheaper than the local phone companies I have set the Dial plan that I can dial a local numbers in my place of residence that prefix here is "0" so any call initiated with zero or just a local number will automaticly get the area code of my place , and line 2 is setup with sipphone.com and I use it to call US by pressing # before every call since siphone has a cheaper US rate than Voicestick.<br>NOW my questions is , is it possible by using dial plan make any call on line one that has a prefix of "1" go to line 2 (sipphone) so I wouldn't have to press # every time I'm calling home and also I can just revert back to pap2's original firmware since I only use line 2 for outgoing .]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,17093677</guid>
<pubDate>Sun, 15 Oct 2006 20:02:28 EDT</pubDate>
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<title>PAP2 to spa3000 Http Digest</title>
<link>http://www.dslreports.com/forum/remark,17058661</link>
<description><![CDATA[<A HREF="/useremail/u/1282335"><b>havarian</b></A> : Has any one manage to use http digest between pap2 & spa3000.<br>I have pap2 in one country and spa3000 in another country.<br>I want to enable one stage dialing using "http auth" from the pap2 to the spa3000 pap2.<br>I managed to do that using pin & non authentication. but it always fail for "http auth"<br><br>in my speed dial I set<br>123@sipproxy-ip:port;usr="user id";pwd="user password"]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,17058661</guid>
<pubDate>Tue, 10 Oct 2006 02:22:59 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,17006401</link>
<description><![CDATA[<A HREF="/useremail/u/1268357"><b>chicha36</b></A> : Hi DF.  I've got 5 yes/no values that don't agree with yours on the PSTN tab.  Should they? I don't know the reasons why the 5 are either set to yes or no.  They are: 1) Under NAT Settings: NAT Mapping Enable: Mine is yes, yours no. 2) SIP Settings: Refer-To Target Contact:  Mine is at yes; SIP Remote-Party-ID: Mine is at no. 3) Proxy and Registration: Register: Mine is at no; Use OB Proxy in Dialog: Mine is at yes.<br>---------<br>I'm using a Sipura 3102.  Thanks.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,17006401</guid>
<pubDate>Sat, 30 Sep 2006 23:55:50 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,16960116</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  chicha36 <A HREF="/useremail/u/1268357"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>I don't see your PSTN tab values, though!!!  [I'm going to print out those screenshots you posted.]   Can you post a screenshot of your PSTN tab?<br> </DIV>The reason the previous post didn't include the PSTN tab, was that I wasn't using the PSTN side of the adapter at the time (and I was describing how to setup the adapter like I was using it).  <br><br>However, since that time I have added enough PSTN access so I can use the "Line jack" for sending/receiving calls (from the "Line 1" side of the SPA-3000).  Since I have no real VoIP provider on the PSTN side of things, I need to put some "bogus" (made up) VoIP credentials in the PSTN side (I blanked out the "User ID" in the graphic, so just pick a "User ID" that makes sense to you), to allow the internal adapter VoIP call (from one side of the adapter to the other) to work properly.  And I also changed the PSTN "dial plan" (so that any digits could be "dialed" on the PSTN side), as well as restricting access to 127.0.0.1 (the "loopback" address, i.e. only allowing PSTN access from the "Line 1" side of the SPA-3000, not the general internet).<br><br>The resulting screen shots are included with this message.  Due to the length of the PSTN tab (and the resolution on my screen), this tab takes 3 screen shots all by itself...<div class="borderless"><TABLE WIDTH=95% align=center border=0 CELLPADDING=4"><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/16960116?c=1066556&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG class="apic" BORDER=0 TITLE="32510 bytes" WIDTH=600 HEIGHT=757 SRC="/r0/download/1066556.thumb600~eb648debc34b3b0ad8310f53656fd7a1/PSTN1.GIF/thumb.jpg" ALT="Click for full size"></A><br>PSTN tab 1/3</TD></TR><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/16960116?c=1066557&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG TITLE="25475 bytes" BORDER=0 WIDTH=588 HEIGHT=786 SRC="/r0/download/1066557~966f85a41e22fd2987554c4bbbc7e84e/PSTN2.GIF"></A><br>PSTN tab 2/3</TD></TR><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/16960116?c=1066558&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG class="apic" BORDER=0 TITLE="34193 bytes" WIDTH=600 HEIGHT=657 SRC="/r0/download/1066558.thumb600~7410ee1b333e584d2e2ade3e8e1b37fb/PSTN3.GIF/thumb.jpg" ALT="Click for full size"></A><br>PSTN tab 3/3</TD></TABLE></div>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,16960116</guid>
<pubDate>Sat, 23 Sep 2006 15:22:36 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,16959684</link>
<description><![CDATA[<A HREF="/useremail/u/1268357"><b>chicha36</b></A> : Those screenshots came in handy.  I had made a bunch of useful changes the other day, but for one.  Someone had written that a good value for DTMF Playback Level was -1.  Well, VOIP calls worked fine.  The next day, though, when I made an FXO call, I got a long dialtone AFTER dialing the number.  Looking at your regional tab, I changed the value to -9 and that solved the problem.  I don't see your PSTN tab values, though!!!  [I'm going to print out those screenshots you posted.]   Can you post a screenshot of your PSTN tab?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,16959684</guid>
<pubDate>Sat, 23 Sep 2006 13:54:54 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,16881358</link>
<description><![CDATA[<A HREF="/useremail/u/637921"><b>mazilo</b></A> : <div class="bquote"><SMALL>said by vincentdelporte :</SMALL><br><br>maybe it'd be a good idea to extract the tricks into a single document and make it sticky?</DIV>Well, this may not be a bad idea if you start it out ;).<br><SMALL>--<br>Mazi (UK Non-Geo Phone: +44-703-194-2574)</SMALL>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,16881358</guid>
<pubDate>Mon, 11 Sep 2006 11:47:43 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,16880904</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : > DracoFelis : since this thread is getting pretty long, to the point where I can't even go through all of it to find out if someone knows the trick to configure the Linksys to have it dial out a remote IP phone on the Net directly... maybe it'd be a good idea to extract the tricks into a single document and make it sticky?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,16880904</guid>
<pubDate>Mon, 11 Sep 2006 10:21:14 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,16800030</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Does anyone know any tricks to use the remote sipura page, the sipura I have is remote on a slow speed link, when I try to access the sipura page, it opens and cant configure as jumping to different screens takes a while and times out, have used IE, mozilla]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,16800030</guid>
<pubDate>Tue, 29 Aug 2006 09:34:26 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,16570430</link>
<description><![CDATA[<A HREF="/useremail/u/1377147"><b>cwwny</b></A> : I have a Linksys PAP2 device and followed DracoFelis directions to get forward all inbound calls to line2 from line1.  Please help......<br><br>The phone on line2 actually rings but when I pickup the caller's device is still ringing. So my line1 provider is forwarding my line2user@externaldnsname:5061 to line2 and the phone ring but why does it not pickup? <br><br>I have every settings enabled that is documented.  The following is my SIP settings and Make Call Without Reg and Ans Call Without Reg both enabled.  <br><br>Reg Retry Intvl:                 30<br>           Reg Retry Long Intvl:            300<br>           Handle VIA received:             yes<br>           Handle VIA rport:                yes<br>           Insert VIA received:             yes<br>           Insert VIA rport:                yes<br>           Substitute VIA Addr:             yes<br>           Send Resp To Src Port:           yes<br>           STUN Enable:                     yes<br>           STUN Test Enable:                yes<br>           STUN Server:                     stun.sipgate.net:10000<br>           NAT Keep Alive Intvl:            15]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,16570430</guid>
<pubDate>Tue, 25 Jul 2006 01:29:31 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,16547116</link>
<description><![CDATA[<A HREF="/useremail/u/530828"><b>DrFillster</b></A> : This tip (see cytlor's post on page 8) should be a sticky.  Thanks ctylor.<br><br>I couldn't get the RTP packet size down to 0.03; had to settle at 0.02 with latency of medium.<br><br>Made a huge difference in quality of calls.<br><br>Thanks again]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,16547116</guid>
<pubDate>Fri, 21 Jul 2006 13:44:47 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,16415777</link>
<description><![CDATA[<A HREF="/useremail/u/1264021"><b>Sukru Bey</b></A> : <div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><br><br><div class="bquote"><SMALL>said by Pelikano  :</SMALL><br><br>Enable IP Dialing: yes<br></DIV>If you are going to suggest a setting, please also include a short description as to why you want to use that setting.  Otherwise, nobody knows the context of your "trick".</DIV>I am glad Pelikano suggested Enable IP Dialing: Yes<br><br>I have 2 PAP2. One has regular PAP2 fw, and the other has SPA 1001 fw. I have no problem of calling a sipphone.com conferance feature using speed dial (e.g. 12226253703@proxy01.sipphone.com on Speed dial 5) from SPA 1001, but not PAP2. All settings are the same. So, I wasn't able to understand why SPA 1001 calls 12226253703@proxy01.sipphone.com but not PAP2.<br><br>Right after Enable IP Dialing: Yes , I was able to call from PAP2. So I suggest every one to do the same if they are having problems making IP calls.<br><SMALL>--<br>SUKRU BEY, a Linksys PAP2 user</SMALL>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,16415777</guid>
<pubDate>Sat, 01 Jul 2006 11:58:18 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,16385020</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : the only ports Zingotel uses to get into the device are either 80 or 8765. The 10000-20000 UDP, 5060 UDP, and 443 TCP are fowarded so packets always reach their destination and don't get caught up and lost on your network. These ports are usually the same no matter what VoIP service you use.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,16385020</guid>
<pubDate>Tue, 27 Jun 2006 01:22:52 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,16172193</link>
<description><![CDATA[<A HREF="/useremail/u/1360986"><b>gadgetavi8or</b></A> : Thanks to all for the useful tips.  I was forced to solve a few problems that I could not find mentioned, I hope these tips are helpful.<br><br>I have an SPA3000 (Linksys version) on Broadvoice.  I am using a analog POTS (Plain old telephone service) line for local calls and 911, I am using BV/VoIP for LD calls.  <br><br>(*xx|#xx|91xS0|[346]11|0|00|[2-9]xxxxxxS0|1xxx[2-9]xxxxxx.|xxxxxxxxxx.)<br><br>I ran into some "interesting" problems where calls from FXS/LINE1 to the gw0/FXO/PSTN would end up with a dial tone or sometimes a recording.  In this dial plan 7 digit calls or 91x calls.<br><br>Upgrading to the latest SW (BV is one version behind) did not help.  After much research, I determined the problem was the DTMF Playback level default is too high for my analog line.  Adjusting to -3 to -5 allowed for all local calls to work correctly.  <br><br>In the process of debugging this issue I changed the "PSTN Answer Delay:" to 1 and forgot to change it back to the default of 16.  This created a similar but unrelated issue when receiving calls on the PSTN.  Calls were correctly transferred to line one but callers would hear a dial tone and the FXS line would ring once or not at all.  (I think this tone is the VoIP1 interface dial tone), either way this issues were addressed by increasing this delay to something like five+ rings or 16+ seconds.<br><br>The fine posts regarding BV were on target, I chose them over Sunrocket because they support for BYOD and they were on board of providing me the admin PWDs.<br><br>One final item, I ran ping tests to all of the BV proxies and changed their setting from NYC to MIA, the proxy that had the best ping times for me.  Call quality was noticeably improved. An easy optimization if your VoIP provider has multiple proxies.<br><br>I hope these tips help as much as many of the previous posts helped me.<br><br>Thanks again<br><br>Doug]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,16172193</guid>
<pubDate>Thu, 25 May 2006 19:48:17 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,16128055</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : I'm using an SPA-2100 on ZingoTel.  At first, I was quite happy with the box since calls sounded great.  I later was running some speed tests on my cable connection using &raquo;<A HREF="http://www.speakeasy.net/speedtest/" >www.speakeasy.net/speedtest/</A> and I noticed that I was getting very poor performance on my upload speed.  When I went to my cable tech support, we unhooked the SPA-2100, and damned if the cable speed didn't pop up to what it should be.  The SPA-2100 was eating up 2/3 of my cable connection bandwidth!!!  <br><br>I called and talked with ZingoTel, and it turns out that the QoS (Quality of Service) feature in the SPA-2100 (turned on by default) does indeed eat up the bandwidth coming out of the box.  I even found some references on the web to this issue:<br><br>(&raquo;<A HREF="http://faq.sipbroker.com/tiki-index.php?page=SPA-2100" >faq.sipbroker.com/tiki-index.php&middot;&middot;&middot;SPA-2100</A>)<br>Under Sipura Flaws, I found:  "QoS reduces available bandwidth even when there is no active phone call. I don't think there is any good reason to reduce available bandwidth except during an active phone call. (However, ALL the devices I know of that implement QoS do it this way, so it's not just a Sipura problem.)"  <br><br>The only way to beat this was for them to turn of the QoS (which I did), and then my bandwidth went back to normal.  Problem was, the sound on the calls wasn't very good.  So I decided to upgrade my cable bandwidth in the hopes that would help.  It didn't help, but now I'm addicted to the higher speed!  LOL!<br><br>Now to be clear, I had the SPA-2100 set up in a typical manner where it was connected between my cable modem and my PC, acting as a router.  I suffered with this for a while, but then about a week ago, I bought a wireless router.  My girl friend wanted to be able to connect with her laptop when she was at my place, and I liked the idea of a hardware firewall over a software firewall, so I got the new router.<br><br>So now I had the cable modem, then the SPA-2100, then the wireless router, and then my PC, all daisy chained together.  All was working good except the quality of the phone service sucked.<br><br>Well I decided something just wasn't right, and maybe I needed a different setup since this just wasn't getting the job done.  So I called back to the ZingoTel folks to try to get them to send me a different ATA box, but this time I got someone in tech support that was a bit more knowledgeable.  He told me that if I connected the SPA-2100 to my new router and also connected my computer to the router (instead of the back of the SPA-2100), that my speed wouldn't be killed by the QoS, the SPA-2100 would work great, and we could turn the QoS back on for good quality calls.  He was exactly right.  It now works perfectly and I have great sound and good speed.<br><br>It turns out that when you connect your PC to the router built into the SPA-2100 box (as you're typically instructed to do), the QoS severely limits the bandwidth that gets through to your PC.  However, if you instead have another router available to you, just connect both the SPA-2100 and the PC up to that router, and all will be good.<br><br>Once caveat.  He did tell me that I'd need to adjust the router's firewall setting for the SPA-2100.  This is done by first going to the phone, dailing **** and then 110#.  This will give you the IP address that the router has assigned to the SPA-2100.  Then go into the router set up, and add that IP address to the router's DMZ so that the SPA-2100 is now outside the router's firewall.  <br><br>I'm not sure this is really necessary since the phone works fine without doing it.  My guess is that he wanted me to do this is only so that their tech support can talk to the SPA-2100 without my router firewall getting in the way.  I've actually turned that off since I think it's good that the SPA-2100 is behind my firewall, too.  However, I may need to enable it if I talk to their tech support in the future.<br><br>Hope you find this useful.<br><br>John]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,16128055</guid>
<pubDate>Fri, 19 May 2006 03:49:07 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,16114831</link>
<description><![CDATA[<A HREF="/useremail/u/728530"><b>SuperCPA</b></A> : SPA2K and SPA3K<br><br>I&#146;m not sure if this qualifies as a &#147;trick&#148;, or just a combination or iteration of utilizing the power of the SPA&#146;s.<br><br>My office setup.  Each of the SPA&#146;s is connected to a two line phone, and the SPA&#146;s are on the same LAN.  I have a fixed IP for a WAN address, and set the SPA&#146;s also to a fixed internal IP.  A variation of this procedure may also work with a dynamic WAN IP if utilizing something like DynDNS.org, if supported by your router.<br><br>The SPA3K is set up with Broad Voice on Line 1, port 5060, and Delta Three on the PSTN Line, port 5061.  The SPA2K has FWD on line 1, port 5062, and Voice Pulse Connect on Line 2, port 5063.  Only the Broad Voice line accepts an incomming call.  The other lines are outgoing only.<br><br>The objective is effectively a direct dial hot line form the phone attached to the SPA2K to the SPA3K PSTN Line.<br><br>On the SPA3K SIP Tab, enable &#147;Send Resp To Src Port&#148; down by the STUN settings.  I&#146;m not sure if the is required, but it allows both calls to my FWD and IPKall numbers to ring the SPA3K Line 1 directly.<br><br>On the Line Tab&#146;s of each ATA adapter ensure that &#147;Use DNS SRV&#148; and &#147;DNS SRV Auto Prefix&#148; is set to no.  I also have enabled &#147;Make Call Without Reg&#148; and &#147;Receive Call Without Reg&#148; on both adapters.<br><br>The real power of both the SPA&#146;s is in the dial plan.<br><br>For the SPA3K, add:<br><div class="code"><PRE><span class="codetext">&lt;#0:&gt;&lt;:gw0&gt;</SPAN></PRE></DIV>to the Line 1 dial plan. On the PSTN Line, add:<br><div class="code"><PRE><span class="codetext">&lt;:1937&gt;&#91;2-9&#93;xxxxxx</SPAN></PRE></DIV>to &#147;Dial Plan 1&#148;.  You could use any default area code.  On the VoIP To PSTN Gateway Setup, set &#147;One Stage Dialing&#148; to yes.  Set the &#147;Line 1 VoIP Caller DP&#148;, &#147;Line 1 Fallback DP&#148;, and &#147;VoIP Caller Default DP&#148; to 1.  Add your WAN IP address to the &#147;VoIP Access List&#148;.<br><br>For the SPA2K, for the Line 1 and Line 2 dial plans add:<br><div class="code"><PRE><span class="codetext">&lt;#0:&gt;&lt;:User ID SPA3K PSTN Line@Your WAN IP Address:5061&gt;</SPAN></PRE></DIV>and<br><div class="code"><PRE><span class="codetext">911&lt;:User ID SPA3K PSTN Line@Your WAN IP Address:5061&gt;.</SPAN></PRE></DIV><br>The result is that dialing #0 from any of the three VoIP Lines results in a PSTN dial tone, and all calls to 911 from the SPA2K  are now automatically dialed from the PSTN Line on the SPA3K.  I did make a test call to 911 form the SPA2K, and it does work.<br><br>I&#146;m still thinking about PSTN Line access security, but for me it&#146;s generally not an issue, as there is no long distance provider set up through the phone company, and the dial plan restricts PSTN calls to the local area code.  It works for me, my office set up, and the way that the phones are used.  Any comments, criticism or suggestions would be appreciated.   <br>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,16114831</guid>
<pubDate>Wed, 17 May 2006 11:43:05 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,16078002</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Does the SPA-3102 improve upon "two registered providers not allowed" problem?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,16078002</guid>
<pubDate>Thu, 11 May 2006 18:54:33 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15953083</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  halalani <A HREF="/useremail/u/1351716"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>I have just bought SPA 3000. I have two SIP accounts and need help as how to set-up the two to ring to one phone.</DIV><B>Reminder:</B><br>This is NOT a questions thread, this is a "tricks" (solutions) thread.  Please start new threads for questions, and only post "useful tricks" here.<br><br><B>That said, here is the "tricks" you can use to do this:</B><br><br>1) If both providers require you to be "registered" (to receive inbound calls), you are SOL, as the SPA-3000 only gives you one "registered" inbound VoIP line for receiving calls (the PSTN VoIP provider is "special", and can NOT be used for normal inbound calls).<br><br>2) However, if one provider will accept inbound "SIP URI" (VoIP address) calls, and the other provider will let you "forward" to a SIP URI, than it is easy.  In that case, have the one provider "registered" as your "Line 1" provider, and have the other provider forward to the 1st provider.  Voila, both will "ring your phone" (one directly, and the other via the forwarding).<br><br>3) If at least one of the providers allow you to forward to a SIP URI, but the other provider won't accept the inbound call, than you can try setting up this "trick" to forward directly to your adapter.  It's a PITA to do, but it does work nicely once you get all the details right:<br>&raquo;<A HREF="http://faq.sipbroker.com/tiki-index.php?page=Inbound%20Calls%20Directly%20to%20your%20LinkSys%20or%20Sipura" >faq.sipbroker.com/tiki-index.php&middot;&middot;&middot;20Sipura</A>.<br><br>NOTE:  The above details all talk about getting a 2nd/3rd/etc provider to "ring your adapter".  However, since you only have one main provider on the SPA-3000, you also have to use some "trick" to call out via those other providers.  The easiest way to handle that, is the SPA-3000 "gateway" slots.  Please look at older posts in this thread for details about how to properly use the SPA-3000's "gateways"...]]></description>
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<pubDate>Sun, 23 Apr 2006 11:17:09 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15952893</link>
<description><![CDATA[<A HREF="/useremail/u/529880"><b>ebruce</b></A> : <div class="bquote"><SMALL>said by  halalani <A HREF="/useremail/u/1351716"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>Hi,<br><br>I have just bought SPA 3000. I have two SIP accounts and need help as how to set-up the two to ring to one phone.<br><br>Thanks<br> </DIV>If I'm not mistaken, you can only have 1 incoming SIP provider (and one incoming PSTN) with a SPA-3000. It only allows for multiple outgoing SIP providers.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15952893</guid>
<pubDate>Sun, 23 Apr 2006 10:28:23 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15952430</link>
<description><![CDATA[<A HREF="/useremail/u/1351716"><b>halalani</b></A> : Hi,<br><br>I have just bought SPA 3000. I have two SIP accounts and need help as how to set-up the two to ring to one phone.<br><br>Thanks]]></description>
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<pubDate>Sun, 23 Apr 2006 07:00:14 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15944788</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <B>Here's how to take control of how often your adapter "registers":</B><br><br>I'm sure you all know about the "<B>Register Expires:</B>" parameter (on the "Line x" tab).  This is supposed to control how often your adapter registers (the parameter is in seconds).<br><br>However, I have been trying to figure out why the registration interval could sometimes be a lot shorter, or a lot shorter than I supposedly set in "Register Expires:".  I finally found some of the other settings (and other factors that effect registration).  Here they are (as best I can figure):<br><br>It appears that the SIP provider you try to contact can override the registration interval (by the SIP messages they send when you try to register).  This can greatly change how often you thought you would register...<br><br><B>To take control back, here are some other settings to adjust:</B><br><br>You can specify a MINIMUM normal registration interval, below which the provider can't register more often, by setting the "<B>Reg Min Expires:</B>" value (advanced "SIP" tab).<br><br>You can specify a MAXIMUM normal registration interval which you will always reregister by (even if the provider wants to register less often) by setting "<B>Reg Max Expires:</B>" (again, SIP tab).<br><br>In addition, the two parameters "<B>Reg Retry Intvl:</B>" and "<B>Reg Retry Long Intvl:</B>" (also "SIP" tab) seem to control how long you wait to try registering again, if a registration attempt "fails".  This can be important, as you often want to try again "reasonably soon" if/when the registration has a problem and "fails".  Because while you aren't registered, you aren't receiving inbound calls (and with some providers, you also can't make outbound calls when not registered).]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15944788</guid>
<pubDate>Fri, 21 Apr 2006 19:13:22 EDT</pubDate>
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<item>
<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,15782020</link>
<description><![CDATA[<A HREF="/useremail/u/1104216"><b>krishn</b></A> : i apologise,<br><br>If you wish I would delete the message. I gotta learn how to delete it though.<br><br>Krishn]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15782020</guid>
<pubDate>Wed, 29 Mar 2006 01:11:28 EDT</pubDate>
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<item>
<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,15774151</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  krishn <A HREF="/useremail/u/1104216"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>my voip provider is sipdiscount and I wish to generate a dial plan which would allow me to call ONLY USA and CANADA (numbers starting with 1) and nothing else, if I try to dial something else it would give a fast busy signal. Anybody got any idea how I would do it?</DIV>How to do it with a LinkSys/Sipura adapter has already been explained earlier in this thread.  Specifically, look at this post:<br>&raquo;<A HREF="/forum/remark,13813346">Blocking costly area codes via your "Dial Plan".</A><br><br><div class="bquote"><SMALL>said by  krishn <A HREF="/useremail/u/1104216"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>I will be using my DTA flashed to become BVA 8051 with AR 160 firmware.</DIV>So why are you posting in a "<B>Sipura</B> Tricks" thread (when your post seems to imply you are using a hacked Packet8 adapter, <B>not</B> a LinkSys/Sipura model adapter)?!?<br><br>Not only is this thread for "tricks" (i.e. solutions, not questions), but your adapter isn't even a LinkSys/Sipura model (nor would the details of setting up your ATA be anything like setting up a LinkSys/Sipura adapter)!  <br><br>NOTE:  There is nothing wrong with using a hacked Packet8 adapter, if that's what you want to do.  I have an old Packet8 DTA from when I had Packet8 service, and I've thought of flashing it at some point (if I ever feel the need to have an extra VoIP adapter around the house).  But such discussions don't IMHO belong in the middle of a message thread about using LinkSys/Sipura model adapters (as a hacked Packet8 DTA has little in common with how you setup/use a Sipura model adapter)!]]></description>
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<pubDate>Tue, 28 Mar 2006 00:04:20 EDT</pubDate>
</item>

<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15773834</link>
<description><![CDATA[<A HREF="/useremail/u/578422"><b>lucky644</b></A> : I can't believe I didn't see this thread earlier..<br><br>Perhaps someone here might be able to answer this.<br><br>I'm going to buy a SPA-9000 for work, we need a system to forward off to our cells when we're not in the office..<br><br>Does anyone know of a VOIP provider that can port a Canadian number that will let me use the SPA-9000?<br><br>I have Vonage, but they won't tell me my details to manually configure my own PBX/VOIP box.  :(]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15773834</guid>
<pubDate>Mon, 27 Mar 2006 23:13:26 EDT</pubDate>
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<item>
<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,15756915</link>
<description><![CDATA[<A HREF="/useremail/u/1104216"><b>krishn</b></A> : my voip provider is sipdiscount and I wish to generate a dial plan which would allow me to call ONLY USA and CANADA (numbers starting with 1) and nothing else, if I try to dial something else it would give a fast busy signal. Anybody got any idea how I would do it? I will be using my DTA flashed to become BVA 8051 with AR 160 firmware.<br><br>thanks to all those who even read it and helping hands will go to heaven !<br><br>Krishn]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15756915</guid>
<pubDate>Sat, 25 Mar 2006 03:45:02 EDT</pubDate>
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<item>
<title>bridge device</title>
<link>http://www.dslreports.com/forum/remark,15722135</link>
<description><![CDATA[<A HREF="/useremail/u/1341788"><b>jeanclaudeno</b></A> : I am looking for a "telephone adapter" which can permit me to bridge 2 telephones lines. This is adapter is suppose to be able to permit a remote user to access the 2nd telephone line from the 1st line. By example, if at my home I have 2 analog telephone lines, while I am out, I am supposed to be able to "dial in" to the adapter from one line (while I am remote), and "dial out" using the second line. <br>Please advise me if you are aware of this product and can you refer me to a place which can provide me such an adapter. (jeanclaudenow@yahoo.com)<br>Thanks.]]></description>
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<pubDate>Mon, 20 Mar 2006 12:34:09 EDT</pubDate>
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<item>
<title>telephone adapter</title>
<link>http://www.dslreports.com/forum/remark,15721948</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : I am looking for an ANALOG "telephone adapter" which can permit me to bridge 2 telephones lines. This is adapter/device is suppose to be able to permit a remote user to access the 2nd telephone line from the 1st line. By example, if at my home I have 2 analog telephone lines, while I am out, I am supposed to be able to "dial in" the adapter from one line, and while I am remote, "dial out" using the second line. <br>Please advise me if you have this product available otherwise please refer me to a place which can provide me such an adapter. <br>Thanks.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15721948</guid>
<pubDate>Mon, 20 Mar 2006 12:00:23 EDT</pubDate>
</item>

<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15712594</link>
<description><![CDATA[<A HREF="/useremail/u/620161"><b>igi</b></A> : Topic #1: Jitter/RTP<br><br>My experience has been slightly different, hope it helps somebody. My main use for the sipura boxes is to speak with family/friends overseas (from the US standpoint) using FWD. The other side uses a Sipura box through an ADSL connection. I use broadband (cable). I got a lot of mileage by dropping the jitter to "low" but quite disastrous results when dropping the RTP packet size to 10ms. I think it's because of the uplink speed for ADSL overseas. So I settled for the original 30ms on their end, 20ms on my end, and low jitter on both. That works great, and I'd rather have low latency (I really don't like the "walkie-talkie" conversations so prevalent now with cell phones).<br><br>Topic #2: the Sipura "web bug"<br>I can get that to work fine, if I go through a proxy server. I have a linux box that I can use as such, and I've tried this with other proxy servers. Not fool-proof but way better than a direct connection. I don't know if teh Linux buffers are solving this issue. So maybe those windows users can try  going through a proxy for remote accessing a Sipura.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15712594</guid>
<pubDate>Sat, 18 Mar 2006 18:59:20 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15700895</link>
<description><![CDATA[<A HREF="/useremail/u/1340291"><b>ctylor</b></A> : I discovered a good way to test your connection for jitter, using SIPPhone/Gizmo-Project's Welcome Recording (1-747-474-5000 in Gizmo, or *74717474745000# through SIP Broker). And I have to admit, in listening to that long boring monologue by that voice actor, I did get a fair amount of blips and voice-dropouts when my setting was at Low. I am doing the test at the busiest time of usage but by the same token, this is the time when I would probably make most of my personal phone calls. Moving the Jitter Buffer setting back to High seemed to help a fair amount, so I guess I have to actually adopt that setting myself "just to be safe". So FYI, I am not even ultimately adopting my own tweak. Ha!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15700895</guid>
<pubDate>Thu, 16 Mar 2006 23:23:30 EDT</pubDate>
</item>

<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15695135</link>
<description><![CDATA[<A HREF="/useremail/u/1340291"><b>ctylor</b></A> : Needless to say in the previous post I formatted it completely wrong and where it said ctylor said in blockquotes is actually Draco's words and where it said DracoFelis said, are of course my words. Sorry.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15695135</guid>
<pubDate>Thu, 16 Mar 2006 07:06:13 EDT</pubDate>
</item>

<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15693995</link>
<description><![CDATA[<A HREF="/useremail/u/1340291"><b>ctylor</b></A> : <div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR><div class="bquote"><SMALL>said by ctylor  :</SMALL><BR><BR>While ctylor found a "Network Jitter Level: <B>low</B>" worked well for him, my tests (with my ISP) showed differently.  While it is true that a setting of "low" results in the lowest possible latency/echo (which is "a good thing"), I started having problems with "stuttering sounds" (fraction of a second drop-outs of sound) when I used the "low" setting.  So I boosted my setting to "medium" which helped (but didn't eliminate the effect), and eventually put it back to the default of "high" (which works reasonably well with my ISP).  Granted, I lost the lower latency/echo that I gained by setting it to "low", but using a setting of "high" also caused me to lose the constant "stuttering" on my VoIP line!  So lowering the "Network Jitter Level:" setting is clearly a YMMV setting.  It's worthwhile trying, because it will lower latency/echo, but be prepared to set it back up if/when you start getting "stuttering" on the line...<br><br>NOTE:  Yes, that means that I'm currently using trick A (more bandwidth used, by lowering my RTP packet size), but not trick B (because, in my case, lowering my jitter buffer caused problems with packet loss, which was a worse problem for our family than the latency/echo caused by the default jitter buffer size)...<br> </DIV>Hi Draco, were you testing this setting using 'real world' conditions of actual phone calls with people, or mostly with  IVR systems and echo tests (like FWD's)? I have difficulty replicating your stuttering/packet discard problem when the Jitter Buffer setting is on "Low" on echo tests, IVRs, or any of the phone calls I've had in the last week.<br><br>(Well not entirely true: I had a distinct problem last night when bit torrent was up and running--the echo test was almost comical! The end to end latency increased from very slight to nearly a second and half, and every fourth word (when it would finally arrive two seconds later!) would have a syllable or two get 'dropped'. That is the only stutter I've been experiencing since I've made this change on my SPA-2100.)<br><br>So I think with this one, Draco is definitely right: YMMV--your mileage may vary. I would have to guess there are two, potentially three main factors that determine whether you naturally experience a lot of network jitter with your phone calls: 1) Your ISP, 2) Your VOIP Provider and distance in ms to your sip proxy. Both 1 and 2 differ between Draco and me. Also thirdly, I am using a SPA-2100, while he is using a SPA-3000. So our test conditions do differ. But I encourage people to try it and see if they get Draco's problem of uncorrected jitter on their connection. I don't find I do on mine.]]></description>
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<pubDate>Wed, 15 Mar 2006 23:42:34 EDT</pubDate>
</item>

<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15689473</link>
<description><![CDATA[<A HREF="/useremail/u/1053070"><b>rizzo2dial</b></A> : <div class="bquote"><SMALL>said by  rizzo2dial <A HREF="/useremail/u/1053070"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>Here's how to configure LINE1 of the SPA-1001 to dial "911" on LINE2:<br><br>1) In the "<B>Phone</B>" tab, set the "Line 2 Select Code" to be the default value of "#"<br><br>2) In the "<B>User 1</B>" TAB, configure a "Speed Dial [2-9]" entry of your choice as "#911" (for dialing "911" on LINE2).<br><br>3) Assuming you configured entry "Speed Dial 2" in the previous step, insert into the beginning of the LINE1 Dialing plan the following:<br><div class="code"><PRE><span class="codetext">&lt;911:2&gt;S0|</SPAN></PRE></DIV>If you configured a different "Speed Dial [2-9]" entry, replace the ":2" above with ":<I>x</I>"<br>(where <I>x</I> is the configured "Speed Dial [2-9]" entry #).<br><br>When dialing 911, you'll hear a LINE2 dialtone for a split second followed by the true 911 call on LINE2.<br><br>Even if you dial any digits after 911 on LINE1 (i.e. 9111, 91111, 911111, etc.), they'll be ignored (which is precisely what you want for 911 purposes).<br><br>With this solution, you retain full LINE2 functionality should you want/need to manually dial out on that line.<br><br>Rizzo</DIV>It dawned on me that w/ the solution above, if somebody were to inadvertently dial "2#" on their phone, it would connect them to 911.  This could result in an unexpected/unwanted visit from the local police department.<br><br>Here's your choice of modified <B>LINE1 dialing plans prefixes</B> to continue to allow "911" to dial SPEED DIAL ENTRY #2 (in the USER1 tab) while preventing the directly dialed "2#" SPEED DIAL ENTRY from calling 911:<br><br><U>Option #1:</U> Directly dialing "2#" <B>results in a FAST BUSY SIGNAL</B><br><div class="code"><PRE><span class="codetext">&lt;911:2&gt;S0|&lt;2#:&gt;S0|</SPAN></PRE></DIV><br><U>Option #2:</U> Directly dialing "2#" <B>gives the LINE2 DIAL-TONE</B><br><div class="code"><PRE><span class="codetext">&lt;911:2&gt;S0|&lt;2#:#&gt;S0|</SPAN></PRE></DIV><br><U>Option #3:</U> Directly dialing "2#" <B>dials a phone number of your choice</B><br><div class="code"><PRE><span class="codetext">&lt;911:2&gt;S0|&lt;2#:xxxxxxxxxxx&gt;S0|<br>Replace 'xxxxxxxxxxx' w/ the phone number of your choice.</SPAN></PRE></DIV><br>Personally I like Option #2, although Option #3 lets you convert "2#" back into a psuedo-Speed Dial Entry for NON-911 purposes (while allowing the REAL Speed Dial Entry to call 911 on LINE2).<br><br>Rizzo<br>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15689473</guid>
<pubDate>Wed, 15 Mar 2006 13:18:16 EDT</pubDate>
</item>

<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15689038</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by ctylor :</SMALL><BR><BR><B>A. Trading Bandwidth for Better Sound Quality<br>B. Changing Your Jitter Buffer Setting</B><br><br>Draco asked me to add this trick to the list. Apparently all Sipuras ships with an RTP Packet Size set to 0.030 (i.e. 30ms) by default, or 33 packets per second (pps). This of course causes some degree of added latency and issues revolving around lost or discarded packets leading to greater quality loss than if a shorter interval is used.<br><br>It seems most sites in the internet recommend, or at least treat as standard, that you should usually only set the RTP Packet Size to 30ms when you use G.723.1, and to 20ms (i.e. 50 pps) when you pretty much use anything else the Sipura presently supports. 10ms (i.e. 100 pps) is also an option for the highest quality sound with the lowest degree of latency (especially when combined with a jitter buffer setting of low) when using G.711 or G.729 but at the cost of utilizing the highest bandwidth possible.<br><br>At 10ms (highest quality) / 20ms (generally recommended setting) / 30ms (Sipura default; required for G.723.1), the overall bandwidth average for ____ is:<br>G.711 - 126kbps / 95.2kbps / 84.7kbps<br>G.729ab - 70.4kbps / 39.2kbps / 28.7kbps<br><br>A full interactive chart with these codecs and several others can be found here so you can find the right balance between codec and sample period to suit your uplink capacity.<br><br>Your network jitter setting is another area to address in your attempt to reduce latency and improve audio quality. The following tip may have undesirable consequences on certain connections at certain congested times, but overall my experience with it has been very positive.<br></DIV>BTW:  Yes, I was the one who put ctylor up to posting here.  After his excellent writeup of these issues on the Voxilla.com forums, I just wanted this info preserved here (so that it would be available to those trying to learn how to best setup their LinkSys/Sipura adapters).  I would just like to add a couple of points:<br><br>A) Lowering the RTP packet size (to either 0.020, or all the way down to 0.010) is a straight trade-off of using more bandwidth for better sound quality.  And remember the extra 30k+ cost of using an RTP packet size of 0.010 (vs the default 0.030) is paid for ALL CODECs, even "low bandwidth" ones such as G729.  In fact, setting the RTP packet size to 0.010 can more than double the normal bandwidth cost of G729 (all the way up to 70k+ bandwidth usage).  While that is still lower than G711 (much less G711 with this tweak), keep that bandwidth usage in mind if/when you are on a bandwidth limited ISP.<br><br>NOTE:  If your VoIP provider doesn't let you use G711 (and limits you to "lower bandwidth" CODECS, such as the previously mentioned G729), this is one way to raise the bandwidth used (and therefore also the sound quality used) by whatever CODEC(s) your provider does allow to use...<br><br>B) On the question of your "Network Jitter Level:" setting, no matter what you set it to you will be using the same amount of bandwidth.  So the trade-off here, is better sound quality (and less "latency"/echo in the call) if you have a "low jitter" ISP connection, vs being more "conservative" with jitter (at the expense of possible "sound breakup").<br><br>While ctylor found a "Network Jitter Level: <B>low</B>" worked well for him, my tests (with my ISP) showed differently.  While it is true that a setting of "low" results in the lowest possible latency/echo (which is "a good thing"), I started having problems with "stuttering sounds" (fraction of a second drop-outs of sound) when I used the "low" setting.  So I boosted my setting to "medium" which helped (but didn't eliminate the effect), and eventually put it back to the default of "high" (which works reasonably well with my ISP).  Granted, I lost the lower latency/echo that I gained by setting it to "low", but using a setting of "high" also caused me to lose the constant "stuttering" on my VoIP line!  So lowering the "Network Jitter Level:" setting is clearly a YMMV setting.  It's worthwhile trying, because it will lower latency/echo, but be prepared to set it back up if/when you start getting "stuttering" on the line...<br><br>NOTE:  Yes, that means that I'm currently using trick A (more bandwidth used, by lowering my RTP packet size), but not trick B (because, in my case, lowering my jitter buffer caused problems with packet loss, which was a worse problem for our family than the latency/echo caused by the default jitter buffer size)...]]></description>
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<pubDate>Wed, 15 Mar 2006 12:21:09 EDT</pubDate>
</item>

<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15688852</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  rizzo2dial <A HREF="/useremail/u/1053070"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>Here's how to configure LINE1 of the SPA-1001 to dial "911" on LINE2:<br><br>1) In the "<B>Phone</B>" tab, set the "Line 2 Select Code" to be the default value of "#"<br><br>2) In the "<B>User 1</B>" TAB, configure a "Speed Dial [2-9]" entry of your choice as "#911" (for dialing "911" on LINE2).<br><br>3) Assuming you configured entry "Speed Dial 2" in the previous step, insert into the beginning of the LINE1 Dialing plan the following:<br><div class="code"><PRE><span class="codetext">&lt;911:2&gt;S0|</SPAN></PRE></DIV>If you configured a different "Speed Dial [2-9]" entry, replace the ":2" above with ":<I>x</I>"<br>(where <I>x</I> is the configured "Speed Dial [2-9]" entry #).<br></DIV>Hey, cool!  <br><br>Your "trick" is a very clever way to easily add "line 2" dialing (including 911 dialing, where the VoIP provider with 911 is on "line 2") to an SPA-1001's "dial plan" on an SPA-1001. <br><br>But even more than that, your "trick" lets you use a "speed dial" slot wherever you want in a LinkSys/Sipura "dial plan" (instead of having to access the "speed dial" by the single digit corresponding to that speed dial slot)!  I never knew you could use a "speed dial" slot like that!<br><br>And the really "cool" thing about your "trick", is that it is not just SPA-1001 specific.  While it is true that only the SPA-1001 allows you to access "line 2" via a "speed dial", the "trick" of accessing a speed dial in your "dial plan" seems to work on other Sipura adapters as well. Specifically, when I put the following pattern into my dial plan of my Sipura SPA-3000 (to test accessing speed dials from the dial plan), I was correctly connected to the location of my "speed dial 2" when I dialed "12" on the phone:<br><div class="code"><PRE><span class="codetext">&lt;12:2&gt;S0</SPAN></PRE></DIV><br>Hmmm...  Let's see...<br>This could end up being really handy, if I ever want to translate several different call numbers to the same target location, without having to change the dial plan each time I want to redirect them.  Instead, I just point all those targets to a single "speed dial" slot, and update the "speed dial" if/when I want to point them all somewhere else.  And this should also help keep down the length of my "dial plan" (by putting some of the logic in a "speed dial" or three), and Sipura dial plans do have a maximum number of characters they will accept.  And...<br>]]></description>
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<pubDate>Wed, 15 Mar 2006 11:51:57 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15682637</link>
<description><![CDATA[<A HREF="/useremail/u/1053070"><b>rizzo2dial</b></A> : <div class="bquote"><SMALL>said by  rizzo2dial <A HREF="/useremail/u/1053070"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>   :</SMALL><BR><BR>Speaking of SPEED DIALING, the USER1 tab "Speed Dial [2-9]" entires actually support dialing out on the opposite line of the SPA-1001.  For example, if the "Line 2 Select" key is left as the default value of "#" and a "SPEED DIAL x" entry on LINE1 gets configured as "#18005551212", it will dial "18005551212" on LINE2 (w/ a LINE2 dial-tone being heard for a split-second).<br><br>Unfortunately this same behavior doesn't occur when configuring it in a DIAL PLAN speed dial (as suggested in the quote below).  It also doesn't work when convertijng a DIAL PLAN speed dial entry into a USER TAB speed dial entry:<br><div class="code"><PRE><span class="codetext">&lt;911:2#&gt;S0</SPAN></PRE></DIV>Converts DIAL PLAN speed dial "911" to USER TAB "Speed Dial 2."  While dialing "2#" directly allows dialing out on LINE2 (if configured to do so), dialing "911" (where it gets converted to "2#") doesn't work.</DIV>I just figured out how to get a DIAL PLAN speed dial entry (like the one above) to successfully dial a USER TAB speed dial entry! In the DIAL PLAN SPEED DIAL entry, don't end the USER TAB SPEED DIAL entry w/ a '#' (pound) symbol.<br><br>Here's how to configure LINE1 of the SPA-1001 to dial "911" on LINE2:<br><br>1) In the "<B>Phone</B>" tab, set the "Line 2 Select Code" to be the default value of "#"<br><br>2) In the "<B>User 1</B>" TAB, configure a "Speed Dial [2-9]" entry of your choice as "#911" (for dialing "911" on LINE2).<br><br>3) Assuming you configured entry "Speed Dial 2" in the previous step, insert into the beginning of the LINE1 Dialing plan the following:<br><div class="code"><PRE><span class="codetext">&lt;911:2&gt;S0|</SPAN></PRE></DIV>If you configured a different "Speed Dial [2-9]" entry, replace the ":2" above with ":<I>x</I>"<br>(where <I>x</I> is the configured "Speed Dial [2-9]" entry #).<br><br>When dialing 911, you'll hear a LINE2 dialtone for a split second followed by the true 911 call on LINE2.<br><br>Even if you dial any digits after 911 on LINE1 (i.e. 9111, 91111, 911111, etc.), they'll be ignored (which is precisely what you want for 911 purposes).<br><br>With this solution, you retain full LINE2 functionality should you want/need to manually dial out on that line.<br><br>Rizzo<br>]]></description>
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<pubDate>Tue, 14 Mar 2006 14:52:56 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15682407</link>
<description><![CDATA[<A HREF="/useremail/u/1053070"><b>rizzo2dial</b></A> : <div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>I know you can reach a better solution with an SPA-3000.  Because with an SPA-3000, you just put your 911 provider on one of the 4 "gateway" slots, and then just tell your dial plan to use the "gateway" for 911 dialing.</DIV>I agree that an SPA-3000 would work perfectly.<br><br><div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>Edit:  If you do have an SPA-3000, check out a previous "trick" I posted in this (long) thread, on how to use the four SPA-3000 "gateway" slots to store (and use) the login credentials of additional outbound only VoIP providers!</DIV>For the situation I'm working w/ at the moment, an SPA-3000 is not available.<br><br><div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>BTW:  <br>Don't forget that if you are willing to give up E911, and do simple "speed dialing" (without the enhanced location info sent to your emergency center), you can replace a dialed 911 with a quick "speed dial" to whatever is a proper "emergency number" for your area, simply by playing with your "dial plan".  And that "trick" (already posted in a previous message in this thread) works in most LinkSys/Sipura adapters, simply by modifying your "dial plan"!</DIV>Speaking of SPEED DIALING, the USER1 tab "Speed Dial [2-9]" entires actually support dialing out on the opposite line of the SPA-1001.  For example, if the "Line 2 Select" key is left as the default value of "#" and a "SPEED DIAL x" entry on LINE1 gets configured as "#18005551212", it will dial "18005551212" on LINE2 (w/ a LINE2 dial-tone being heard for a split-second).<br><br>Unfortunately this same behavior doesn't occur when configuring it in a DIAL PLAN speed dial (as suggested in the quote below).  It also doesn't work when convertijng a DIAL PLAN speed dial entry into a USER TAB speed dial entry:<br><div class="code"><PRE><span class="codetext">&lt;911:2#&gt;S0</SPAN></PRE></DIV>Converts DIAL PLAN speed dial "911" to USER TAB "Speed Dial 2."  While dialing "2#" directly allows dialing out on LINE2 (if configured to do so), dialing "911" (where it gets converted to "2#") doesn't work.<br><br><div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>For example, if you check with your local emergency services, and find that a local emergency number is "1-555-555-1234" (that is a bogus number, use a number appropriate for your area), you can add this little pattern to the front of your dial plan (again, modified to use your real "emergency number", not the bogus 1-555-555-1234) to trap for a user dialing 911, and instead "speed dial" your local emergency number:<br><div class="code"><PRE><span class="codetext">&lt;911:15555551234&gt;S0</SPAN></PRE></DIV></DIV>Since E911 is available on the VoIP provider on LINE2 (normally used for incoming calls only), the solution above is precisely one of the things trying to be avoided.<br><br>Rizzo<br>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15682407</guid>
<pubDate>Tue, 14 Mar 2006 14:11:54 EDT</pubDate>
</item>

<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15680341</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  rizzo2dial <A HREF="/useremail/u/1053070"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</SMALL><BR><BR>Another flaw is that this implementation doesn't allow LINE2 to be used as a backup VoIP provider for other outgoing calls (unless you add more to the LINE2 dialing plan and can dial your other # within 1 second).<br><br>I'm hopeful a better solution can be reached; however, I haven't been successful in otherwise getting a LINE1 dialing plan instruction to dial out on LINE2.</DIV>I know you can reach a better solution with an SPA-3000.  Because with an SPA-3000, you just put your 911 provider on one of the 4 "gateway" slots, and then just tell your dial plan to use the "gateway" for 911 dialing.<br><br>Edit:  If you do have an SPA-3000, check out a previous "trick" I posted in this (long) thread, on how to use the four SPA-3000 "gateway" slots to store (and use) the login credentials of additional outbound only VoIP providers!<br><br>But I don't know if there is a better solution for E911 on an SPA-1001.  Your solution (for the SPA-1001) was IMHO very clever, and thanks for posting!<br><br>BTW:  <br>Don't forget that if you are willing to give up E911, and do simple "speed dialing" (without the enhanced location info sent to your emergency center), you can replace a dialed 911 with a quick "speed dial" to whatever is a proper "emergency number" for your area, simply by playing with your "dial plan".  And that "trick" (already posted in a previous message in this thread) works in most LinkSys/Sipura adapters, simply by modifying your "dial plan"!  <br><br>For example, if you check with your local emergency services, and find that a local emergency number is "1-555-555-1234" (that is a bogus number, use a number appropriate for your area), you can add this little pattern to the front of your dial plan (again, modified to use your real "emergency number", not the bogus 1-555-555-1234) to trap for a user dialing 911, and instead "speed dial" your local emergency number:<br><div class="code"><PRE><span class="codetext">&lt;911:15555551234&gt;S0</SPAN></PRE></DIV><br>Edit:  <br>Of course, the above "trick" assumes that you can acquire a real number to dial in the case of an emergency (and some areas only allow "911", without having a "real number" that also reaches the emergency center).  And it also assumes that you have a VoIP provider on your adapter that lets you dial that number (i.e. for most VoIP companies, that number will probably dial as a normal "long distance call").  And even with the above limits, you still won't get the enhanced "location info" that "real 911" will give you (so you will have to let the person on the other end of the line know where you are located, when you talk to them).  <br><br>But at least that (speed dial type 911) "trick" does let someone dial 911 on your phone, and reach real "emergency personal" appropriate for your area!  And unlike the other "tricks", that approach does NOT require you to have a VoIP account that has "real 911" enabled on it (because you are essentially having the LinkSys/Sipura itself convert 911 into a normal long distance call to an appropriate pre-chosen "emergency number" of your choice)!<br>]]></description>
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<pubDate>Tue, 14 Mar 2006 07:50:15 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15679696</link>
<description><![CDATA[<A HREF="/useremail/u/1053070"><b>rizzo2dial</b></A> : <div class="bquote"><SMALL>said by  p2pvoice <A HREF="/useremail/u/892472"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>   :</SMALL><BR><BR>DracoFelis:<br><br>I have SPA1001 with two virtual lines. Line 1 is set up for "home" and Line 2 is set up for "business" with a different ringtone. Each line is an extension on my Asterisk server. So, by using a single phone, I can tell from the ringtone, whether an incoming call is a "home" call or a "business" call. I use primarily Line 1 for outgoing calls using Asterisk's outbound routing.<br><br>Here is the "trick: I need to implement:<br><br>My business number is an IPKall number, which "points" to my "business" extension on my Asterisk server.<br><br>My home number is "ported" to an ITSP and it "points" my "home" extension.<br><br>This ITSP is fully 911 compliant; however, I use this ITSP for incoming calls only (don't use this ITSP for outgoing calls). Since this ITSP is fully 911 compliant, I would like only the 911 call to go through this ITSP instead of the other ITSPs configured in my Asterisk's outbound routing.<br><br>Will this "trick" (inspired by your SIPPhone post) work?<br><br>911 <br><br>I don't know where to put UN/PW?<br><br>BTW, my Asterisk server is registered to My911ITSP.com<br><br>Thanks</DIV>I have a need to do a similar thing; however, I don't have an Asterisk server involved.  The setup I'm working with is as follows:<br><br>Equipment: PAP2 (running SPA1001 3.1.8 firmware) <br>LINE1: Outgoing VoIP line (from a provider w/o any 911/E911 support)<br>LINE2: Incoming VoIP line (from a provider w/ unlimited incoming, metered outgoing, but w/ true E911 support)<br><br>Normally calls are to go out on LINE1; however, for 911 calls, they MUST go out on LINE2.  I've been playing around w/ this all day and have thus far come up w/ a tolerable (athough not ideal) solution.  To describe the solution, I'll replace "911" with "<B>555</B>" (so that you can configure the solution and see how it works):<br><br>1) In the SPA1001's PHONE tab, configure the "Line 2 Select Code" to be "<B>555</B>" instead of "#"<br><br>2) In the LINE1 tab, add the following to the beginning of the Dialing Plan: "<B>555S0|</B>" (this will ensure that 555 gets detected immediately on the LINE1 side)<br><br>3) On the LINE2 tab, set the entire dialing plan to be a HOTLINE:<br><div class="code"><PRE><span class="codetext">(S1 &lt;:18005551212&gt;)</SPAN></PRE></DIV>The above will automatically dial 1-800-555-1212 after 1 second of LINE2 dial-tone.  (Once you're ready to configure this "for real," replace "18005551212" with "911").<br><br>The "S1" condition above is why this solution is not fool proof.  A true "HOTLINE" dialing plan starts with "S0" (for 0 second pause); however, when the adapter's default line (for dialing out) is LINE1, and when switching over to LINE2 (via "555" or "#" or whatever), the "S0" ZERO SECOND PAUSE doesn't get honored.  I don't know why that is, but that's one flaw w/ this implementation.<br><br>Another flaw is that this implementation doesn't allow LINE2 to be used as a backup VoIP provider for other outgoing calls (unless you add more to the LINE2 dialing plan and can dial your other # within 1 second).<br><br>I'm hopeful a better solution can be reached; however, I haven't been successful in otherwise getting a LINE1 dialing plan instruction to dial out on LINE2.<br><br>Rizzo<br>]]></description>
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<pubDate>Tue, 14 Mar 2006 02:16:29 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15664687</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : <B>A. Trading Bandwidth for Better Sound Quality<br>B. Changing Your Jitter Buffer Setting</B><br><br>Draco asked me to add this trick to the list. Apparently all Sipuras ships with an RTP Packet Size set to 0.030 (i.e. 30ms) by default, or 33 packets per second (pps). This of course causes some degree of added latency and issues revolving around lost or discarded packets leading to greater quality loss than if a shorter interval is used.<br><br>It seems most sites in the internet recommend, or at least treat as standard, that you should usually only set the RTP Packet Size to 30ms when you use G.723.1, and to 20ms (i.e. 50 pps) when you pretty much use anything else the Sipura presently supports. 10ms (i.e. 100 pps) is also an option for the highest quality sound with the lowest degree of latency (especially when combined with a jitter buffer setting of low) when using G.711 or G.729 but at the cost of utilizing the highest bandwidth possible.<br><br>At 10ms (highest quality) / 20ms (generally recommended setting) / 30ms (Sipura default; required for G.723.1), the overall bandwidth average for ____ is:<br>G.711 - 126kbps / 95.2kbps / 84.7kbps<br>G.729ab - 70.4kbps / 39.2kbps / 28.7kbps<br><br>A full interactive chart with these codecs and several others can be found here so you can find the right balance between codec and sample period to suit your uplink capacity.<br><br>Your network jitter setting is another area to address in your attempt to reduce latency and improve audio quality. The following tip may have undesirable consequences on certain connections at certain congested times, but overall my experience with it has been very positive.<br><br>Draco offers the following summary of what the Sipura jitter settings means:<br><BLOCKQUOTE>NOTE: If I understand things correctly, the "Network Jitter Level" setting doesn't control the size of the jitter buffer (which is dynamically figured out by the Sipura), but rather controls how quickly the Sipura decides it has good jitter (and thereby lowers the jitter buffer). By setting this setting to "low", you tell the Sipura to quickly assume that a low jitter is going to continue to remain. But the problem is, if the Sipura too quickly makes this assumption (and you do run into jitter), than you will get a "drop out" (or other sound artifact) as a result of the jitter. By leaving it at "high", the Sipura waits a longer time to make sure the lower jitter level really is constant, before setting the buffer to that lower level.</BLOCKQUOTE><br><br>This is a good summary for what the setting does. I suggest you try on your connection changing it from "high" and setting it to "low". You will instantly notice a decrease in latency in echo tests (and when combined with the above 'trick' of setting the codec sample rate to 10 milliseconds will come very close to eliminating VOIP latency, echo, and cross-talking where you interrupt each other because of lag).  Keeping the jitter buffer at the default of high is in my experience 'too conservative' a setting for a decent broadband connection and adds extra useless latency for no speech quality improvement. Normally, most phone calls have no/little jitter as is. And normally when there is bad jitter & packet discard issues, the cause is some program like bit torrent swamping your connection and destroying your pings to everywhere. And setting the buffer even to "extremely high" will never overcome the lag introduced by a computer on your LAN running bit torrent or a multiplayer game. A decent connection (and no bit torrent, etc.) will normally hardly have any jitter anyway and so the Sipura default for jitter correction provides unwanted dose of noticeable lag to all your conversations.<br><br>I hope you find these tricks useful.]]></description>
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<pubDate>Sat, 11 Mar 2006 20:26:47 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15658873</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  p2pvoice <A HREF="/useremail/u/892472"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>Will this "trick" (inspired by your SIPPhone post) work?<br><br><div class="code"><PRE><span class="codetext">911 S0 &lt;:@My911ITSP.com&gt;</SPAN></PRE></DIV><br>I don't know where to put UN/PW?</DIV><B>In order to use an alternate (not your "registered") VoIP account for 911 (only), you would need one of the following options:</B><br><br>A) Have a VoIP provider that doesn't require you to supply a userid/password.  This is highly unlikely for 911, as how would that provider be able to identify you to the 911 folks if you don't log in.<br><br>B) Or arrange to have the userid and password for "My911ITSP.com" be the same as the userid/password that you are using for the VoIP provider on the line.  This may be possible, if "My911ITSP.com" allows you to pick your userid and password.  In that case, just make sure you pick the same userid/password as your main VoIP provider is using (for that line), and when when you redirect to "My911ITSP.com" (using your above dial plan trick) it will properly authenticate, because redirecting to an alternate proxy will still use the registered VoIP's userid and password.<br><br>C) Or if you have an SPA-3000 (but not the other model Sipuras), you could put "My911ITSP.com" onto a "gateway" slot, and then modify your dial plan to dial via that gateway.  For example, if the login credentials for "My911ITSP.com" were on "gateway 4", you could force all 911 calls out via that provider by using:<br><div class="code"><PRE><span class="codetext">911 S0 &lt;:@GW1&gt;</SPAN></PRE></DIV><br><B>Warning:</B>  <br>I do NOT have a VoIP company with true 911, and I wouldn't want to test this even if I did.  So while all this should work "in theory", it has not been tested by me...<br>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15658873</guid>
<pubDate>Fri, 10 Mar 2006 21:45:07 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15650914</link>
<description><![CDATA[<A HREF="/useremail/u/892472"><b>p2pvoice</b></A> : In my previuos post, "cut and paste" didn't work properly.<br>Here is the corrected section:<br><br>Will this "trick" (inspired by your SIPPhone post) work?<br><br><div class="code"><PRE><span class="codetext">911 S0 &lt;:@My911ITSP.com&gt;</SPAN></PRE></DIV><br>I don't know where to put UN/PW?<br><br>Thanks<br>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15650914</guid>
<pubDate>Thu, 09 Mar 2006 19:32:38 EDT</pubDate>
</item>

<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15650867</link>
<description><![CDATA[<A HREF="/useremail/u/892472"><b>p2pvoice</b></A> : DracoFelis:<br><br>I have SPA1001 with two virtual lines. Line 1 is set up for "home" and Line 2 is set up for "business" with a different ringtone. Each line is an extension on my Asterisk server. So, by using a single phone, I can tell from the ringtone, whether an incoming call is a "home" call or a "business" call. I use primarily Line 1 for outgoing calls using Asterisk's outbound routing.<br><br>Here is the "trick: I need to implement:<br><br>My business number is an IPKall number, which "points" to my "business" extension on my Asterisk server.<br><br>My home number is "ported" to an ITSP and it "points" my "home" extension.<br><br>This ITSP is fully 911 compliant; however, I use this ITSP for incoming calls only (don't use this ITSP for outgoing calls). Since this ITSP is fully 911 compliant, I would like only the 911 call to go through this ITSP instead of the other ITSPs configured in my Asterisk's outbound routing.<br><br>Will this "trick" (inspired by your SIPPhone post) work?<br><br>911 <br><br>I don't know where to put UN/PW?<br><br>BTW, my Asterisk server is registered to My911ITSP.com<br><br>Thanks<br><br> ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15650867</guid>
<pubDate>Thu, 09 Mar 2006 19:24:51 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15497374</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by Pelikano :</SMALL><BR><BR>Enable IP Dialing: yes<br></DIV>If you are going to suggest a setting, please also include a short description as to why you want to use that setting.  Otherwise, nobody knows the context of your "trick".<br><br>And FWIW, in my experience you rarely want to turn on "Enable IP Dialing:", as that setting rarely does any good, and can in fact cause HARM with advanced "dial plans"!<br><br>Remember, the ONLY thing that setting does is allow you to use the *-key to MANUALLY key in a call (from your "phone") by IP ADDRESS (not DNS address, only the full NUMERIC IP ADDRESS).  However, turning that setting on does have the potential to interfere with normal dialing (as setup by your dial plan).  <br><br>Furthermore, even with that setting OFF, you can still dial any SIP URI (including numeric IP address based SIP URIs) of your choice by either of the following two methods:<br><br>1) Put the SIP URI in one of the "Speed Dial" slots, and then just use the "speed dial" when you want to call that address.  This is perhaps the easiest way, but it is limited to your 8 "speed dial" slots.  Furthermore there was at least one version of the Sipura firmware that had a "bug" with this feature.  However, I believe that bug is fixed in the latest 3.x firmware from Sipura (and it never was present in the older 2.x firmwares).  So with most firmware versions, this approach works fine (and I do in fact use that approach to dial a small number of direct SIP URIs).<br><br>and/or 2) Modify your "dial plan" to let you dial any SIP URIs you are interested in calling.  Remember, the flexibility of the dial plan syntax, makes adding SIP URI "translations" pretty easy.  For example, below is a dial plan piece/translation that lets you directly call people on the SIPphone (Gizmo project), even when you don't use their service:<br><div class="code"><PRE><span class="codetext">1 747 xxx xxxx S0 &lt;:@proxy01.sipphone.com&gt; | </SPAN></PRE></DIV><br>And remember that BOTH of the above two approaches can be used with the same adapter (i.e. you can use "speed dials" and/or "dial plan changes" to call the SIP URIs of your choice).  And neither of the above two approaches needs (or benefits from) turning on "Enable IP Dialing".  In fact, I routinely use both approaches with "Enable IP Dialing: <B>no</B>"!<br>]]></description>
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<pubDate>Fri, 17 Feb 2006 09:43:24 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15414953</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Enable IP Dialing: yes<br><br>;)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15414953</guid>
<pubDate>Mon, 06 Feb 2006 13:51:35 EDT</pubDate>
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<title>Re: Sipura Remote Config over a Slow ADSL Link</title>
<link>http://www.dslreports.com/forum/remark,15328316</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  meirgreen <A HREF="/useremail/u/979651"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR><div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</SMALL><BR><BR>If you are trying to access the Sipura (admin) web pages remotely, you may trip over a Sipura bug.  Apparently, the "web server" on a Sipura adapter will try to keep sending page data until it is "acknowledged" by the remote web browser, even AFTER the Sipura exceeds it's internal memory dedicated to web page buffering.  :(<br>...<br>According to the forums over on Voxilla, you can work around this bug by limiting your web page buffer (on the computer you are accessing the Sipura from) to around 20K....<br></DIV>How do I limit the web page buffer?</DIV>Here's the original thread over on Voxilla.com about this:<br>&raquo;<A HREF="http://voxilla.com/PNphpBB2-viewtopic-t-4131.html" >voxilla.com/PNphpBB2-viewtopic-t-4131.html</A><br><br>So it's not the "web buffer" per se, but rather it looks like the TCP receive buffer/window for the OS you are using.  While I'm am not sure exactly how this translates from unix (which the poster with the suggestion was using) to Windows, I do have an educated guess.<br><br>What I would try on a Windows machine (to see if it fixes the problem for you) is to download &raquo;<A HREF="/drtcp">/drtcp</A> from this board, and then use it to set your "TCP Receive Window" to say 16384 (a nice round number below 20k), and then reboot your computer (so that the new TCP settings take effect).  If all goes well, that may be all you need to do to prevent the Sipura from overloading it's buffer (when accessed from that Windows machine).  Give it a try, and let us know if it helps (I can't personal try it, as I don't have the address for any Sipura that is "timing out" on it's Admin Page load)...]]></description>
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<pubDate>Wed, 25 Jan 2006 17:33:05 EDT</pubDate>
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<title>Re: Sipura Remote Config over a Slow ADSL Link</title>
<link>http://www.dslreports.com/forum/remark,15327645</link>
<description><![CDATA[<A HREF="/useremail/u/979651"><b>meirgreen</b></A> : <div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>If you are trying to access the Sipura (admin) web pages remotely, you may trip over a Sipura bug.  Apparently, the "web server" on a Sipura adapter will try to keep sending page data until it is "acknowledged" by the remote web browser, even AFTER the Sipura exceeds it's internal memory dedicated to web page buffering.  :(<br>...<br>According to the forums over on Voxilla, you can work around this bug by limiting your web page buffer (on the computer you are accessing the Sipura from) to around 20K....<br></DIV>How do I limit the web page buffer? Neither IE nor Firefox work (even with retries) with a remote SPA3000 over a slow (96 kbps upload) ADSL link. Someone wants me to remote configure their Sipura - emergency priority - and the advanced admin page doesn't finish loading. I couldn't find any information in google or voxilla. Could you please post a link to the solution?<br><br>Please help!<br><br>Thanks,<br>Meir<br>NoSpam20060125@levtov.com]]></description>
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<pubDate>Wed, 25 Jan 2006 16:18:46 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15314119</link>
<description><![CDATA[<A HREF="/useremail/u/1305204"><b>t_rajan_m</b></A> : Thanks for the forwarding post. This is very useful. I just wanted to know whether this forwarding works in SPA 3000?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15314119</guid>
<pubDate>Mon, 23 Jan 2006 22:32:09 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15249182</link>
<description><![CDATA[<A HREF="/useremail/u/1314123"><b>Voipnut</b></A> : Anyone know if a Sipura 2000 will accept the 2002 firmware and thereafter function as a 2002? Or does it break it ?? It would be neat to be able to upgrade like this.]]></description>
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<pubDate>Sun, 15 Jan 2006 06:38:54 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15190722</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Michigan,<br>I am useing Linksys Pap2 (10 device * 2 line), i am haveing a serious problem.<br><br>- When ever any call i make and disconnect the call, it doesnt disconnect in my mobile , its keep ringing in my mobile and just go to voice mail.<br><br>- i read your forum <br>off hook warnning tone > 1400@0,2060@0,2450@0,2600@0;30(.1/.1/1+2+3+4)<br>&raquo;<A HREF="http" >www.dslreports.com</A><br>&raquo;<A HREF="http://michigantelephone.mi.org/blog/" >michigantelephone.mi.org/blog/</A><br><br>but it doesnt made any changes. <br><br>could you please guide me , i didnt find any source where i can solve this problem.<br><br>reg]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15190722</guid>
<pubDate>Sat, 07 Jan 2006 03:52:30 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15151415</link>
<description><![CDATA[<A HREF="/useremail/u/223314"><b>HD_Ride</b></A> : <div class="bquote"><SMALL>said by  HD_Ride <A HREF="/useremail/u/223314"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>I know this has probably been answered several times however I&#146;m overwhelmed with all the configuration settings in the spa3000. <br> </DIV>Disregard I found a PDF over on voxilla.com and it wasn&#146;t as bad as I thought it would be]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15151415</guid>
<pubDate>Mon, 02 Jan 2006 10:44:54 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15136385</link>
<description><![CDATA[<A HREF="/useremail/u/223314"><b>HD_Ride</b></A> : I know this has probably been answered several times however I&#146;m overwhelmed with all the configuration settings in the spa3000. here goes, I have my own unlocked spa3000 that I configured yesterday for inphonex &#147;pay as you go&#148; plan and it&#146;s up and running.<br>Now I would like to add another &#147;pay as you go&#148; to my spa3000 however I&#146;m not sure how to do it, in fact I don&#146;t know where to start?  As most of you know (and I just realized) the spa2000 has a line2 tab but the spa3000 is configured with additional VoIP lines differently. I guess my question is, does a really simple &#147;how to&#148; step-by-step idiot-proof instructions on how to do this exist, like a PDF tutorial?  also, if there are two lines configured on the device how do you select or toggle between lines?  Do you need a line1/line2 splitter for the spa3000? Obviously confused, thanks in advance]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15136385</guid>
<pubDate>Fri, 30 Dec 2005 22:55:14 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15078914</link>
<description><![CDATA[<A HREF="/useremail/u/1304723"><b>Lefteris9</b></A> : Hi,<br><br>As we saw here thanks to DracoFelis, we can forward incoming calls from one line to the other, but how about forwarding outgoing calls?!?<br><br>I have a Linksys Pap2 and I'm trying to set up a dial plan which will allow me to place outgoing calls to both lines (Line1 configured to sipdiscount and Line2 configured to a Greek Voip provider).<br>What I'm basically trying to do is set up the dial plan so that if I dial #9 then the Phone number, the Phone is placed through Line2, while if I disal a number without #9 the Phone is dialed from Line1.<br><br>I understand that this is easy to set up in the Sipura 3000 because it supports gateways, but is it possible to add support for this to the Linksys pap2 (i.e. Sipura 2100)?<br><br>So far I was not able to do this, so I'm seeking for help....]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15078914</guid>
<pubDate>Thu, 22 Dec 2005 08:10:58 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15072621</link>
<description><![CDATA[<A HREF="/useremail/u/1304414"><b>Bollie Bolst</b></A> : @dracefelis<br><br>Forwarding is working, at least the other phone is ringing.. but I don't hear either myself talk or the other person.. What can be wrong..<br><br>I use the Sipura 2002]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15072621</guid>
<pubDate>Wed, 21 Dec 2005 12:58:19 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15062520</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : The info on forwarding 1 line to another didn't work for me.  I have a WRT54GS router, and I am running DD-WRT (12/16/05 v23 Beta 2 version) with SIPatH.  I am able to ring both Line 1 and Line 2 from an outside line (mobile phone), if I don't have call forwarding.  But if I try to forward the call as per the instructions, I just get a blank tone.  I have tried using the IP of the router (both internal and external), the IP of the SPA2k, and 127.0.0.1.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15062520</guid>
<pubDate>Tue, 20 Dec 2005 08:42:10 EDT</pubDate>
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<title>help me!!!</title>
<link>http://www.dslreports.com/forum/remark,15028088</link>
<description><![CDATA[<A HREF="/useremail/u/1302071"><b>leiba2652</b></A> : i'm trying to set up a network so i can access it remotely - i.e. from another computer on the internet - and have no clue what to do - what ports to open if any, firewall's disabled, etc. <br><br>i'm using a vonage/linksys wrtp54g router.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15028088</guid>
<pubDate>Thu, 15 Dec 2005 12:11:48 EDT</pubDate>
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<title>Re: Inbound calls DIRECTLY to your adapter!</title>
<link>http://www.dslreports.com/forum/remark,15022836</link>
<description><![CDATA[<A HREF="/useremail/u/1223857"><b>BigMatza</b></A> : Thanks... Sorry about that.  I moved my post to &raquo;<A HREF="/forum/remark,15022809">[Other] One-way audio problem for Direct-to-PAP2</A><br><br>I also updated my results and included some screenshots for you.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15022836</guid>
<pubDate>Wed, 14 Dec 2005 18:17:23 EDT</pubDate>
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<title>Re: Inbound calls DIRECTLY to your adapter!</title>
<link>http://www.dslreports.com/forum/remark,15022162</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  BigMatza <A HREF="/useremail/u/1223857"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>1. When dialing a FWD PSTN-to-FWD gateway, using "option 2", and entering my 6-digit FWD account number, the connection is fine.<br>2. When dialing my IPKall number DIRECTLY from the same phone I just tried the FWD gateway method from, the IPKall phone rings and picks up.  The IPKall phone can receive audio from the caller, but the caller cannot receive audio from the IPKall phone.</DIV>Especially since you have tested this in your router's DMZ, the problem is most likely in your adapter's settings (instead of your router's setup).  Remember, the direct into your Sipura (or in your case a PAP2) trick is VERY SENSITIVE to the exact Sipura settings (much more so than FWD is, because FWD has the "registration" to help it, whereas the "direct into your adapter" trick has to do everything itself)!  <br><br>In particular, you need to NOT be using an "outbound proxy" (if you are, the outbound will go via the "outbound proxy" producing 1-way audio).  You NEED to use something close to my recommended STUN settings (to properly deal with the NAT of your router).  And for the same reason, you need to have "NAT Mapping Enable: yes".  And don't forget the "Ans Call Without Reg: yes" setting either.<br><br>NOTE:  In my case I ONLY "forward" the main SIP port (letting STUN and the SIP handshaking process open the audio ports), and audio works fine 2-way.  So you shouldn't have to forward every port under the sun, to get this to work.<br><br>If you email your adapter settings (bring up the "advanced" "admin login" page, do a "save as" to mysettings.htm, and then email the mysettings.htm file as an "attachment"), I can verify the setup.<br><br>BTW:  Please start a new thread for questions.  Let's try to keep this (sticky) thread for actual "tricks" posts...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15022162</guid>
<pubDate>Wed, 14 Dec 2005 17:00:37 EDT</pubDate>
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<title>Re: Inbound calls DIRECTLY to your adapter!</title>
<link>http://www.dslreports.com/forum/remark,15021873</link>
<description><![CDATA[<A HREF="/useremail/u/1223857"><b>BigMatza</b></A> : DELETE - Posted my concern in a new thread.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15021873</guid>
<pubDate>Wed, 14 Dec 2005 16:27:20 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15014388</link>
<description><![CDATA[<A HREF="/useremail/u/1150714"><b>mark2</b></A> : <div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR><div class="bquote"><SMALL>said by  mark2 <A HREF="/useremail/u/1150714"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</SMALL><BR><BR>does anyone know the disconnect tone string for israel???</DIV>Voxilla.com has a "regionalization wizard" .....<br> </DIV>the wizard only changes the dialtone, ringer to sound like the one used locally, but doesn't do anything with the disconnect tone. ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15014388</guid>
<pubDate>Tue, 13 Dec 2005 18:29:03 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15014328</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  mark2 <A HREF="/useremail/u/1150714"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>does anyone know the disconnect tone string for israel???</DIV>Voxilla.com has a "regionalization wizard" (web page that helps you setup your Sipura for various parts of the world).  You might give that a try, and see if Israel is included in their list of supported areas/countries...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15014328</guid>
<pubDate>Tue, 13 Dec 2005 18:18:59 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,15013896</link>
<description><![CDATA[<A HREF="/useremail/u/1150714"><b>mark2</b></A> : does anyone know the disconnect tone string for israel??? when i use the pstn voip gateway i can't get the pstn to hang up unless i force it. via **#. any ideas]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,15013896</guid>
<pubDate>Tue, 13 Dec 2005 17:21:20 EDT</pubDate>
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<title>Re: Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14959060</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Sipura does not release tools for provisioning to the end users, only VoIP providers. These tools create a type of a configuration file that can be used to mass deploy/update the units in service.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14959060</guid>
<pubDate>Tue, 06 Dec 2005 11:20:31 EDT</pubDate>
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<item>
<title>Re: Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14958301</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : And what's the way to get provisioning tools?<br>What is the meaning of "provisioning" in sipura's terms?<br>It is a software or what?<br><br>Thanks!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14958301</guid>
<pubDate>Tue, 06 Dec 2005 09:24:57 EDT</pubDate>
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<title>Re: Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14958275</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Unfortunatelly, this is one of the parameters that can only be set through provisioning.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14958275</guid>
<pubDate>Tue, 06 Dec 2005 09:18:33 EDT</pubDate>
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<title>Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14957389</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Hi there.<br><br>Please could you help me to protect my sipura-3000 from being factory reset via IVR menu?<br>I mean how I can set the value Protect_IVR_Factoryreset to "YES".<br><br>Thank you!]]></description>
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<pubDate>Tue, 06 Dec 2005 03:02:41 EDT</pubDate>
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<title>Inbound calls DIRECTLY to your adapter!</title>
<link>http://www.dslreports.com/forum/remark,14876886</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : If you are feeling really feisty, I have figured out how to accept inbound SIP URI calls directly to a Sipura (bypassing all VoIP providers), even if/when you have the line "registered" with some other VoIP provider.  And with STUN and dynamic DNS, you can do this from behind a NAT router, with a dynamic IP address.  I documented the procedure on this SIP Broker Wiki page:<br>&raquo;<A HREF="http://faq.sipbroker.com/tiki-index.php?page=Inbound%20Calls%20Directly%20to%20your%20LinkSys%20or%20Sipura" >faq.sipbroker.com/tiki-index.php&middot;&middot;&middot;20Sipura</A><br><br>Oh yeah, and if you haven't checked out the free SIP Broker VoIP service ( &raquo;<A HREF="http://sipbroker.com" >sipbroker.com</A> ) yet, give it a try.  Not only does it let you call a lot of other VoIP services "for free", they even have a "ENUM service" that will try to lookup a free VoIP address that corresponds to a telco number you would like to call.  Way cool....<br><br>Disclaimer:  <br>I have been volunteering some time to help out SIP Broker.  So I am a little biased about their service.  But I still think their service is "cool", otherwise I wouldn't be helping them.  And hey, even if you don't find them as useful as I do, you can't go too wrong with a price of "free"....]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14876886</guid>
<pubDate>Thu, 24 Nov 2005 22:04:15 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14815682</link>
<description><![CDATA[<A HREF="/useremail/u/1027622"><b>rocku24_7</b></A> : how to well i have broadvoice lite ($6 unlimited incall) on line 1 and sipdiscount.com (unlimeted outcall US +) on PSTN gateway, the voxilla wizard also put sipD on GW 1 on the line 1 page. i craft a plan that send all dial calls from handset on line1 to call @gw1, when call is place it ring ok but disconect 5sec after called party picks up. the PSTN----VOIP call works fine (remember this is the same voip provide on gw1 on line that giving truble) why is it working well on pstn page but not online1? is there another way to do this all suggestions are welcome <div class="borderless"><TABLE WIDTH=95% align=center border=0 CELLPADDING=4"><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap WIDTH=33%><A HREF="/r0/download/924335~59dc542ca51c12a73a27fa3230fcebc6/Sipura_SPA_Config@BV%26sipdis.zip"><IMG  align=absmiddle TITLE="download" SRC="http://i.dslr.net/silk/compress.png" border=0 width=16 height=16><IMG SRC="http://i.dslr.net/1ptrans.gif" WIDTH=10 HEIGHT=1 border=0><big>Sipura_SPA_C&middot;&middot;&middot;pdis.zip</big></A> <small>16,934 bytes</small><br><small>(Sipura_SPA_Config@BV&sipdis.htm)</small></TD></TABLE></div>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14815682</guid>
<pubDate>Wed, 16 Nov 2005 02:17:12 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14790410</link>
<description><![CDATA[<A HREF="/useremail/u/1267578"><b>loosenutvt</b></A> : Hi DracoFeilis,<br>I have spa-3000, howto I setup both sipphone( for in-coming) and sipdiscount (for out-going) on line1. I have followed the instruction but can't get it to work. I can't dial out via sipdiscount. What is the dialplan looks like? <br>Thanks]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14790410</guid>
<pubDate>Sat, 12 Nov 2005 15:01:49 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14742423</link>
<description><![CDATA[<A HREF="/useremail/u/1285783"><b>Stubri</b></A> : Has any tried connecting the phone and line jacks of a Sipura 3000 with a Y cable and was it successful if so did the Line1 handset ring with a call coming on VOIP2? How was the PTSN side of the 3000 setup?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14742423</guid>
<pubDate>Sun, 06 Nov 2005 01:29:09 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14670613</link>
<description><![CDATA[<A HREF="/useremail/u/151200"><b>druber</b></A> : Don't know how well-known this is, but I did find out that by default VoicePulse enables ringthrough from the FXS port to the FXO port on a SPA-3000, so if someone calls my POTS number, I don't have to worry about missing the call (and don't need to forward it to the VP number, either.)  This way, I can set my network unavailable number on VP to the POTS number.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14670613</guid>
<pubDate>Thu, 27 Oct 2005 12:58:58 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14588825</link>
<description><![CDATA[<A HREF="/useremail/u/941095"><b>david_gruenb</b></A> : Thanks Erwin_D for the reboot tip.  saves me from having to get up every time I make a change.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14588825</guid>
<pubDate>Sun, 16 Oct 2005 02:01:07 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14584562</link>
<description><![CDATA[<A HREF="/useremail/u/833931"><b>Erwin_D</b></A> : <div class="bquote"><SMALL>said by  voipfima <A HREF="/useremail/u/818989"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>Is it true that Sipura 1001 silence suppression is only for outgoing voice?</DIV>Yes. That option only controls wether your own device will suppress <I>your</I> silence. If you hear cut-outs, it means the <I>other</I> party has silence suppression enabled on their device.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14584562</guid>
<pubDate>Sat, 15 Oct 2005 13:11:48 EDT</pubDate>
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<title>How to reboot your Sipura?</title>
<link>http://www.dslreports.com/forum/remark,14584508</link>
<description><![CDATA[<A HREF="/useremail/u/833931"><b>Erwin_D</b></A> : As some have pointed out, making changes to your Sipura settings sometimes requires a reboot or powercycle. (Personally, I never have to do this on my 2002 with 3.1.5.) However, powercycling the device everytime you change your settings may not be the haelthiest thing for the powersupply... or the Sipura!<br><br>There is no "Reboot" button in the configuration page, but you can reboot your Sipura by entering the following URL:<br><br><div class="code"><PRE><span class="codetext">http://&#91;your-Sipura-IP&#93;/admin/reboot</SPAN></PRE></DIV><br>And your Sipura will powercycle itself!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14584508</guid>
<pubDate>Sat, 15 Oct 2005 13:02:39 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14501214</link>
<description><![CDATA[<A HREF="/useremail/u/818989"><b>voipfima</b></A> : Is it true that Sipura 1001 silence suppression is only for outgoing voice? I signed up with GalaxyVoice and disabled silence suppresison. However it still seems to be on for incoming calls. Low volume music cuts off, low volume conversation cuts off as well. Their tech support claims that silence suppression is off server side, but it surelly doesn't sound like it...  I setup a FWD service on second line and it sounds great with the same Voice parameters. Any help will be appreciated.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14501214</guid>
<pubDate>Tue, 04 Oct 2005 10:04:49 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14498484</link>
<description><![CDATA[<A HREF="/useremail/u/448156"><b>ArgMeMatey</b></A> : New 3.1.7c firmware for the SPA 3000 can reportedly now play a warning tone when calls are being routed to the PSTN port.  There are other enhancements and bug fixes, too, and users at Voxilla have reported that the Line 1 to PSTN echo problem from 3.1.5 appears to be fixed, too.  <br><br>See the release notes at &raquo;<A HREF="http://www.sipura.com/Documents/rnote/rn3k-3.1.7c.htm" >www.sipura.com/Documents/rnote/r&middot;&middot;&middot;1.7c.htm</A><br><br>Edit:  PSTN warning tone toggle is located on the PSTN Line Tab under Audio Configuration, and despite what the release notes say, it is labeled "Warn Outgoing PSTN Call".]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14498484</guid>
<pubDate>Mon, 03 Oct 2005 22:08:39 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14487112</link>
<description><![CDATA[<A HREF="/useremail/u/1045297"><b>davidlear</b></A> : "Every voipbuster id requires at least one letter"<br>This is true BUT possibly the valid number  could also include A,B,C or D because international number formats using MF4 keypads are comprised of the digits 0-9,ABCD*# - this is just a thought BUT the statement is true. <br>This means you register a voipbuster account which includes A,B,C or D and if Sipura has stuck to the correct international format for keypads then this ought to do the trick - if not then perhaps a suggestion to Sipuras technical support that their dialplan law is not accurately reflecting standards may result in an update in a future firmware release, they may not these days wish to do it to assist people tinkering with the boxes and possibly upsetting providers BUT in principle they are neglecting standards if they fight it or refuse. So a softly, softly politically correct approach may be the appropriate move. I do hope this assists. BTW I am at Astricon, Ca. next week - any one going ?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14487112</guid>
<pubDate>Sun, 02 Oct 2005 09:44:10 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14486590</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Saving the Web Page as.... works in Firefox very well.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14486590</guid>
<pubDate>Sun, 02 Oct 2005 03:48:00 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14442777</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  YOUR_UGLY_VT <A HREF="/useremail/u/484104"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>      :</SMALL><BR><BR>I have a sipura SPA-2002 and an NCH Music on Hold Server running on my computer. How can I configure it to play the music when i put someone on a flash-hook hold? </DIV>To my knowledge, the Sipuras can't do this directly.  However, I see two options:<br><br>1) If your "music on hold" server is designed to hook between a real phone line and a phone, then it should also work to hook it between a Sipura and a phone (since the Sipuras mimic real phone lines).<br><br>2) Or you could have the Sipuras register to an IP based PBX (such as *), instead of doing VoIP directly with the outside world.  If you did that, you could use the "music on hold" feature of that PBX.<br><br><B>Note to everyone:</B><br>  <br>I started this thread to have somewhere to post <B>SIPURA TRICKS YOU KNOW HOW TO DO</B>.  While I realize the temptation to post where people seem to know what they are talking about, we risk having the questions drown out the actual posts about "tricks".  And with this thread already over 100 messages (and growing), it's already becoming hard to find the details for the "tricks" out there.<br><br><B>So by all means post your "TRICKS" here.  But PLEASE start your own message thread if you want to ask questions!  Thanks!</B>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14442777</guid>
<pubDate>Sun, 25 Sep 2005 22:57:00 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14442377</link>
<description><![CDATA[<A HREF="/useremail/u/484104"><b>YOUR_UGLY_VT</b></A> : I have a sipura SPA-2002 and an NCH Music on Hold Server running on my computer. How can I configure it to play the music when i put someone on a flash-hook hold?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14442377</guid>
<pubDate>Sun, 25 Sep 2005 21:53:57 EDT</pubDate>
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<title>Configuration of SPA-3000 for gateways</title>
<link>http://www.dslreports.com/forum/remark,14405125</link>
<description><![CDATA[<A HREF="/useremail/u/901172"><b>will792</b></A> : I am trying to configure FWD as an additional outgoing gateway on SPA-3000. The main VOIP provider is VoicePulse. They lock almost all configuration, dial plan, Line 1 tab and so on. The only fields that they enabled for me (upon a request) are Gateway 1 through 4. They told me to put in gateway information in the following format fwd_user_id;fwd_password@fwd.pulver.com and use 101#, 102# ... to select outgoing gateway.<br><br>This did not work so after checking with a sipura manual I changed syntax to @fwd.pulver.com;uid=my_fwd_number;pwd=my_fwd_password .<br><br>The result is the same; fast busy signal after dialing a number, i.e. 102#613 (for gateway 2, Echo test for FWD).<br><br>Any ideas what I am missing? Maybe it is not possible to configure gateway accounts with "Gateway 1..4" fields only?<br><br>My firware version is 2.0.10(GWc) and it cannot be change (VP locks it).<br><br>Thank you in advance for any information.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14405125</guid>
<pubDate>Tue, 20 Sep 2005 14:49:01 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14402491</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : I have a Sipura SPA-2002, however I believe that other Sipura adapters (PAP2, SPA-1000, SPA-10001, SPA-2000, SPA-831 & SPA-3000) all have a simular "User1" (and User2, etc.) tab(s).<br><br>If your ATA was provisioned by a VoIP provider typically they don't lock down this tab.  My questions are:<br><br>1. Call Forward Settings - What is this? Is there anything special I can put in here.  i.e.: make this line selective ring other SIP devices, other SIP devices on other VoIP networks via IP dialing, etc.<br><br>2. Selective Call Forward Settings - Again, what is this?  Can I specify if I get a call from 212-555-5555 automatically dump to voicemail.  Or call forward that caller elsewhere... could be of use at a known local payphone's number.  If I call my adapter's associated phone number I can have it forward out via VoIP to a normally long distance number." <br><br>3. Distinctive Ring Settings - What is the correct syntax for this option?  What exactly does this do?<br><br>The Sipura user tabs aren't really documented very well at all from what is on Sipura's website.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14402491</guid>
<pubDate>Tue, 20 Sep 2005 04:58:20 EDT</pubDate>
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<title>Using &#x22;SIP Broker&#x22; with your Sipura adapter.</title>
<link>http://www.dslreports.com/forum/remark,14387469</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : Just today, I signed up with another useful free service, you can configure your Sipura adapters to use.  And like my previous FWD "trick", you can sometimes use this service even if/when you already have the Sipura configured for another VoIP service!  <br><br>The service is known as "SIP Broker", and is available as a free signup from &raquo;<A HREF="http://www.sipbroker.com/" >www.sipbroker.com/</A> .  And like FWD, this service has a LOT of "peering numbers" (&raquo;<A HREF="http://www.sipbroker.com/sipbroker/action/providerWhitePages" >www.sipbroker.com/sipbroker/acti&middot;&middot;&middot;itePages</A> ).<br><br>To use this service, you will need a free account from them.  Because (unlike FWD) they will only let you use their proxy (for making outbound calls) when you have a free account with them (whereas FWD lets anyone use their proxy for making outbound calls).  The "account" you setup with "SIP Broker" is somewhat strange, as you will actually use the SIP URI of an existing VoIP service you can receive calls at.  For example, I signed up with "xxxxxx@fwd.pulver.com" (where xxxxxx was really my 6 digit FWD number), so that inbound "SIP Broker" calls (for me) will forward to my FWD account.  You then pick a password that does NOT have to be the same as the one you used for the VoIP provider you register with "SIP broker" with (you will only use the SIP Broker password to get into the SIP broker web portal).<br><br>Now, I agree with the poster (voip247) in the voxilla.com forums, when he posted this thread: &raquo;<A HREF="http://voxilla.com/PNphpBB2-viewtopic-t-5033.html" >voxilla.com/PNphpBB2-viewtopic-t-5033.html</A> .  You should <B>IGNORE</B> "SIP Broker's" recommended setup (that uses an "outbound proxy"), as it is a real PITA.  Instead, here are two "friendly" ways to setup a Sipura adapter to use SIP Broker:<br><br>1) The way that the poster recommended (in this thread: &raquo;<A HREF="http://voxilla.com/PNphpBB2-viewtopic-t-5033.html" >voxilla.com/PNphpBB2-viewtopic-t-5033.html</A> ), was to make sure you have the Sipura line registered with the SAME provider (and account) as the one you told "SIP Broker" about (when you signed up).  If you do that, than you can simply use a custom dial plan piece to call out via "SIP Broker".  For example, the following dial plan will let you make any SIP Broker call by dialing "# 7 SIP_broker_digits #".  Again this will ONLY work if/when the registered account on the line, matches the "USER_ID" you gave "SIP Broker" when you first opened your SIP Broker account.<br><br><div class="code"><PRE><span class="codetext">&lt;#7:&gt;&#91;x*&#93;.&lt;#:&gt;S0 &lt;:@sipbroker.com&gt; </SPAN></PRE></DIV>NOTE: I've tested this 1st approach, and it does seem to work with my SPA-3000.  And voip247 (on the voxilla.com forums) also says it works with other Sipura models as well.  So this is probably the preferred way to setup your Sipura with "SIP Broker", as long as you have the line "registered" with the same account/provider you gave SIP Broker when you setup your SIP Broker account!  In fact, even with an SPA-3000 this may be "the preferred way", as it doesn't "waste" one of your 4 "gateway" VoIP slots!<br><br>NOTE:  In this 1st case, SIP Broker is getting your SIP username and password (for your primary VoIP account, you registered with "SIP Broker") directly from your Sipura (by the fact that you have that provider "registered" with your adapter).  That's why the account IDs have to match, for this "trick" to work with "SIP Broker"!<br><br>2) Or if you have an SPA-3000, you can also setup "SIP Broker" on an SPA-3000 "gateway" field (even when your "SIP Broker" account does NOT match your primary "Line 1" registered provider).  To do this, setup your SPA-3000 similar to this FAQ from "SIP Broker": &raquo;<A HREF="http://www.sipbroker.com/sipbroker/action/static?itemOID=40066" >www.sipbroker.com/sipbroker/acti&middot;&middot;&middot;ID=40066</A><br>Once that is done, you can use a dial plan like the following, to again allow "# 7 any_digits #" to be dialed.<br><div class="code"><PRE><span class="codetext">&lt;#7:&gt;&#91;x*&#93;.&lt;#:&gt;S0 &lt;:@GW1&gt;</SPAN></PRE></DIV><br>NOTE: Again, you do NOT have to have your SIP Broker password match the password your VoIP provider (that you use with SIP Broker) uses!  And in this 2nd case of the SPA-3000 setup (using one of the "gateway" fields), it doesn't even appear to matter what you put into the "GWx Password:" field, except when you use SIP Broker "peering numbers" to call your own provider.  i.e. when I put a bogus password into my "GW4 Password:" (I was using SIP Broker or my 4th gateway entry), I could call other peering numbers via SIP Broker, but got a "busy signal" when trying "#7393 fwd_number #" to indirectly call FWD numbers (i.e. my own provider) from SIP Broker.  But I could still make other "peering calls" (i.e. a provider other than my own provider) via that arrangement.  For example "#7 258 9123 #" correctly called the UK "time number" via "SIP Broker", even with my "GW4 Password:" field did NOT match my FWD password (or my "SIP Broker" password either)!<br><br><B>[EDIT] Breaking news!</B><br>Shortly after I posted my setup, I ran across an announcement in the "SIP Broker" forums.  Apparently just TODAY (9/17/2005), "SIP Broker" decided to open their proxy to the world.  So you no longer have to sign up with one of their free accounts, to use them as an outbound VoIP provider on your Sipura!  <br><br>Instead, you can just use something like the dial plan in technique #1 (of this note), even without a "SIP Broker" account.  You can still sign up with a free "SIP Broker" account if you like, especially if you want to setup an inbound SIP Broker "alias" to ring your phone.  But  according to &raquo;<A HREF="http://www.sipbroker.com/forums/showthread.php?t=82" >www.sipbroker.com/forums/showthread.php?t=82</A> the free account is now OPTIONAL (just as it is with FWD, and their proxy)!]]></description>
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<pubDate>Sat, 17 Sep 2005 22:20:31 EDT</pubDate>
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<title>You may need to &#x22;power cycle&#x22; after making changes</title>
<link>http://www.dslreports.com/forum/remark,14337860</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : Power cycle your Sipura after making changes.  It's sometimes needed.  :(<br><br>I just got done wasting several minutes trying to "debug" a pretty simple change in my "dial plan".  After going over the dial plan with a fine tooth comb, I was unable to find ANY reason why my new entry was just giving me a busy signal when I tried dialing using it.  Obviously something was wrong, but what?  I tried simplifying the entry, routing it to another provider, removing all my entries that "block" various dialing patterns, etc.  And nothing was working (and yes, I was using "Submit All Changes" between each attempt).<br><br>So on a hunch, I tried unplugging the adapter, waiting a few seconds, and plugging it back in.  Guess what?  The entry (that was previously giving me so many "headaches") started working.  Obviously this is a bug with the Sipura (or at least the version of the Sipura firmware I have loaded in my SPA-3000), where the Sipura doesn't always properly "reset" after you make some changes.  But it is nice to know about this "gotcha", because the "fix" is pretty easy if you know the "trick" (just power cycle your Sipura after making changes).  That way, you can be sure that the Sipura "reset" after making your changes, because the power cycling forces a "reset".<br><br>NOTE:  I was able to make other changes after the power cycle, and have the changes work as soon as I pressed "Submit All Changes" (and waited the few seconds for the automatic reset).  And I've also had other occasions (in the past) where the changes were "live" as well.  But it's also clear (from this experience) that the Sipura can get to a point where it won't accept new changes UNTIL you power cycle the adapter.  So by power-cycling after you make changes, you are more assured that the changes really "took"...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14337860</guid>
<pubDate>Sun, 11 Sep 2005 00:02:17 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14259411</link>
<description><![CDATA[<A HREF="/useremail/u/833931"><b>Erwin_D</b></A> : <div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>You might want to rethink that, when you realize that two SIP calls CAN'T be on the same UDP/SIP port at the same time.</DIV>Ah OK, I didn't realize that. I haven't tested multiple calls on the same port, however, calling line 2 from line 1 seemed to work. Anyway, it's back to 5060 and 5061.<br><br><div class="bquote">NOTE:  I don't know if that is your current problem with my "trick"</DIV>Probably not... I testes both methods. It made no difference. But I have given up on trying to forward calls using the Sipura. I 'discovered' (I found an old DSLR post using Google) that FWD has a forwarding feature that works the same way, and used that one to forward to line 2 on my Sipura, with perfect results! Here's the 'secret' link:<br><br>&raquo;<A HREF="http://account.freeworlddialup.com/index_new.php?section_id=114" >account.freeworlddialup.com/inde&middot;&middot;&middot;n_id=114</A><br><br>Be patient when using this; This setting takes up to 10 minutes to kick in. But finally, I can make and receive calls from both providers using only one physical line on my Sipura 2002!]]></description>
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<pubDate>Wed, 31 Aug 2005 05:08:58 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14256684</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  Erwin_D <A HREF="/useremail/u/833931"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>I have found that you don't necessarily need different ports for each account, it only tends to compicate things. I have set both account in my SPA 2002 to port 5060; the Sipura knows which line to ring by looking at the username. Calling the other line works as well.</DIV>Ouch!  <br><br>You might want to rethink that, when you realize that two SIP calls CAN'T be on the same UDP/SIP port at the same time.  That's just a limit of how SIP keeps the data straight.  So what really happens if two calls are supposedly on the same UDP port (for example, when making a 3-way call on a single Sipura line), is that only the FIRST call to connect actually uses that port.  For the other calls, the data actually goes via a 2nd (unused) SIP port, instead of using the port you expect.<br><br>Not only is this "automatic pick a port" more confusing, but it can also make a mess of STUN and "NAT traversal", as you no longer have a static mapping between what you expect for the ports, and what port is actually used.  So you are just buying yourself trouble, by taking that approach!<br><br>NOTE:  I don't know if that is your current problem with my "trick" (as there are many other minor issues/tweaks that could interfere with that trick working as expected), but it is one possible explanation as to why my trick isn't working for you.  Remember, when you forward the FWD line to your "VoIP Buster line", you are (however briefly) invoking both lines.  But since you have an ambiguous relationship as to which port is on which line, you may be "tripping yourself up", with the needed "NAT traversal".  If it were me, I would suggest you leave the VoIP buster line on the default 5060 port (as that is the target of your forwarding/"SIP reinvite"), but put the FWD line on it's own unique port (5064, perhaps?).  That way, you still get the benefit of using the default SIP port for the target of your forwarding and/or "peer to peer calls", but you also keep the FWD line from trying to use/"take over" the default SIP port...<br><br>NOTE:  Due to possible "3-way calls" on a line, I always treat SIP ports as if two of them were used up for each line/device.  For example, I might assign one line to port 5060 (the default) and the 2nd line to port 5062 (skipping port 5061).  The idea is, that if the line with port 5060 has a "3-way call" (i.e. two VoIP calls on that line "at the same time"), it will really be using port 5060 (the port assigned to that line) _AND_ port 5061 (the next unused port, as a result of the 2nd VoIP call)!  So by skipping SIP ports like this, I am already prepared for 3-way calls on a SIP line.]]></description>
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<pubDate>Tue, 30 Aug 2005 20:45:13 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14254063</link>
<description><![CDATA[<A HREF="/useremail/u/833931"><b>Erwin_D</b></A> : I have found that you don't necessarily need different ports for each account, it only tends to compicate things. I have set both account in my SPA 2002 to port 5060; the Sipura knows which line to ring by looking at the username. Calling the other line works as well.<br><br>Reason for setting both accounts to port 5060 is that some clients/servers ignore the port number all together on (re)invites.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14254063</guid>
<pubDate>Tue, 30 Aug 2005 12:42:14 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14253796</link>
<description><![CDATA[<A HREF="/useremail/u/833931"><b>Erwin_D</b></A> : Yes, I had STUN enabled and pointing to stun.fwdnet.net:3478, but 'Send Resp To Src Port' was set to No. I'll try Yes...<br><br>[...]<br><br>It didn't work, still a fast busy.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14253796</guid>
<pubDate>Tue, 30 Aug 2005 12:06:06 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14252566</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  Erwin_D <A HREF="/useremail/u/833931"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>Now all I need is forwarding to line 2 to work.<br><br>Like many, I tried everything. My router is setup correctly, forwarding both 5060, 5061 and all RTP ports to my SPA 2002. I have a fixed external IP and all LAN IP's are fixed as well.</DIV>Do you have STUN setup on the Sipura (bottom of the SIP tab, on the "Advanced" Admin Login)?  If not, try my STUN settings.  <br><br>I originally tested that trick with my setup (which uses STUN), and the more I learn about how that trick works (or doesn't work) for other people, the more I'm getting convinced that a proper STUN setup (to handle the "NAT traversal issues" of translating between your public IP and your LAN IP) may be part of the puzzle.  <br><br>So give my STUN settings (included in the attachment), a try:<div class="borderless"><TABLE WIDTH=95% align=center border=0 CELLPADDING=4"><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/14252566?c=883889&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG class="apic" BORDER=0 TITLE="7430 bytes" WIDTH=600 HEIGHT=123 SRC="/r0/download/883889.thumb600~814bd85e15159f3ae9e7709712647f97/STUN.GIF/thumb.jpg" ALT="Click for full size"></A><br>A useful set of STUN settings</TD></TABLE></div>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14252566</guid>
<pubDate>Tue, 30 Aug 2005 08:44:51 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14252341</link>
<description><![CDATA[<A HREF="/useremail/u/833931"><b>Erwin_D</b></A> : Hello Draco,<br><br>I've also had heaps of trouble geeting my SPA 2002 to work wit two providers, but I'm nearly there! At least I got so far that I can call both providers from the same phone. I have line 1 set to FWD, port 5060, and line 2 to Voipbuster, port 5061. At first I tried calling Voipbuster from line 1 (using &lt;:@sip.voipbuster.com&gt; in line 1 dialplan), but that didn't work. So now My phone is connected to line 2 and can call FWD numbers using &lt;:@fwd.pulver.com&gt; in the dialplan.<br><br>Now all I need is forwarding to line 2 to work.<br><br>Like many, I tried everything. My router is setup correctly, forwarding both 5060, 5061 and all RTP ports to my SPA 2002. I have a fixed external IP and all LAN IP's are fixed as well. I did some testing with X-Lite, but I suspect that FWD is getting mixed up doing this.<br><br>If I register with X-Lite (using port 5062), which is set NOT to send my internal IP, FWD still sends back my computers internal IP, but it *may* be X-Lite doing some translating. Anyway, calling myself from X-Lite has the following results.<br><br>With no forwarding in the SPA, line 1 rings: An Invite message is first sent back to X-Lite, which sends back a Busy (486). Then FWD calls my SPA and it rings.<br><br>With forwarding enabled, I see some funny things... If I forward to anything other then my own external IP, I see X-Lite actually trying to call that IP; FWD sends a "302 temorarely moved" with the forwarded IP. If it is forwarded to my external IP, FWD send back the internal IP of my computer running X-Lite! And of course the call will fail. Again, X-Lite may be translating the contact info, so this is understandable.<br><br>Actually, now that I think of it... At one time I had both lines set to port 5060, and when I forwarded to the SPA's own internal IP, it DID ring! But no luck using my external IP. Also, looking at the X-Lite logs, I could not see the port being forwarded to; the port seemed to be ignored in the reinvite message.<br><br>With no X-Lite running (and letting all contact info expire), forwarding still does not work. I tried calling in from PSTN-to-VoIP gateway, with different results. At first, I got a tape saying "Invaled extension", now all I get is a fast busy. With the forward gone, all works well (it rings line 1).<br><br>The Cfwd All Dest under Line 1 reads: edokter@[83.160.x.x]:5061<br><br>I tried everything else too, but I think this is the proper setting. To exclude my SPA as being the culprit, I'd like to try calling myself using the above format. If that works, then something is wrong with FWD. Can I send you a PM?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14252341</guid>
<pubDate>Tue, 30 Aug 2005 07:32:17 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14239533</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  aup2 <A HREF="/useremail/u/822077"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>  :</SMALL><BR><BR>When using the Sipura (e.g., 2100) to do P2P calling without registering with a Proxy, does the unit use the settings for the particular line from which the call is made, even though the line is not registered?</DIV>I don't have an SPA-2100, and the manual doesn't say.  But in my limited experience/testing of this (on calls between my SPA-2000 and my SPA-3000), the answer appears to be "yes".<br><br>For example, even when calling an "unregistered" Sipura line, I still had to properly match the "userID@sipura_address:port" programmed into the target Sipura, in order for the call to go through.  Likewise, when I forced a lower bandwidth CODEC on one adapter or another, I could hear the lower quality sound the resulted.  So it at least appears that your "line settings" are paid attention to, even if/when you don't have the line "registered" with any VoIP provider.  Among other things, this means that you should put in some credentials (including UserID), even if they are "bogus", and will only be used as part of controlling your P2P call setup...<br><br>NOTE:  If you want to do some P2P VoIP call tests, PM me with your contact info, and a good time to call.  If I'm not too busy at the time, I'll be happy to run a few call tests (against my SPA-3000), to see what happens.<br><br>NOTE:  You appear to be able to get inbound calls BOTH from a "registered" VoIP provider (if you have one), AND an inbound P2P call on the same line (if you set things up properly).  This little fact could possibly be useful, in some setups, as a way to let additional "virtual numbers" also ring the same VoIP "phone"!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14239533</guid>
<pubDate>Sun, 28 Aug 2005 13:57:48 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14238862</link>
<description><![CDATA[<A HREF="/useremail/u/822077"><b>aup2</b></A> : When using the Sipura (e.g., 2100) to do P2P calling without registering with a Proxy, does the unit use the settings for the particular line from which the call is made, even though the line is not registered? I am specifically interested in the preferred codec settings. If this is not the case, is there a way to force this?<br>Thanks.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14238862</guid>
<pubDate>Sun, 28 Aug 2005 12:12:22 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14231142</link>
<description><![CDATA[<A HREF="/useremail/u/1006166"><b>venk25</b></A> : Jakes,  DracoFelis has posted a possible cause for this problem earlier (on page 3 of this thread).  I think it has to do with the Sipura sending a SIP forward/redirect URI (URI of your other line) back to the calling SIP server and the calling server not supporting it.  If the caller is calling using a service that allows this (like FWD), the call should be forwarded to your other line.  <br><br>I haven't tried it and so, haven't seen it work ! Am just saying based on a few of my SIP packet sniffing sessions :)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14231142</guid>
<pubDate>Sat, 27 Aug 2005 02:23:48 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14230220</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  JakesOnline <A HREF="/useremail/u/1243389"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR>I tried forwarding to line1id@127.0.0.1:5060, line1@myDynDNSname.com:5060 and line1id@myinternetip:5060.<br><br>i'm pretty sure i followed every step. <br><br>ports are forwarded properly on the nat as well.<br><br>Any ideas?<br> </DIV>Even though I posted that "trick", it appears to be a YMMV thing (works for some, and not for others).  <br><br>The main trouble with "tricks" like this, is that they have multiple pieces that ALL have to work right to get the trick to work.    Get even one minor piece out of place, and they don't work.  And worse yet, finding what is wrong, can be a little like "finding a needle in the haystack".<br><br>In such a situation, about the best you can do, is try to split up the steps as much as possible, and see if you can get the individual pieces to work.  You then "fix the pieces" as you find them, until you have all the individual parts working, and hopefully you can then put the pieces together to get everything to work.  But you have a lot better chance of getting things working as desired, if you don't try "debugging the setup" all at once (but instead try to "bite off" smaller chunks, and get each of those smaller chunks working, before moving on).<br><br>For example, PM someone with a Sipura elsewhere on the internet (me if you like) the "line1id@myinternetip:5060" info you were using (along with a good time to call), and see if that person can call your Sipura line "peer to peer" (i.e. use that "line1id@myinternetip:5060" info to call your Sipura directly).  If that works, you know that 1/2 of the setup needed for the "forwarding trick" (i.e. the ability of the line to "receive the call" from the other line) is working.  But even if that "peer to peer call" doesn't work, at least you have narrowed down the problem area, and can then look to get that part of the setup working (before moving on to the other parts of that "trick").]]></description>
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<pubDate>Fri, 26 Aug 2005 23:20:34 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14229893</link>
<description><![CDATA[<A HREF="/useremail/u/1243389"><b>JakesOnline</b></A> : <div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A> :</SMALL><BR><BR><B>And now (drum roll), how to forward all inbound calls to the OTHER line:</B><br>This is VERY USEFUL, because it either lets you have a TWO VoIP accounts that both "ring" the same phone, OR lets you use one account for all incoming, and a 2nd account for all outgoing (by putting the "phone" on the line with the outgoing VoIP service, and then forwarding all incoming calls on that other VoIP line to that one)! </DIV>I tried to forward incoming FWD calls from line 2 to line 1 which I use for voxee outbound. The calls go directly to the FWD message center. <br><br>I tried forwarding to line1id@127.0.0.1:5060, line1@myDynDNSname.com:5060 and line1id@myinternetip:5060.<br><br>i'm pretty sure i followed every step. <br><br>ports are forwarded properly on the nat as well.<br><br>Any ideas?]]></description>
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<pubDate>Fri, 26 Aug 2005 22:36:43 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14190340</link>
<description><![CDATA[<A HREF="/useremail/u/1250967"><b>kreil</b></A> : Aaaargh!<br><br>sipgate.co.uk support:<br><br>"We do not currently support call forwarding and reinvite.<br><br>There are no methods to forward your number to another Sipgate number."<br><br>Thoroughly disentchanted, I am. :mad:<br><br>That only leaves either having two phones sitting side by side, a hardware gadget to combine two phonelines, or an SPA-1001.<br>Or the Fritz!Box. I'm increasingly wondering...<br><br>&raquo;<A HREF="http://www.avm.de/en/index.php3" >www.avm.de/en/index.php3</A><br><br>Best wishes,<br><br>David.]]></description>
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<pubDate>Mon, 22 Aug 2005 04:25:33 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14188511</link>
<description><![CDATA[<A HREF="/useremail/u/1250967"><b>kreil</b></A> : Thanks, DracoFelis!<br><br>That's very useful to know. Off to ponder what to do an on the hardware side...<br><br>Best regards,<br>David.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14188511</guid>
<pubDate>Sun, 21 Aug 2005 21:28:37 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14188171</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  kreil <A HREF="/useremail/u/1250967"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>My earlier question regarding this may have drowned in all the other problems I've had but: Do you have any experiences with how the Sipura SPAs live behind a normal router that is not especially designed to prioritize voice traffic in an upload heavy environment?</DIV>In my limited experience, Sipura adapters seem to perform about as well as any ATA would when behind a non-QOS router.  <br><br>In particular, if you are in a tight bandwidth situation (heavy downloading on a PC, for example), you will likely notice the sound quality drop.  But if you have sufficient bandwidth (I'm on a 1.5meg/256k DSL line, for example), you will likely notice no sound quality issues UNLESS you are actively using that bandwidth at the time.  I rarely have sound quality issues with my SPA-3000, and I have not (yet) put it behind a QOS router.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14188171</guid>
<pubDate>Sun, 21 Aug 2005 20:34:08 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14188084</link>
<description><![CDATA[<A HREF="/useremail/u/1250967"><b>kreil</b></A> : Sure, you're perfectly right! I was just trying to clarify which ports you were referring to (I don't have an SPA3000 in front of me - yet?)<br><br>Say, I have just been pointed by colleagues to a non-Sipura device called Fritz!Box Fon by the German company AVM, which apparently is similar to the SPA3000 in features just better documented. Moreover, it is apparenlty running Linux inside the small quiet box (bit bigger than the SPAs), and there are inofficial firmware patches that activate a telnetd and let you login to it from the outside. Sounds like haven for finding tricks to do more with it than intended. Then again, I've never seen it mentioned on these forums. Have you heard anything about it ever?<br>Also, while there are versions of it that have a DSL modem built in and which do QoD traffic shaping, they don't have that for cable users (like myself).<br>So I'm back to a situation where I might just as well get an  SPA1001 for the little I need.<br><br>My earlier question regarding this may have drowned in all the other problems I've had but: Do you have any experiences with how the Sipura SPAs live behind a normal router that is not especially designed to prioritize voice traffic in an upload heavy environment?<br><br>With best regardd,<br><br>David.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14188084</guid>
<pubDate>Sun, 21 Aug 2005 20:19:53 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14187934</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  kreil <A HREF="/useremail/u/1250967"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR> What do you mean with the "line" jack of the SPA3000? It would have an FXS port going to the analogue phone, an FXO port to go to the (analogue) phone company, and an Ethernet port.</DIV>The SPA-3000 has two RJ11 telco jacks labeled "Phone" and "Line" (in addition to the RJ45 jack for the ethernet).  <br><br>The "Phone" jack is really an FXS port, and the "Line" jack is really an FXO port.  I think they just labeled them "Phone" and "Line" (and also used those terms in their web interface and their documentation), to make them easier for non-telco people to understand.  Since Sipura refers to them as "Phone" and "Line", I continued those labels, instead of calling them FXS/FXO, to avoid confusing people who might wonder which jack is which.<br><br>It makes sense in a way, as people are used to hooking up a "phone" to a jack labeled "Phone".  Likewise, it makes sense to "a normal user" to hook up a "phone line" to a jack labeled "Line".  But how many people that aren't already telco experts would know what an FXS or an FXO is?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14187934</guid>
<pubDate>Sun, 21 Aug 2005 19:55:18 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14187795</link>
<description><![CDATA[<A HREF="/useremail/u/1250967"><b>kreil</b></A> : Amazing! :o<br><br>Just imagine they would actually configure their devices to make this easy, there would probably be a TelCo revolt! ;)<br><br>Many thanks again for the heads up!<br><br>With best regards,<br><br>David.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14187795</guid>
<pubDate>Sun, 21 Aug 2005 19:32:10 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14187624</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  kreil <A HREF="/useremail/u/1250967"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>Regarding using the "line" of the SPA3000 - isn't this a POTS FXO port to go to the phone company?</DIV>Yes.  <br><br>The Sipura manual describes the "Line" port as what you hook up a real telco/POTS "phone line" to (if you wish to use the telco/POTS interconnect features of the device).  And that is in fact a useful thing to do with the "Line" port.  But it is not the only possible use of the "Line" port.  <br><br><B>Here are some other uses clever people over on the Voxilla.com forums have also made of the "Line" port:</B><br><br>1) Since the "Line" port is expecting a telco "phone line", anything that mimics a real phone line will also work.  So you can plug the "Phone" port of some other adapter, into the "Line" port of the SPA-3000.  This works, because the "Phone" side of pretty much any VoIP adapter is pretending to be a "phone line" to the "phone" they expect you to plug into the device.  Since the SPA-3000 "line" port is designed to work with a real "phone line", and the ATA you are plugging it into is pretending (to the "phone" it thinks it's hooked up to) to be a telco line, both adapters are "happy".  This allows you to combine VoIP features of an SPA-3000, with features from another "locked" VoIP adapter (on the same "phone").<br><br>2) One especially clever user (no it was not me) over on Voxilla.com, figured out that it is actually OK to connect the SPA-3000's "Line jack" (that expects to be hooked up to a phone line, and mimics a phone) and "Phone jack" (that expects to be connected to a "phone", and mimics a "phone line") together.  You might wonder why you would ever do such a "silly" thing.  The reason is, that it allows you to call into your SPA-3000 via the PSTN side "VoIP provider", authenticate (PIN access) with the adapter, to let you call out via the "telco line".  But in this case, since you have hooked the two jacks together (possibly via a telco "Y" cable, so you can also hook up a real "phone"), the "Phone" side of the SPA-3000 thinks that you have just picked up the phone in the house (as soon as the "Line" side of the SPA-3000 takes the "telco line" off hook)!  The practical upshot of this, is that you can call into your SPA-3000 by VoIP, and then out again making that call "as if" you were making it directly from the SPA-3000.  Round about way of doing things, but very clever IMHO.<br><br>3) And if you ever later setup an * server, you can redirect your SPA-3000 to the * box (instead of having the SPA-3000 do VoIP directly with the outside world).  If you do this, you essentially get one FXS port (the "Phone" jack of the SPA-3000) and one FXO port (the "Line" jack of the SPA-3000) that can be remotely controlled by your * server.  Again, I haven't done this myself yet (mostly because I haven't yet setup an * box).  But it is nice to know that my investment in Sipura adapters will _NOT_ be "wasted" if/when I do go to *, as I can just recycle the adapters as FXS/FXO interfaces for *!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14187624</guid>
<pubDate>Sun, 21 Aug 2005 19:07:47 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14187490</link>
<description><![CDATA[<A HREF="/useremail/u/1250967"><b>kreil</b></A> : Dear DracoFelis,<br><br>Maybe I'm confused here. What do you mean with the "line" jack of the SPA3000? It would have an FXS port going to the analogue phone, an FXO port to go to the (analogue) phone company, and an Ethernet port. Which of these are you referring to, or is there another, fourth port?<br><br>Best wishes,<br><br>David.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14187490</guid>
<pubDate>Sun, 21 Aug 2005 18:50:12 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14187419</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  ejrobinson <A HREF="/useremail/u/812490"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>1. Why use stun? Some people here say stun is to be avoided if possible. You seem to think the opposite.</DIV>While a public "routable" IP address (i.e. not being behind a NAT router) is probably best, I prefer STUN to the alternate ways of doing NAT traversal (if, like many of us, you are "stuck" behind a NAT router).  As I see it, here are the pros/cons to the three ways a Sipura can do "NAT traversal" (i.e. the three ways a Siupra can work behind a NAT router):<br><br>1) "Outbound Proxy" is popular with service providers, because it is also the way that works with the greatest variety of routers.  However, it has two serious side effects IMHO.  First off, it forces all calls to go via the provider's SIP proxy, which means no "tricks" like "SIP reinvite" to redirect the call.  And 2nd (and more serious), because all calls go via the provider's SIP proxy, you are pretty much stuck with only one VoIP provider (even on an SPA-3000)!<br><br>2) The 2nd way to do NAT traversal (properly run behind a NAT router), is to manually put in the outside IP addresses and ports.  This works, but can be a pain to setup.  And furthermore, you will have to manually change it whenever (for example), your dynamic IP address changes.  i.e. this is a PAIN to maintain.<br><br>3) That leaves STUN as the last (and my preferred) choice.  STUN is essentially like #2 above, but it automates the process.  Essentially, you connect to a STUN server somewhere on the internet, and that server will echo back to your device "you connected to me from external IP address so and so, on port such and so").  The device (in this case a Sipura adapter) then uses that STUN info (that was echoed back) to auto-configure itself as if you used technique #2 above.  So you get the benefits of technique #2, without having to manually configure the setup, and without having to reconfigure things if/when your IP address changes.  And the only "price" you pay for the auto-config, is that you are dependent upon using an external "STUN server".  That's why I prefer the STUN method of NAT traversal.<br><br><div class="bquote"><SMALL>said by  ejrobinson <A HREF="/useremail/u/812490"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>2. Would it be possible to use say lingo (and its ata) and a sipura 3000 at the same time, using say broadvoice or another voip service? If so, how?</DIV>I've not tried it, however it should be possible.  <br><br>What you would do is plug the "Phone" output of your lingo ATA into the "Line" jack of the SPA-3000.  You would then configure your SPA-3000 to access both your BYOD VoIP accounts (setup in the SPA-3000 itself) AND the analog "phone line" (on "gateway 0" of your SPA-3000) from the "Phone" hooked up to the SPA-3000 (and you will have to do this step, as the SPA-3000 is NOT configured this way by default).  Since the "Phone Line" hooked up to the SPA-3000 is really your Lingo adapter (in this example), you should (at least in theory) be able to access both the VoIP services programmed directly into the SPA-3000 and the services on the external (lingo) adapter from the same "phone".<br><br>Essentially what this setup does, is make the SPA-3000 think you are using both VoIP services and an existing "phone line" from the SPA-3000.  But the SPA-3000 has no way of knowing that the "phone line" you have hooked it up to, is really a "locked ATA" from another VoIP provider (in this case Lingo).  As far as the SPA-3000 is concerned, it is just using an analog phone line like you told it do.  And as far as the Lingo adapter is convinced, you just plugged a "phone" into it, like it was expecting (it's just that the "phone" in this case, is the "Line" side of the SPA-3000).  So in theory at least, both adapters should be "happy" (because they should both be seeing what they expect on those respective jacks), and you should be able to access both services from the same phone (attached to the "phone" jack of the SPA-3000).    ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14187419</guid>
<pubDate>Sun, 21 Aug 2005 18:37:22 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14187001</link>
<description><![CDATA[<A HREF="/useremail/u/1250967"><b>kreil</b></A> : Dear igi,<br><br><div class="bquote"><SMALL>said by  igi <A HREF="/useremail/u/620161"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>  I did the forwarding (127.0.0.1) from Line 1 to Line 2, when I had FWD on line 1 and Mutualphone on line 2. Never tried the other way around.<br></DIV>Redirecting FWD is less likely to cause trouble because they never loose money from a redirect. So, would you mind trying the other way 'round for me, please? This would give me an idea whether it's likely to be a provider problem. My provider, sipgate, apparently removed their forwarding feature from their web config pages, so I suspect they also don't honour a SIP reinvite message. (Thanks to DracoFelis for suggesting this might be a problem!)<br><br><div class="bquote"><SMALL>said by  igi <A HREF="/useremail/u/620161"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>My router has a whole bunch of ports forwarded to the SPA, so check that part.<br><br>And when you try the proxy option, did you use userid@proxy?<br>That was key in having it working.<br></DIV>My SPA2100 isn't behind a router, so it gets to see all the ports. Yes, I tried userID@service and userID@myIP:myPort, no go. I'm hence stymied as to what to do next. :(<br><br>With many thanks,<br><br>David.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14187001</guid>
<pubDate>Sun, 21 Aug 2005 17:31:46 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14186895</link>
<description><![CDATA[<A HREF="/useremail/u/1250967"><b>kreil</b></A> : Dear DracoFelis,<br><br>Thanks for your fast and helpful reply! :)<br><br>I almost forgot: With the SPA3000 I cannot really have two <B>incoming</B> VoIP accounts, right, not even with forwarding tricks because there is only Line 1, correct? :hmm:<br><br>So, combining the two SPA2100 phone lines with the extra gadget you mentioned seems like a great idea for receiving calls on one phone. It doesn't give me the choice of how to dial out though, for which I'd need an SPA3000, which again only has one incoming line - d'oh... :(<br><br>Have you ever considered / heard of people using the SPA-1001 with the *SE firmware? It sort of sounds like what I need.<br><br>Regarding using the "line" of the SPA3000 - isn't this a POTS FXO port to go to the phone company? The phone lines going "out" to the phones are FXS ports, so I don't think they can be hooked up.<br><br>Alternatively, another setup using two of the devices you found to combine two phone lines might be (dots just to make stuff align)<br><br><TT><br>Cablemodem<br>. |<br>SPA2100 --- Line 1 ----\_____ phone  <br>. | \ ----- Line 2 ----/ .. /<br>. | ...................... /<br>USR8054 (or other router) /<br>. | | .................. /<br>. | PCs ............... /<br>. | .................. /<br>SPA3000 --- Line 1 -- /<br></TT><br><br>with the devices combining the phone lines set to call out on the SPA3000. This would give 3 incoming VoIP accounts and 4+ outgoing ones but it seems a bit like overkill to me.<br><br>Also, you all seem to be using the SPAs behind standard routers. Does your router support QoS/diffserve? And if not, do you observe problems with voice calls when there is a data upload (not download) happening at the same time?<br>This worry was one of the main reasons I went for the SPA2100, which I could put in front of the router.<br><br>Looking forward to hearing from you,<br><br>David.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14186895</guid>
<pubDate>Sun, 21 Aug 2005 17:14:58 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14186813</link>
<description><![CDATA[<A HREF="/useremail/u/812490"><b>ejrobinson</b></A> : Your explanations are really quite interesting, though somewhat over my head at this moment. I have a couple of questions.<br><br>1. Why use stun? Some people here say stun is to be avoided if possible. You seem to think the opposite.<br><br>2. Would it be possible to use say lingo (and its ata) and a sipura 3000 at the same time, using say broadvoice or another voip service? If so, how?    <br><br>   -er]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14186813</guid>
<pubDate>Sun, 21 Aug 2005 17:02:29 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14186666</link>
<description><![CDATA[<A HREF="/useremail/u/620161"><b>igi</b></A> : David,<br><br>  I did the forwarding (127.0.0.1) from Line 1 to Line 2, when I had FWD on line 1 and Mutualphone on line 2. Never tried the other way around. My router has a whole bunch of ports forwarded to the SPA, so check that part.<br><br>And when you try the proxy option, did you use userid@proxy?<br>That was key in having it working.<br><br>I.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14186666</guid>
<pubDate>Sun, 21 Aug 2005 16:35:00 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14185568</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  kreil <A HREF="/useremail/u/1250967"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>From what you say, I probably need to swap my SPA2100 for an SPA3000.</DIV><div class="bquote"><SMALL>said by  igi <A HREF="/useremail/u/620161"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>I'm starting to think that I should go ahead and replace the 2000 for a 3000, that's what you did, right?</DIV>Yes, I did get an SPA-3000 to replace my SPA-2000 (I'm now using the SPA-2000 mostly as a backup/test VoIP device).  And for my needs, I find the SPA-3000 much more flexible than my older SPA-2000.<br><br>However, before you two rush off to spend money based upon my posts, please "look before you leap".  Take a minute to think about what features you need, and decide if the SPA-3000 has all those features (or if you will still have to do "work arounds").  While the SPA-3000 is IMHO much more flexible then the other Sipura models, and may in fact be what you need in your situation, it still has it's limits.  And it is quite possible to "bump your head" against the limits of the SPA-3000!  <br><br>IMHO one of the more limiting restrictions of the SPA-3000, is that you really only have ONE VoIP slot that lets you "ring your phone" (i.e. the main VoIP provider on the "Line 1" tab of the SPA-3000).  If you want to have two separate  "inbound" VoIP accounts (that both "ring your phone"), you are going to be back to cute "forwarding games" to get the 2nd one to ring (even on an SPA-3000)!  This is simply a limit of the SPA-3000, that those of us using it have to live with!<br><br>OTOH, if you have one account you want to "ring your phone", and one (or actually any number up to 4) other VoIP accounts that you use for "calling out" only, then the SPA-3000 is "the right tool for the job".  Remember, the 4 "gateway" slots on the SPA-3000 (as useful as they are) are for "calling out" only!  So it is reasonably easy to accept inbound calls via one provider, and do outbound calls via another (or even choose which provider to call out via, by how you dialed the call).  But if you want multiple INBOUND (ring your phone) VoIP accounts, the SPA-3000 won't do the job by itself (because the SPA-3000 only has one slot for a VoIP provider that can "ring the phone").<br><br><B>NOTE:  Here are some other "out of the box" ideas that you might want to consider, before making your final decision on what to get:</B><br><br>1) If you want a "hardware solution" for combining two "phone lines" (which could also be the two phone line outputs of an SPA-2000, or SPA-2100), the $25 device (SW18A) at the top of this merchant page, may do the trick for you.  This device is designed to hook up a one line answering machine, up to two separate "phone lines", and automatically "answer" whichever one is currently "ringing" (and outbound calls always go via the line chosen by the device's switch).  I don't see why you couldn't use such a device to allow a phone to "share" two "lines" of a VoIP adapter (for example, plugging this device into the 2-jacks of an SPA-2000, and then plugging your phone into this device), but I haven't tried it myself so YMMV:<br>&raquo;<A HREF="http://www.sandman.com/lineshar.html" >www.sandman.com/lineshar.html</A><br><br>2) If you do decide to buy an SPA-3000, and keep your existing adapter (instead of selling the old adapter on Ebay), you should be able to combine both adapters for greater features.  Remember, the SPA-3000 has a "line" jack, designed for hooking up a real phone line.  But I don't see any reason why the output of another Sipura adapter wouldn't look enough like a "phone line", to allow the SPA-3000 to use it.  This should (in theory) let you put a inbound/outbound VoIP account on an older adapter, and have the SPA-3000 fully use that account, in addition to the 5 VoIP account (1 inbound/outbound + 4 outbound only) that you can program directly into the SPA-3000.<br><br>3) Don't overlook any "forwarding features" that a provider may offer (for example, on their web portal, if they have one).  If you can tell an incoming VoIP provider to forward the call to the provider you have registered in your Sipura line, you should be able to "ring" the line with that provider (although you may have some latency or other issues doing it).]]></description>
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<pubDate>Sun, 21 Aug 2005 13:32:03 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14185345</link>
<description><![CDATA[<A HREF="/useremail/u/1250967"><b>kreil</b></A> : Hi igi,<br><br>It seems you got a bit further with your SPA2000 than I did with my SPA2100, I couldn't even get the 127.0.0.1 forward to work.<br>Were you able to try this with both of your providers? I mean, could you forward from either provider to the other, or did it work only in one particular direction?<br><br>Inspired by your proxy example, I tried forwarding to proxy.at.sipgate.net --> nothing.<br><br>As for replacing the SPA2xxx, I wonder whether the SPA1001 with the *SE firmware wouldn't be the cheapest and perhaps easiest (?) solution to having at least two providers "online". Apparently, with one FXS line, one can switch between two provider accounts and get both of them ringing the same phone (different ring tone). They say...<br><br>With best regards,<br><br>David.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14185345</guid>
<pubDate>Sun, 21 Aug 2005 12:47:16 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14184699</link>
<description><![CDATA[<A HREF="/useremail/u/620161"><b>igi</b></A> : DracoFelis,<br><br>  Thanks, now I see.<br><br>Having said all that, do you have any guess why I can't make your port-forwarding trick to work? I know it's a wild question....<br><br> I'm starting to think that I should go ahead and replace the 2000 for a 3000, that's what you did, right? Probably can sell the spa2k on ebay for something reasonable.<br><br>Thanks,<br>I.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14184699</guid>
<pubDate>Sun, 21 Aug 2005 10:48:00 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14184159</link>
<description><![CDATA[<A HREF="/useremail/u/1250967"><b>kreil</b></A> : Dear DracoFelis,<br><br>Thank you so much for your fast and friendly reply!<br><br>Some thoughts and further questions:<br><br><div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>For "dial plan" help, pay very careful attention to the "Appendix 1" (the last 3 or 4 pages of the PDF containing the user guide).<br></DIV>Ah, thanks!! :)<br><br><div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR><div class="bquote"><SMALL>said by DKreil:</SMALL><BR><BR>- SIP account at provider (sipgate, Austria):<br>4832525@sipgate.at<br></DIV>This one should work if BOTH of the following are true:<br><br>1) Anyone off the net can call your phone by calling the SIP URI "4832525@sipgate.at", even if/when they don't have an account with "sipgate.at". If this isn't the case, the forwarding will fail, because you have just forwarded to a destination that doesn't accept the call.<br></DIV>Yes, this is my "external" SIP number, and anyone should be able to call it.<br><br><div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>2) The VoIP provider you are registered with on the line you are forwarding from supports the "SIP reinvite" (call redirect) message. If not, call forwarding will never work from that line, as the Sipura doesn't appear to actually do the forwarding (instead it tells the calling SIP party/device to call somewhere else, and not all SIP providers/devices will follow that redirect message)!<br></DIV>Oh... I see. This might actually be the problem. I think sipgate used to have a forwarding feature on its website but removed it since. If they are thorough, they would also not reply to SIP reinvite messages :(<br>They do have voicemail though, so I wonder whether they do that via a SIP reinvite message or with a different internal method...<br><br>Do you know of anyone who got SIP reinvite working with sipgate lately? (In older threads, people just report using the sipgate config feature on the web that was then still available.)<br><br>Else, do you perhaps know anyone using a provider offering UK numbers and supporting SIP reinvite?<br><br>What I am basically looking for is placing <I>and</I> receiving calls on one phone to <I>and</I> from two different SIP provider accounts.<br><br>From what you say, I probably need to swap my SPA2100 for an SPA3000. While at the moment my setup is<br><br>Cablemodem<br>  |<br>SPA2100 (w/NAT)  --  phone<br>  |<br>USR8054 router (w/NAT, again) and WiFi access point<br>  |<br>PCs<br><br>this means<br><br>Cablemodem<br>  |<br>USR8054 router (w/NAT) and WiFi access point<br>  |    |<br>  |  SPA3000<br>  |<br>PCs<br><br>As the friendly folks from USR have not been able to reply to my repeated enqiries whether their device supports QoS/diffserve, I worry whether in this setup I will have voice over data prioritization. Do you have any experiences regarding this with your own setup/router?<br><br>Lastly, I noticed quite strong echo on my calls, despite leaving the default echo cancelling activated. Is there something I can do about this in my setup, or is this a provider issue, a VoIP technology issue, or more likely to come from the non-VoIP leg of the call?<br><br>Many thanks again for your kind help!<br><br>With best regards,<br><br>David.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14184159</guid>
<pubDate>Sun, 21 Aug 2005 08:19:45 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14182555</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  igi <A HREF="/useremail/u/620161"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>Anyway, I tried something else: in the forwarding field of the User 1 sipura page, I entered xxxxxx@sip.winradius.net, where xxxxxx is my Mutualphone account.</DIV>Doing it that way, you aren't really forwarding the call to your other line per se.  <br><br>What you are really doing is forwarding the call to the MutualPhone proxy, which is then calling your registered MutualPhone adapter (the other side of your Sipura).  As long as MutualPhone lets people on the internet call you via their proxy (using that technique), that sort of "forwarding" should work.<br><br>However:<br><br><div class="bquote"><SMALL>said by  igi <A HREF="/useremail/u/620161"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR> This works fine, and my question now is about quality and latency.</DIV>Since you aren't forwarding the call directly to the other line, the call is going via a more indirect route (i.e. via MutualPhone, and back to you).  This will increase latency some.  You will have to decide if this increased latency is a problem in your environment.<br><br><div class="bquote"><SMALL>said by  igi <A HREF="/useremail/u/620161"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>Question #2: my line 2 codec is the worst G729a. If another FWD user calls me (using the better, G711 codec), is this the one used when line1 gets forwarded to line2?</DIV>If you are forwarding a FWD call to the MutualPhone proxy (which seems to be what you are doing), MutualPhone will talk to that SIP device via whatever CODECs both the MutualPhone proxy and the calling party can handle.  If their is no CODECS in common (for example, the caller is using G711, and the MutualPhone proxy can't handle that CODEC), then the call will fail.  <br><br>Once MutualPhone answer the call, it will either redirect it directly to you, or (more likely) call you and "relay" the voice stream (i.e. think of it like a "3-way" call at MutualPhone, connecting you and the inbound FWD caller).  This means that you will hear the call via whatever CODEC MutualPhone talks to your SIP adapter by, and the remote site will use whatever CODEC they are talking to MutualPhone by.  Furthermore, if those aren't the same CODECs (for example, you use G729a with MutualPhone, and they use G711), then MutualPhone will have to translate the CODECS (which could degrade sound quality further).<br><br>However, even with the potential lower sound quality (and higher latency) of doing things this way, it may still be "worth it" to you, if it allows you to receive both MutualPhone and FWD calls on the same "phone"/adapter (especially if that lower sound quality is not especially objectionable when on the phone).]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14182555</guid>
<pubDate>Sat, 20 Aug 2005 23:14:50 EDT</pubDate>
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<title>Re: How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14182197</link>
<description><![CDATA[<A HREF="/useremail/u/620161"><b>igi</b></A> : Hi DracoFelis,<br><br>  Man, I tried getting your line1-to-line2 trick to work on my Sipura 2K, and no luck. Line 1: FWD, Line 2: Mutualphone. I can forward internally no problem (127.0.0.1), but once I get out to my outside IP address (I do have one), I get two types of behaviour:<br><br>1. If I call my FWD using a softphone, Line 2 rings but I can't hear. Hearing from the softphone is OK.<br><br>2. If I call from a regular phone using IPKALL, I always get the recorded IPKALL message, never rings.<br><br>Anyway, I tried something else: in the forwarding field of the User 1 sipura page, I entered xxxxxx@sip.winradius.net, where xxxxxx is my Mutualphone account. This works fine, and my question now is about quality and latency. In a previous posting you told me that latency is not an issue when sip proxies talk to each other (would that be still the case the way I did it?). Question #2: my line 2 codec is the worst G729a. If another FWD user calls me (using the better, G711 codec), is this the one used when line1 gets forwarded to line2?<br><br>Sorry if some of these questions are obvious.... and thanks a lot for your postings.<br><br>I.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14182197</guid>
<pubDate>Sat, 20 Aug 2005 22:16:59 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14181895</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by DKreil:</SMALL><BR><BR>Is there any form of documentation of call plan syntax around or did you learn all that from the published examples & trial and error? I have paged through the Users' guide but it seems more like a list of features available for provisioning by VoIP providers.</DIV>The user guide leaves a lot to be desired, but it is still a good reference.  For "dial plan" help, pay very careful attention to the "Appendix 1" (the last 3 or 4 pages of the PDF containing the user guide).  Try to read them carefully, as the majority of the dial plan syntax is actually documented there!  Beyond that, ask over on the Voxilla.com Sipura forum, as there are a LOT of knowledgeable people willing to help "the lost" with setting up a dial plan to do something specific.<br><br>NOTE:  Many of my dial plan "tricks" came from simply reading the user manual, trying to really understand what it MEANS (instead of just glossing over it), and then simply "thinking outside the box" as to what I could make the Sipura do with that syntax.  I then just "ran the experiment" to see if I could actually carry out what my hypothesis said I should be able to do (in most cases yes, but not all the time).<br><br><div class="bquote"><SMALL>said by DKreil:</SMALL><BR><BR>I have tried to learn from your examples but I cannot get the call forwarding to work on my SPA-2100.</DIV>As has been mentioned in this thread, I discovered AFTER first posting that "trick" that it doesn't work for all users or all VoIP providers.  So YMMV with that forwarding trick.  That said:<br><br><div class="bquote"><SMALL>said by DKreil:</SMALL><BR><BR>- localhost & port<br>  4832525@127.0.0.1:5061<br></DIV>This may work for an INTERNAL call within your LAN.  But it will NOT work for "forwarding".<br><br>Apparently, the Sipura forwarding works by telling the calling party to redirect their call elsewhere.  Not only does this mean that the "elsewhere" has to be seeable from outside your LAN, but it also means that if your incoming VoIP provider refuses the "redirect", the call won't forward!  Some VoIP providers will ALWAYS refuse the redirect, so "call forwarding" with a Sipura appears to be a YMMV thing.<br><br><div class="bquote"><SMALL>said by DKreil:</SMALL><BR><BR>- SIP account at provider (sipgate, Austria):<br>  4832525@sipgate.at<br></DIV>This one should work if BOTH of the following are true:<br><br>1) Anyone off the net can call your phone by calling the SIP URI "4832525@sipgate.at", even if/when they don't have an account with "sipgate.at".  If this isn't the case, the forwarding will fail, because you have just forwarded to a destination that doesn't accept the call.<br><br>and 2) The VoIP provider you are registered with on the line you are forwarding from supports the "SIP reinvite" (call redirect) message.  If not, call forwarding will never work from that line, as the Sipura doesn't appear to actually do the forwarding (instead it tells the calling SIP party/device to call somewhere else, and not all SIP providers/devices will follow that redirect message)!<br><br><div class="bquote"><SMALL>said by DKreil:</SMALL><BR><BR>- external IP & port<br>  4832525@voice.kreil.net:5061<br></DIV>The external IP and port seems to work only if all of the following are true:<br><br>1) That really is the line's userid, external IP, and port.   Pay special attention to make sure that the (dynamic) DNS really points to your external IP.<br><br>2) Make sure you didn't forget to forward the SIP port (in your case UDP 5061) on your router to your Sipura.  While you normally don't have to do "port forwarding" when you have a line "registered", in the case of direct IP calling (and this trick is essentially doing a call forward directly to the other side of your Sipura), you do need the port "forwarded" at your router, or the call likely won't ring!<br><br>3) You remembered to set "Ans Call Without Reg: yes" on the line receiving the forwarded call.<br><br>and finally 4) And even if/when you do all the above, it may still not work, because forwarding is still dependent upon the calling VoIP provider/device listening to the Sipura when it tells it to "go somewhere else".  And some VoIP providers routinely ignore such "SIP reinvite" messages!<br><br><div class="bquote"><SMALL>said by DKreil:</SMALL><BR><BR>Is it likely that it's a difference between the SPA2100 and the 3000? Is it worth going for the 3000 instead?</DIV>"Forwarding" a VoIP call seems to be "hit or miss" with pretty much any Sipura (since the Sipura doesn't itself do the forwarding, and not all calling parties cooperate with the forwarding).<br><br>However, the SPA-3000 does give you some options that the other Sipura models don't.  But those extra features may or may not help you, depending upon exactly what you are trying to accomplish.  <br><br>IMHO here are the places the SPA-3000 seems to do better in:<br><br>1) The SPA-3000 makes it pretty easy to have one main inbound/outbound VoIP provider, and up to 4 additional "outbound call only" providers ON THE SAME PHONE.  This gives you a lot of flexibility in setting up where the call goes when you call different numbers (whereas most of the Sipura models only have one VoIP provider per line, although many of the Sipuras have "2 lines", whereas the SPA-3000 really only has one).  FWIW:  This is the main reason I went with the SPA-3000.  This feature alone, makes "mix and match" (such as having one VoIP provider for inbound calls, and another provider for outbound calls) many times easier than the other Sipura models!<br><br>2) The SPA-3000 has a "Line" jack, that lets you hook up a real phone line to the unit (and then also access that "phone line" from the phone hooked up to the Sipura).  While I haven't yet experimented with this myself, many other people have apparently got this working.  And this feature does allow the SPA-3000 to combine the best of VoIP with features of an existing "phone line", as if they were one combined service.  And remember, the "phone line" could be a VoIP adapter from another company (to allow you to combine BYOD VoIP with some provider's "locked down service"), if so desired.  <br><br>3) And if you are really into a painful setup, the true "power users" use the "Line" jack with the "VoIP to/from PSTN" "gateway" features of the SPA-3000.  This allows the unique "hop on and hop off" tricks of the SPA-3000 (which I haven't even attempted to setup yet, btw).  For example, how about calling your Sipura from work (or a cell phone), entering a "PIN code", and getting dial tone AS IF you made the call from home?  With the proper SPA-3000 setup, you can do it!  Likewise, with the proper SPA-3000 setup in another country, you can call VoIP to the other Sipura, enter a PIN code, and then dial out on their local phone line!  Again, only the SPA-3000 (of the various Sipura models) is setup to do this sort of "hop on hop off" tricks, but the possibilities are VERY POWERFUL.  Just keep in mind that those setups are also not trivial to get right (as some of the posts over in the Voxilla.com Sipura forum clearly indicate).<br><br><div class="bquote"><SMALL>said by DKreil:</SMALL><BR><BR>Lastly, I have tried to follow the discussion about using Gateway Accounts on the forum but am not quite sure of the latest: Is there now a way one can use Gateway Accounts or information in the calling plan to switch providers which<br> - require a password<br> - have a different user ID?<br></DIV>Yes.  And that is one of the key features of the 4 "gateway" provider slots!  But this is an SPA-3000 only feature, so it won't work on your SPA-2100.<br><br>To use the "gateway" slots, enter things like this:<br><br>"Gateway x: " gets "userid@sip_proxy" (i.e. your user account at your provider).<br><br>"GWx Auth ID: " gets your account "userid" (yes, I know it's silly to have it here AND in the "Gateway x" field, but that's how Sipura made it work...<br><br>"GWx NAT Mapping Enable: yes" (assuming you are behind NAT, which most of us are).<br><br>"GWx Password: " is your SIP password with that provider (obviously).<br><br>Once you have the above info entered into a gateway, it's then just a matter of setting up your "dial plan" to allow you to dial out via that gateway entry.  For example, I have included the following pattern in my dial plan, to auto-route normal USA calls via my "gateway 1" provider (in my case DialPad.com):<br><div class="code"><PRE><span class="codetext">1&#91;2-9&#93;xx&#91;2-9&#93;xxxxxxS0 &lt;:@GW1&gt; </SPAN></PRE></DIV>The important thing to remember with the "gateway" fields (besides the fact that they are an SPA-3000 only feature), is that it is a 2-part process.  You have to setup the "gateway" fields correctly with your SIP/login info, AND you have to modify your "dial plan" so that your Sipura knows what calls to route via that alternate outbound VoIP provider!<br>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14181895</guid>
<pubDate>Sat, 20 Aug 2005 21:20:41 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14180978</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : This is <B>fantastic</B> and <B>exactly</B> what I'd need!! <br><br>Is there any form of documentation of call plan syntax around or did you learn all that from the published examples & trial and error? I have paged through the Users' guide but it seems more like a list of features available for provisioning by VoIP providers.<br><br>I have tried to learn from your examples but I cannot get the call forwarding to work on my SPA-2100. I have tested the following forwarding destinations, but all without success (forwarding from line 1 to line 2):<br>- localhost & port<br>  4832525@127.0.0.1:5061<br>- SIP account at provider (sipgate, Austria):<br>  4832525@sipgate.at<br>- external IP & port<br>  4832525@voice.kreil.net:5061<br>When I call the number from on my mobile, after a delay, I get "number error do you wish to retry", while when calling with a softphone (x-lite), the forward is simply ignored (!), and I get passed through to the account's voicemail after a while.<br><br>What else could I try?<br><br>Is it likely that it's a difference between the SPA2100 and the 3000? Is it worth going for the 3000 instead?<br>I opted for the 2100 because I could place it infront of my router, hence avoiding some QoS issues. If I got the 3000, I'd have to put it behind my router and I have no idea whether that supports QoS VoIP prioritization (it's an USR8054).<br><br>Lastly, I have tried to follow the discussion about using Gateway Accounts on the forum but am not quite sure of the latest: Is there now a way one can use Gateway Accounts or information in the calling plan to switch providers which<br> - require a password<br> - have a different user ID?<br>In my case, I need an account in the UK and one in Austria, currently using sipgate (sort of).<br><br>I'd be grateful for your thoughts!<br><br>In return, I'm afraid I don't have any tricks yet on the configuration side that I could share. I could only tell you about the joys of bridging different analogue phone cabling standards (different in <I>all</I> the countries my life has touched: US, UK, Germany, Austria) but I doubt you want to hear that ;)<br><br>With many thanks again for your help<br>and best regards,<br><br>David.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14180978</guid>
<pubDate>Sat, 20 Aug 2005 18:55:48 EDT</pubDate>
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<item>
<title>How to do an SPA-3000 setup like mine...</title>
<link>http://www.dslreports.com/forum/remark,14179920</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : Considering how I struggled getting my SPA-2000 setup (and I'm a senior level IS type), and later moving onto the SPA-3000 (which also took a little work), I should have realized this earlier.  However, it took me a while to realize that even a "basic setup" can be a "trick", due to the lack of good guides on which of the 500 or so options a Sipura presents you with are the good ones to override/set (when first getting started).  With that in mind, here is an edited version (with some of my personal/private info blocked out) of my SPA-3000 setup, and why I made some of the choices I did.  Hopefully, it will help someone else get their initial setup working (or working better).<br><br>NOTE:  This is for an SPA-3000, but many of the options also are present on an SPA-2000 (and I presume most of the other Sipura models as well).<br><br>NOTE:  Some of the Sipura tabs are too big to fit on one screen capture (GIF file).  When that happens, I broke the screens up into two (and in one case 3) screen captures.<br><br><B>FWIW:  Here is the rough order I would recommend for setting things up in:</B><br><br>While it is pretty obvious, the 1st thing you need to do, is to physically hookup your Sipura.  Ethernet cable goes to your router/switch, power goes to the Sipura's wall wort, and a normal/analog/POTS phone gets plugged into the "Phone" port of your Sipura (or the 1st phone port if you are using a Sipura with multiple "Phone" ports).  At this point, use a real phone for the setup/testing, even if you are later going to wire the Sipura into your house wiring (like I have btw).<br><br>NOTE:  If you are using an SPA-3000, make sure you plug the phone into the "phone" jack (it's labeled on the plastic case, but you have to look closely to see it), not the identically looking (but electrically quite different) "Line" jack!  But at this point, you do NOT need to hook up your "phone line" (leave that for a later more advanced setup, once you have the basics working).<br><br>If you got the above physical hookup working, and your router's DHCP server gave the Sipura a LAN IP address, you should be able to do the next step, which is have the Sipura talk to you.  Pick up the phone, and quickly press "****" (for "*" characters in a row) on the phone.  The Sipura should talk back to you saying you are in the "Sipura configuration menu".  Press "110#" (number 110, followed by the "#" key), and the Sipura should tell you (over the phone) what the Sipura's current IP address is.  Copy down the address for later use.<br><br>NOTE:  If it comes back "0.0.0.0", you have problems with DHCP (or network connectivity), and can't proceed to the next step.  In that case, check your DHCP server, unplug the power from the Sipura, plug the power back in, and try again.  If that still doesn't work, you will have to consult the Sipura manual on how to setup your IP address from the phone (it's a pain, and I've never had to take that step).  Otherwise, you now have the address of the Sipura, to proceed to the next step.<br><br>At this point, you should be able to fire up your web browser, and point it at the IP address of the Sipura.  This should (assuming this is an initial setup, and there are no passwords on your Sipura) bring up the Sipura's web interface as a normal user.  Click on the "Admin Login" link (upper right of screen), followed by the "Advanced" link (also upper right).  This will put you into the Sipura's Admin/Advanced setup (where you really want to be IMHO).<br><br>The 1st thing I did at this point, was configured the SYSTEM tab to turn off DHCP access, and manually configured static settings for my LAN.  The reason for this, is so I could later know exactly what LAN IP address the Sipura was at.  Not only does this later save the "pick up the phone and press ****  110#  game", it also gives you options to forward ports from your router to your Sipura (and some of my advanced "tricks" require it).  In my case, I choose x.x.x.96 (on my LAN), because I was not using that address, and I had already excluded the 1-100 range from my router's DHCP server (so another device on the LAN wouldn't also get the Sipura's IP address).  I also setup my router as the primary DNS server (with my ISP as the 2nd DNS), and used "pool.ntp.org" for the time server.  I picked that time server, as the "load balancing" built into the "time pool" project is more "net friendly" then having your devices hit the main net time servers.<br><br>When the above static network setup is done, triple check it, before proceeding to the next step.  You don't want to "lock yourself out", by having a static network setup that doesn't let you into the web interface!  Once you are sure you have that setup correctly, hit the "Submit All Changes" button, and reboot the Sipura (i.e. unplug power, wait a few seconds, and plug power back in).  At this point (and beyond), you should access the Sipura via the static IP address you chose.  Again, you want to switch to the "Advanced" part of the "Admin Login".<br><br>Next, I recommend setting up your "NAT Support Parameters" for the Sipura.  For most of us this will mean "STUN".  The "STUN" settings are at the bottom of the "SIP" tab.  The effect of these settings are not obvious, and turning on a setting you think should help, can often break things.  Conversely, failing to set a setting you need, can also make things not work!  After spending several hours working through this, I found the "NAT Support Parameters" I documented in my SIP.GIF file seem to work well (on a variety of VoIP providers).  So feel free to copy these STUN settings "as is" at this point (and then press "Submit All Changes").  NOTE: It is OK to use "stun.fwdnet.net" as your STUN server, even if you later setup the Sipura with some other VoIP provider (since STUN servers are NOT tied to the VoIP provider you are using).  Of course, if you have another preferred STUN server to use, that would work as well.<br><br>Decide what SIP ports you want to use for your Sipura.  The main consideration is to not interfere with other VoIP devices (or soft phones) that are on the same ports.  The Default UDP 5060 for "Line 1" will work for most people.  I just currently have my Sipura on 5063, because I was previously using it on the same LAN as another SIP device that was already using port 5060.  Setup the ports on the Sipura "Line" tab, as you like.  It also will help inbound calling (although depending upon your VoIP setup and router, it may not be necessary), if you forward the chosen UDP SIP ports from the router to the Sipura (which is a LOT easier, if you have already setup the Sipura with a static LAN IP address).<br><br>At this point, you can try putting your main VoIP service into the Sipura (or test first with a free service like "Free World Dialup").  In my case, I chose FWD as my main service, so that I could receive inbound calls (as well as make outbound calls) with FWD.  If possible, try to avoid using the "Outbound Proxy:" field (which in general should NOT be needed with the STUN settings I mentioned before), as use of that field may prevent many of the advanced "tricks" that are mentioned in this thread.  However, at a minimum you will want to fill in the "Proxy:" field (with your chosen provider), the "UserID:" field with your VoIP account (in the case of FWD, your account and your phone number are the same), and the "Password: " field with your account password.  For most of us, you will probably also want to specify "NAT Mapping Enable: yes", "NAT Keep Alive:  yes", and "Register: yes".  Other settings may be necessary depending upon your provider (for example, FWD works much better IMHO if you customize your "dial plan" first, and some providers need you to specify something specific in the "Display Name:" field).  And many of us like to lower our "Register Expires:" setting from its default of 3600 seconds (or once/hour), to something much shorter (I have mine set to 300, or once every 5 minutes).<br><br>Use the phone to test your setup.  Do you have "dial tone", when you pick up the phone?  Can you call some "test number" and hear the other end?  Can you call some "echo test" number, and hear your voice echoed back at you?  Do you have another phone (or a friend?) that can call you, and see if your Sipura rings?  Can you talk when you do?  If the answer to all these questions is "yes", you probably have your first VoIP provider setup and working on the Sipura.  If not, you need to start playing with the settings, trying to get it to work.<br><br>NOTE:  If you enter "613@fwd.pulver.com" (without the quotes) into a "speed dial" (on the "User 1" tab), you should be able to just use that speed dial to call the FWD "echo test" (even if/when you have a DIFFERENT provider setup in "Line 1").  For example, my speed dial 3 is setup like this.  So I can get the FWD "echo test" by simply dialing "3#" (three and the "#" key) on the phone.  This can be very useful for checking setups, to see if you have things like "NAT traversal" working, to allow basic outgoing calls!<br><br>Once you have your main VoIP provider setup, you can start to tweak things for added features (including adding additional VoIP accounts on the 4 "Gateway" fields, if you have an SPA-3000, and have additional providers you desire).  In general, many of these tweaks will involve modifying the "dial plan", so that you can access these features from your phone!  Do yourself a favor, and read the section of the manual dealing with how to setup a dial plan, or you will likely be LOST in this process!  In simple terms, the "dial plan" is a description of how you want the Sipura to interpret the digits you dial (from your "phone"), as a combination of commands to the Sipura and actual digits sent via VoIP.  By being "creative" with the "dial plan", you can do some really amazing "tricks" (some of which have already been mentioned in this thread)!  For example, maybe you want some calls (normal USA LD for example) to use one VoIP provider, but you also want to be able to call FWD numbers (and "peering partners") directly?  If so (and yes, I do that in my setup), you will have to modify your dial plan to tell the Sipura how to route the call, based upon what you "dialed" on your phone, and that is done by setting up the "dial plan".<br><br>NOTE:  Do NOT copy what the GIF shows as my "dial plan", as my actual dial plan is MUCH longer (it is cut-off in the screen capture).  Construct your own "dial plan" (starting with a simple/working one, and adding to it to add features "one by one") to meet your needs.  Both this thread and the Sipura forum over on Voxilla.com, are good resources for putting together decent "dial plan" strings.<br><br>NOTE:  The default Sipura "dial plan" seems to be optimized to let you call anywhere, but not very well.  You will IMHO do a LOT better, if you think about what type of numbers you need to call (and which ones you really don't ever want to call), think up a dial scheme that makes that dialing "unique" (for example, I do all of my really "special" (non-default) dialing (call routing) with "# digit normal_dialing_for_that_provider #"), to allow me to override my default dialing setup.  Likewise, "1 area_code number" automatically goes via my "Gateway 1" provider (DialPad.com), so that normal POTS dialing happens just the same as if I was on a POTS line.  But the important thing about a "dial plan" is to first figure out what you want to have happen with a given dialing pattern, and then translate that into the "dial plan" syntax of the Sipura.  You will be amazed at what can be automated this way.<br><br>If you have a Sipura SPA-3000 (but NOT with the other "cheaper" models of Sipura), you can also put another 4 outbound (you can call out, but they can't call you) VoIP providers into the 4 "Gateway" fields.  You will then HAVE TO modify the dial plan, to get access to these providers (as the default Sipura "dial plan" does NOT give you access to your "Gateway" providers)!  If done right, this lets your attached "phone" have immediate access to the main "Line 1" provider (inbound calling, and outgoing) + 4 more "outbound only" VoIP providers, transparently.  For example, I can just as easily call FWD numbers, as POTS numbers (via my choice of DialPad, Teliax, or IconnectHere), or even use the SIPphone "conference rooms", all from the same "phone" (in my case distributed to multiple phones, via my house's existing "line 2" phone wiring) I have hooked up to my SPA-3000.  At the same time inbound FWD calls will also ring that "phone", which also means that inbound POTS calls via my free IPKall number (forwarded to FWD) will also ring the "phone".<br><br>Oh yeah, I also set "Provision Enable: no" (the default is "yes"), on the "Provisioning" tab, so that there is no risk of some VoIP provider later coming around and changing my VoIP settings "behind my back".<br><br>If you have managed to get this far, you will have a very useful SPA-3000 setup.  Beyond that, you can add features (for example, some of the "tricks" in this message thread) slowly over time.  For example, I found MichiganTelephone's tricks (for the "Regional" tab of the Sipura) to be handy in making the device behave much more like a "real" telco line.  And some of you may want to hook up another phone line to the "Line" jack of the adapter, and set that up to work with your Sipura attached phone (just hooking up the phone line to the "Line" jack is not sufficient, you also have to turn on that feature in the Sipura, using the web interface).<br><br>But hopefully, this is enough info to get some of you "in over your heads" started.  And for some of the rest of you, this might give you some ideas for improving your setup.<div class="borderless"><TABLE WIDTH=95% align=center border=0 CELLPADDING=4"><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/14179920?c=878609&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG class="apic" BORDER=0 TITLE="33091 bytes" WIDTH=600 HEIGHT=696 SRC="/r0/download/878609.thumb600~16dda426f62ed7df12a7db8a8b1c6b48/LINE1.GIF/thumb.jpg" ALT="Click for full size"></A><br>Line 1</TD></TR><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/14179920?c=878610&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG class="apic" BORDER=0 TITLE="37859 bytes" WIDTH=600 HEIGHT=671 SRC="/r0/download/878610.thumb600~56b5e233b741fb71be9b2d43c7a3d66d/LINE2.GIF/thumb.jpg" ALT="Click for full size"></A><br>Line 1</TD></TR><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/14179920?c=878611&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG class="apic" BORDER=0 TITLE="8500 bytes" WIDTH=600 HEIGHT=159 SRC="/r0/download/878611.thumb600~8a68bd83df231ed45550cca6f4f7a6d0/LINE3.GIF/thumb.jpg" ALT="Click for full size"></A><br>Line 1</TD></TR><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/14179920?c=878612&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG class="apic" BORDER=0 TITLE="26332 bytes" WIDTH=600 HEIGHT=667 SRC="/r0/download/878612.thumb600~c19a4fbec2b313a9514b03d8aac319a3/PROV.GIF/thumb.jpg" ALT="Click for full size"></A><br>Provisioning</TD></TR><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/14179920?c=878613&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG class="apic" BORDER=0 TITLE="36785 bytes" WIDTH=600 HEIGHT=705 SRC="/r0/download/878613.thumb600~d7b2a557c6332e421d6fca7b5e64b153/REG1.GIF/thumb.jpg" ALT="Click for full size"></A><br>Regional</TD></TR><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/14179920?c=878614&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG class="apic" BORDER=0 TITLE="36459 bytes" WIDTH=600 HEIGHT=710 SRC="/r0/download/878614.thumb600~e7948c0ecbdfd338741124290b2220df/REG2.GIF/thumb.jpg" ALT="Click for full size"></A><br>Regional</TD></TR><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/14179920?c=878615&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG class="apic" BORDER=0 TITLE="36129 bytes" WIDTH=600 HEIGHT=705 SRC="/r0/download/878615.thumb600~faf2ad145c204a83b5c200575a852899/SIP.GIF/thumb.jpg" ALT="Click for full size"></A><br>SIP</TD></TR><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/14179920?c=878616&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG class="apic" BORDER=0 TITLE="18563 bytes" WIDTH=600 HEIGHT=360 SRC="/r0/download/878616.thumb600~19a5cbd303ee077d192389332771eee6/SYSTEM.GIF/thumb.jpg" ALT="Click for full size"></A><br>System</TD></TR><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/14179920?c=878617&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG class="apic" BORDER=0 TITLE="32078 bytes" WIDTH=600 HEIGHT=657 SRC="/r0/download/878617.thumb600~35f576791ef1bd7c8be5961daeb555c5/USER.GIF/thumb.jpg" ALT="Click for full size"></A><br>User 1</TD></TABLE></div>]]></description>
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<pubDate>Sat, 20 Aug 2005 15:44:47 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14131597</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : I currently have two Sipura 3000 adapters, one of which is installed in my parent's house in a different country. The general idea is to call each other through the SIP adapters, but allowing for obtaining dial tone in the other country for PSTN dialing.<br><br>One easy trick in the receiver end was to add 123-555-5555@gw0 as a no-answer-cfw with a high enough delay (say 20 seconds). In this way, an incoming SIP call to the sipura adapter will be picked up automatically after 20 seconds and will be connected to the 123-555-5555 phone number using the PSTN. This setting is usefull for no-answer call my cell phone deal.<br><br>What I'm trying to accomplish now is a speed dialing configuration with FWD so that the number to call is included by the calling party, and the Sipura only has to answer the SIP call, and dial a PSTN number determined in by the caller speed dial.<br>I tried with one stage dialing, but couldn't make it to work. Does anybody has good samples that can be shared?<br><br>IB]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14131597</guid>
<pubDate>Sun, 14 Aug 2005 11:52:03 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14071149</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : i tried thsi for loopback, but does not work. yes i have tried dial plans, yes it works for calls from PSTN line . But if you try with VOIP2 that only allows GW0 (FXO Port).<br>If anyone have done some experiments, please let us know.:(:(<br><br>regards,<br>Bilal]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14071149</guid>
<pubDate>Sat, 06 Aug 2005 04:53:48 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14068094</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  SuperCPA <A HREF="/useremail/u/728530"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>On another point, apparently you can use the gateway accounts in the dial plan of the PSTN line by adding the :@gwX function to the end.  The SPA-3K manual says no, but it does work.<br> </DIV>According to the Voxilla.com forums, that is only true on the latest "version 3" build of the Sipura software.  If you are still on some older firmware version (I'm using the latest version 2 build of the SPA-3000 firmware, for example), that feature isn't available.  <br><br>And at least for the present, the version 3 software seems to be having some "growing pains" (bugs that aren't present in earlier firmware versions).  So YMMV on this "trick".]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14068094</guid>
<pubDate>Fri, 05 Aug 2005 18:46:20 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14067143</link>
<description><![CDATA[<A HREF="/useremail/u/1223723"><b>gnexus</b></A> : Thanks Blohner!<br><br>I will try that. I saw that thread previously at Voxilla, but I didn't pay much attention to it since it mentioned 3000 and I have a 2002.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14067143</guid>
<pubDate>Fri, 05 Aug 2005 16:30:05 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14066178</link>
<description><![CDATA[<A HREF="/useremail/u/650535"><b>blohner</b></A> : Actually there is a utility that works quite well for my SPA-2002 (and I presume for all other sipura's as it is originally made for the 3000). See &raquo;<A HREF="http://voxilla.com/forum-viewtopic-t-2155.html" >voxilla.com/forum-viewtopic-t-2155.html</A>  for details... It saves everything but the passwords - I used it exentsively when I was setting up asterisk (and nothing was working) to switch between running to * extensions and connecting directly to Stanaphone and Sipgate...<br><SMALL>--<br>I am addicted to speed --- OOL speed that is --- <BR>~Help find a cure for cancer~Proud Member <A HREF="http://www.dslreports.com/forum/disco"><I><B>Team Discovery</B></A></SMALL>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14066178</guid>
<pubDate>Fri, 05 Aug 2005 14:26:07 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14065428</link>
<description><![CDATA[<A HREF="/useremail/u/728530"><b>SuperCPA</b></A> : <div class="bquote"><SMALL>said by ABilal:</SMALL><BR><BR>Hi,<br><br>Have you tried with SPA-3000, As i just did , but it does not work, it actually take offhook FXO after 5 sec.<br> </DIV>Hello ABilal,<br><br>No, I have not been able to test using the loop back address (127.0.0.1) to ring the phone with an incoming call from the PSTN VoIP Line provider.  I only have one incoming DID (a numerical Authorization ID and User ID).  I only know that it will work on an SPA-3K, to ring the VoIP phone from the POTS line, when ring through is disabled.<br><br>On another point, apparently you can use the gateway accounts in the dial plan of the PSTN line by adding the :@gwX function to the end.  The SPA-3K manual says no, but it does work.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14065428</guid>
<pubDate>Fri, 05 Aug 2005 13:05:53 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14064600</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Hi,<br><br>Have you tried with SPA-3000, As i just did , but it does not work, it actually take offhook FXO after 5 sec.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14064600</guid>
<pubDate>Fri, 05 Aug 2005 11:09:39 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14062917</link>
<description><![CDATA[<A HREF="/useremail/u/639783"><b>digiblur</b></A> : <div class="bquote"><SMALL>said by  gnexus <A HREF="/useremail/u/1223723"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><br><br>Another thing to note which is not really a Sipura "trick," but is vital if you use a lot of non-standard Sipura settings:<br><br>Back it up!<br><br>Unfortunately Sipura doesn't supply a backup utility. They expect that the unit will be remotely provisioned by the VoIP provider. The way that is done is by using a Sipura utility to compile the settings and push them to the SPA. The compilation utility is not provided to end users. Therefore the only way for an end user to back it up is by saving the web page for each SPA config. tab.<br><br>That is a PITA but it works, just remember you MUST save it as an actual web page. Do not just take a snapshot. Otherwise if you have a long dial plan that is not all visible in the box it won't be saved.<br> </DIV>You don't have to save every tab.  When you go into advanced mode or admin mode it's all one HTML source.  <br><SMALL>--<br>FWD#64466(6PM-11PM GMT-5) <BR> &raquo;<A HREF="/forum/remark,13371620">[Sipura] Make your Sipura Speak! - GetSipura Guide</A><BR> Drop me a PM if you'd like a custom Samurize plugin for your device.</SMALL>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14062917</guid>
<pubDate>Fri, 05 Aug 2005 02:02:49 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14059137</link>
<description><![CDATA[<A HREF="/useremail/u/1223723"><b>gnexus</b></A> : <div class="bquote"><SMALL>said by  SuperCPA <A HREF="/useremail/u/728530"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>BTW, anyone know of a "Sipura Dialing Plans for Dummies" book, and where to get a copy? <br> </DIV>I could use one! I hate reading stuff like books and manuals on the computer. I should really print out the Sipura manual, but my inkjet is in another state so the only printer I have is a high end color dye-sub Tektronix. It takes 1/2 hour to warm up, 1/2 hour to cool down, and a few kW to keep running! Not too practical for small print jobs.<br><br>Some more useful tricks:<br><br><B>User config-Speed dial settings -</B> <SMALL>"The SPA supports user programming of up to 8 long distance, local, international or emergency numbers and/or IP addresses for fast and easy access." If you don't need this spot for other purposes, such as your own special numbers, you can add some useful numbers in this section instead of into the dial plan. That makes them easier to add or read.</SMALL>:<br><br><B>Number examples:</B><br><br><I>yournumber@proxy.yoursipprovider.com</I><br><SMALL>This allows me to quickly check my voice mail, I have to enter it as a SIP URI to get around my other funky IP-IP dial plans, otherwise I get a busy signal like a POTS phone. Entering it like this should also facilitate the call faster since no translation to a SIP URI is needed.</SMALL><br><br><I>17474743246@proxy01.sipphone.com</I><br><SMALL>This is the SIPphone echo test number (you could also use the FWD one, but I think SIPphone's works better). Anything you say is repeated back to you. Use it to check your line for latency and the quality of your call.</SMALL><br><br><I>17474745000@proxy01.sipphone.com</I><SMALL><br>This is the SIPphone welcome number (you could also use the FWD one, but I think SIPphone's works better). Use it to repeat back to you your phone number (it works for any number!) and check the inbound quality of your call.</SMALL><br><br><I>411@proxy01.sipphone.com</I><SMALL><br>This is the SIPphone link to <A HREF="http://www.1-800-555-tell.com/">TellMe</A>, a really awesome IVR (Interactive Voice Response) information service. Use it to check the time, weather, stock quotes. Yeah, you could use the 1-800 number, but I think it is safer using it through SIPphone because it's likely that TellMe uses the callerID info for marketing purposes (why else would they have the service?).</SMALL><br><br>If you can think of more good VoIP test numbers that would be a good addition to the thread please post them!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14059137</guid>
<pubDate>Thu, 04 Aug 2005 17:14:17 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14055258</link>
<description><![CDATA[<A HREF="/useremail/u/728530"><b>SuperCPA</b></A> : <div class="bquote"><SMALL>said by user=DracoFelis:</SMALL><BR><BR>And FWIW while I have gotten that trick to work with my SPA-2000, I have so far been unsuccessful at getting a similar stunt to work on my SPA-3000 (to allow the SPA-3000's PSTN VoIP provider to ring the "phone").  So this is definitely a YMMV "trick".<br></DIV>Instead of forwarding, to ring the phone you might try (if you have not already) using the 127.0.0.1 in the dial plan, for the PSTN Line, that is used by the VoIP caller.<br><br>My PSTN Line dial plan is (xx.|P5:Line1ID@127.0.0.1:5060)<br>Sorry, the greater than and less than brackets do not want to show up.<br><br>It allows my wife to call in on the POTS Line, which has the ring through disabled, and if she does not enter a phone number, after five seconds, the phone attached to Line 1 rings.<br><br>The Sipura manual (July 2004) on page 44, seems to indicate that if authentication method for VoIP callers is set to "none" all callers are accepted for service, and the dial plan used is the "VoIP Caller Default DP".  I would try it myself, but have no incoming DID for my PSTN VoIP Line provider.<br><br>BTW, anyone know of a "Sipura Dialing Plans for Dummies" book, and where to get a copy? ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14055258</guid>
<pubDate>Thu, 04 Aug 2005 08:43:40 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14053070</link>
<description><![CDATA[<A HREF="/useremail/u/1223723"><b>gnexus</b></A> : <div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>4) If I'm correct in what is going on, that would also explain how to directly call a Sipura "peer to peer" without any "service provider" (even FWD) at all!  Anyone with a Sipura want to help test this theory out?  Can another user on the internet directly "ring our phone" by calling the same URI that we have in our "forwarding" trick?  I bet the answer is "yes", but I would like to test this theory out first, before saying for sure that it will work!<br> </DIV>Yes! You are correct Draco. It does work! (of course it does, SIP is P2P so it has to. . .unless there's a Sipura prob. with P2P) You beat me to the next "trick". ;)<br>I was going to post it, but then I had problems:<br><br>I had it working after trying your trick! Now it doesn't however. . . The voice is audible inbound only and I'm trying to troubleshoot. That's why I'm back here at BBR. . .searching through old posts on NAT traversal.<br><br>You would think it's, and somehow it probably is, a NAT traversal problem. It is a strange one, however, because it was working before, and occasionally has since, after settings changes. Then it stops, however. I even tried setting the SPA as DMZ. That still only had it working for a single call. Due to that I'm starting to think that my SPA is screwed up because of that, and intermittent outbound dropouts (every 10s or so) on my regular VoIP provider (and all the other free VoIP services, too). I even tried resetting the SPA with no improvement. . .<br><br>Since it is likely a NAT traversal problem, I am (impatiently) awaiting the next DD-WRT beta. It has a SER built in which will eliminate NAT traversal problems and the use of the STUN gun. :D<br><br>Edit: Still no luck. . .One way audio, fleeting two way audio. I did find out something interesting in the SIP logs, however:<br><div class="bquote">SIPphone server response:<br>Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))<br></DIV>That means the SIPphone proxy server is using the same code as the SER being added to DD-WRT!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14053070</guid>
<pubDate>Wed, 03 Aug 2005 22:30:00 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14051986</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  igi <A HREF="/useremail/u/620161"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>I have a question about latency though in your Line 2/Line 1 forwarding: have you checked if this increases latency/delay when you call another FWD user?</DIV>No, I have not checked.  However, based upon some posts over on Voxilla.com (since the time I came up with that "trick"), it appears that Sipura adapters do NOT do forwarding "internally", but instead send a SIP "reinvite" message back to the calling SIP device.  Essentially, the Sipura sends a message back to the original SIP provider, telling that provider to send the SIP traffic to an alternate destination.  <br><br>As a result, the predicted behavior of my forwarding "trick" is this:<br><br>1) Since the Sipura essentially tells the sending provider to redirect the traffic, any provider that ignores such redirect messages will likely not work with this forwarding trick (and in fact, at least one person who tried my forwarding trick with a commercial VoIP provider appeared to fail for this reason).  Thankfully FWD does pay attention to "reinvite" messages, so this trick can work with inbound FWD calls (which happens to be also what I originally tested it on).<br><br>2) Since the SIP messages aren't really "forwarded", but instead the original caller is told (by the SIP reinvite message) to try sending the call elsewhere, you shouldn't have any more "latency" then if the user had called that other port directly.<br><br>3) This also explains why you have to use the EXTERNAL (internet, NOT LAN) address of your Sipura, and why you have to forward the SIP ports on your router to your Sipura (for this trick to work).  Simply put, it appears that the "SIP reinvite" message (that your Sipura gives out as a result of the "forward" setup) tells the remote SIP device how to call the other port of your Sipura directly "peer to peer".<br><br>4) If I'm correct in what is going on, that would also explain how to directly call a Sipura "peer to peer" without any "service provider" (even FWD) at all!  Anyone with a Sipura want to help test this theory out?  Can another user on the internet directly "ring our phone" by calling the same URI that we have in our "forwarding" trick?  I bet the answer is "yes", but I would like to test this theory out first, before saying for sure that it will work!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14051986</guid>
<pubDate>Wed, 03 Aug 2005 19:52:03 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14051732</link>
<description><![CDATA[<A HREF="/useremail/u/1223723"><b>gnexus</b></A> : Another thing to note which is not really a Sipura "trick," but is vital if you use a lot of non-standard Sipura settings:<br><br>Back it up!<br><br>Unfortunately Sipura doesn't supply a backup utility. They expect that the unit will be remotely provisioned by the VoIP provider. The way that is done is by using a Sipura utility to compile the settings and push them to the SPA. The compilation utility is not provided to end users. Therefore the only way for an end user to back it up is by saving the web page for each SPA config. tab.<br><br>That is a PITA but it works, just remember you MUST save it as an actual web page. Do not just take a snapshot. Otherwise if you have a long dial plan that is not all visible in the box it won't be saved.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14051732</guid>
<pubDate>Wed, 03 Aug 2005 19:18:27 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14048490</link>
<description><![CDATA[<A HREF="/useremail/u/620161"><b>igi</b></A> : Hi DracoFelis,<br><br>Thanks a lot for your master tricks. I have a question about latency though in your Line 2/Line 1 forwarding: have you checked if this increases latency/delay when you call another FWD user? <br><br>I'm interested in using the same trick, but I wouldn't want to add more delay to what I already have, since some of my FWD calls are literally to the other side of the world. If the IP forwarding is done all internal inside the SPA, then I don't see why this would add delay, but with the external IP address trick, are you now opening another routing through the internet?<br><br>Thanks,<br><br>I.]]></description>
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<pubDate>Wed, 03 Aug 2005 12:40:34 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14037556</link>
<description><![CDATA[<A HREF="/useremail/u/1223723"><b>gnexus</b></A> : I just had a Sipura incident:<br><br>The SPA was set for DHCP. Just now suddenly I couldn't access the SPA configs. Turns out the DHCP had renewed onto a different IP. It somehow lost most of the net configs. Whoops! :o<br><br>Recommendation:<br>Use static IP if possible.<br><br>Be sure to reset to DHCP if/when travelling!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,14037556</guid>
<pubDate>Tue, 02 Aug 2005 01:20:30 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,14036320</link>
<description><![CDATA[<A HREF="/useremail/u/1223723"><b>gnexus</b></A> : I have been reading this thread since it began, and I finally got my Sipura a few weeks ago. I have also been looking for other Sipura posts in other forums, and seen Draco's posts at Voxilla. There is a lot of Sipura information, but at the same time I have further questions which Sipura only seems to supply to providers. So I'll be doing more research to work on the provider-type settings, such as provisioning and pushing configs into the Sipura. <br><br>My biggest confusion was over setting up the dial plan syntax. Since I've never set up telecom switches that was new and cryptic. I have now set up the dial plans to allow toll-free calls through SIPphone, thanks to Draco, since my VoIP plan is not unlimited.<br><br>I just found a new (not really Sipura) neat Sipura trick which solves a problem:<br><br>I want people to be able to call me for free via SIP without transiting to PSTN. I was hoping to accomplish that simply by switching providers. Now it appears, however, that my new provider blocks incoming SIP calls even though they allow outgoing. Since I also have free VoIP accounts using SIPphone, FWD, and Earthlink that wouldn't seem to be much of a problem, right? <br><br>At first I thought I would just use the Draco forwarding trick, and forward the SIPphone number at Line2 to my PSTN-connected Line1. So I got that working, thanks to Draco. It did not work, however, until I forwarded my SIP ports to the Sipura. I did not really want to do that, however. I have a single fixed IP, and am using an eyeBeam softphone on that computer along with the Sipura. I was afraid that the eyeBeam wouldn't work if the port was forwarded to the Sipura. I was VERY surprised to find out the softphone still works, even after forwarding the SIP ports to the Sipura! :D <br><br>So there's a trick for ya! You can still use a softphone in conjunction with the Sipura on your network, even when the ports are forwarded.<br><br>Thanks again, Draco! You're my pal for helping me out!<br><br>Edit:<br>I just found another (not really Sipura) neat Sipura trick. Using the next version of DD-WRT in your WRT54G you will <I>easily</I> (unlike now, where it requires compilation) be able to use the built in SER (SIP Express Router) to forward calls through your NAT without STUN and also to (BINGO!) be able to use multiple VoIP devices behind your firewall without using Asterisk. Using the SER your call quality will improve and you won't have to worry about multiple ATAs or softphones working inside your network.]]></description>
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<pubDate>Mon, 01 Aug 2005 22:07:28 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13996160</link>
<description><![CDATA[<A HREF="/useremail/u/1223723"><b>gnexus</b></A> : Has anybody noticed the copyright statement on the Sipura Web Site?<br><br>It specifically prohibits posting Sipura configuration info on "Forums such as Broadband Reports." :(<br><br>I'm sure it applies only to admin stuff,however, and not user setups.<br><br>When I saw it, however, I thought it was a stupid statement, since anybody really interested in hacking the provider setups could easily find the info elsewhere. . . ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13996160</guid>
<pubDate>Wed, 27 Jul 2005 15:58:29 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13974924</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : If you are trying to access the Sipura (admin) web pages remotely, you may trip over a Sipura bug.  Apparently, the "web server" on a Sipura adapter will try to keep sending page data until it is "acknowledged" by the remote web browser, even AFTER the Sipura exceeds it's internal memory dedicated to web page buffering.  :(<br><br>Now the "good news" is that this appears to only be a problem when you are on a "slow" connection (by local LAN standards).  So you are not likely to see this behavior on your local LAN.  But if you are trying to access a Sipura remotely over the internet (as I was earlier today), you may trip over this bug.  According to the forums over on Voxilla, you can work around this bug by limiting your web page buffer (on the computer you are accessing the Sipura from) to around 20K.  <br><br>However, if this seems like too much work to you, you may have another option.  At least on FireFox, when you run into this error, you will see an error page with a "try again" option.  If instead of trying to reload the initial page, you use the "try again" option, FireFox will continue the web page download from the point where the Sipura crashed the transfer, and Voila you have the entire Sipura configuration page (as desired)!<div class="borderless"><TABLE WIDTH=95% align=center border=0 CELLPADDING=4"><TR><TD ALIGN=CENTER VALIGN=CENTER BGCOLOR=#000000 nwrap COLSPAN=3 WIDTH=100%><A HREF="/speak/slideshow/13974924?c=865098&ret=L2ZvcnVtL3IxNDE4MjE5Ny54bWw%3D"><IMG class="apic" BORDER=0 TITLE="47916 bytes" WIDTH=600 HEIGHT=223 SRC="/r0/download/865098.thumb600~ce4f1e931b633133f9ba08e3746b9f55/SIPURA.GIF/thumb.jpg" ALT="Click for full size"></A><br>Example FireFox error page when Sipura fails to remotely show it's web based admin page.</TD></TABLE></div>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13974924</guid>
<pubDate>Sun, 24 Jul 2005 20:10:39 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13922928</link>
<description><![CDATA[<A HREF="/useremail/u/1234836"><b>Blade1212</b></A> : I think I know what the problem might be. The voipbuster userid is text whereas I think the Linksys needs this to be numeric to work.<br><br>The way I tested the parlour trick was to phone from voipbuster(text) -> sipgate(numeric).<br><br>And to add insult to injury... every voipbuster ID need to contain at least one letter.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13922928</guid>
<pubDate>Mon, 18 Jul 2005 16:00:29 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13922674</link>
<description><![CDATA[<A HREF="/useremail/u/1234836"><b>Blade1212</b></A> : Now logged in properly. I tried the Sipura flash but it didn't take.<br><br>The parlour trick actually worked first time with the 127.0.0.1 address, but for some reason the forwarding trick does'nt. When I enter anything in the "Cfwd All Dest:" it goes engaged.<br><br>I think I've reached the end of the line with this one.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13922674</guid>
<pubDate>Mon, 18 Jul 2005 15:29:51 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13914156</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by Blade1212:</SMALL><BR><BR>Do you have any other ideas</DIV>My only suggestion at this point, is to try to get my "call one line from the other one" ("parlor trick") to work first.  The reasoning being, that the one line calling the other will verify all your settings (for going between the adapter lines) except for the "call forwarding".  So if you can get one line to call the other, you are (in theory) only one step/setting away from getting the forwarding to work.<br><br>And FWIW I can't promise which adapters allow this particular "forwarding trick", as that "feature" is NOT documented by Sipura, so there is no promise that they will support it across their line (or even across differing firmware versions for the same adapter).  And FWIW while I have gotten that trick to work with my SPA-2000, I have so far been unsuccessful at getting a similar stunt to work on my SPA-3000 (to allow the SPA-3000's PSTN VoIP provider to ring the "phone").  So this is definitely a YMMV "trick".<br><br>BTW: If you are going to ask questions in this thread, at least have the decency to register (for this forum), so that others can PM (send you a "private message" via this board) you with suggestions...<br><br><div class="bquote"><SMALL>said by Blade1212:</SMALL><BR><BR>is the SPA-2000 flash avalable online - that might be worth a try ??</DIV>If you had carefully read this thread, you would have seen this line in a previous post:<br><div class="bquote">You can download the user manual (and upgraded firmware versions) from this web page:  &raquo;<A HREF="http://www.sipura.com/support/index.htm" >www.sipura.com/support/index.htm</A></DIV>However, I really would <B>NOT</B> recommend putting one adapter's firmware into another adapter.  It might work, but (due to differing internal hardware) it could just as easily turn your working adapter into a "door stop" (i.e. it could completely fry your adapter, and that sort of "damage" would probably not be covered under warantee)...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13914156</guid>
<pubDate>Sun, 17 Jul 2005 13:47:47 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13913651</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : I thought that might have worked, but still no luck.<br><br>I tried the yes/no setttings you suggested along with both the internal and external IP address, but all I get is an engaged tone as soon as I dial Line 2 - if I remove the "Cfwd All Dest:" address, it rings on Line 2 as normal. So it would appear that the problem is getting from Line 2 -> Line 1.<br><br>Do you have any other ideas - is the SPA-2000 flash avalable online - that might be worth a try ??<br><br>tia]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13913651</guid>
<pubDate>Sun, 17 Jul 2005 12:28:21 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13913112</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by blade1212:</SMALL><BR><BR>I have a linksys pap2 which appears identical to the Sipura 2000.</DIV>I don't know much about that device, other than the fact that the firmware development was apparently out-sourced to Sipura, and therefor the device has firmware that is similar to (but not quite identical to) the real Sipura adapters.  So it's anyone's guess which Sipura tricks will work on that device.  But FWIW:<br><br><div class="bquote"><SMALL>said by blade1212:</SMALL><BR><BR>Line 2 config page...<br>Register: no<br>Make Call Without Reg: Yes<br>Ans Call Without Reg: Yes</DIV>The first setting (Make Call Without Reg: Yes) is correctly set on the line you are forwarding from.  <br><br>HOWEVER, the 2nd setting (Ans Call Without Reg: Yes) should go on the line you are forwarding to (i.e. put that setting on "Line 1" of your device).  While it shouldn't harm anything to also leave "Ans Call Without Reg: Yes" on Line2, the side that needs that setting is Line1.<br><br><div class="bquote"><SMALL>said by blade1212:</SMALL><BR><BR>User 2 config page :<br>my_voipbuster_username@myname.dyndns.org:5060<br></DIV>Assuming the entry you are talking about is the always forward entry (what's called "Cfwd All Dest:" on my SPA-2000), your value looks decent to me.  <br><br>However, if it still doesn't work (after fixing your other issue, above), it probably wouldn't hurt to try "my_voip_username@127.0.0.1:5060" just as a test.  Might work, might not, but there is no harm in seeing if the "loopback address" will help in your case.  <br><br>The reason I say this, is that it occurred to me (after writing my previous post on the subject) that which address works (the external address, as in my case, or the loopback address) might depend upon how the forwarding actually occurs.  Remember there are two ways that the SIP standard supports forwarding:  1: You can either tell the other SIP proxy to redirect the call to a new adapter, or 2: you can receive the call directly, and then make your own call to the new location.  <br><br>While the Sipura manual is notably silent on how the Sipura adapters actually do the forwarding, it makes sense that method #1 would use the external address (and require a proxy on the other end that will cooperate with this redirect), whereas method #2 would use a local (loop back) address (and be done fully "locally", and therefore not require a remote proxy to cooperate).  <br><br>So if the Sipura adapters support both methods of "forwarding" (and the manual doesn't seem to say), the adapter might auto-choose which method is used based upon the cooperation (or lack thereof) of the remote proxy (remember I was testing with FWD, which does support/cooperate with method #1 forwarding, and that may be why I had to use the external address to get it to work for me).  If so, there is a chance that your remote proxy is refusing to cooperate with a method #1 redirect, and therefor the Sipura may (if my hypothesis is correct) try method #2.  If so, the "loopback address" has a chance of working in your case.  <br><br>Of course, this is just an educated guess at this point, but the testing of it is reasonably easy to do.  So what have you got to lose by trying?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13913112</guid>
<pubDate>Sun, 17 Jul 2005 10:59:25 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13912298</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : DracoFelis, pls help............<br><br>I have a linksys pap2 which appears identical to the Sipura 2000. I have Line 1 setup with voipbuster and line 2 setup with sipgate.co.uk. I now have hours of torture under my belt fixing stuns, nats, router ports etc - all these are now fixed :)<br><br>My plan is to use your documented trick to route incoming calls from Line 2 -> Line 1 and make the Line 1 phone ring. So the end result would be voipbuster for outgoing calls and sipgate for all incoming (where I have a fixed number allocated to me by sipgate). I also only want to use one phone. <br><br>Here's the key settings I've tried.<br><br>Line 2 config page...<br>Register: no<br>Make Call Without Reg: Yes<br>Ans Call Without Reg: Yes<br><br>EXT SIP Port: <br><br>User 2 config page :<br>my_voipbuster_username@myname.dyndns.org:5060<br><br>The minute I put in the user 2 address, I get an engaged tone when I call the Line 2 number. It doesn't route it to Line 1 either.<br><br>I have ports 5060 and 5061 on my router pointing to my Linksys pap 2 at 192.168.1.20 - I've made this static in my internal network.<br><br>Any ideas what might be wrong ?<br><br>Thanks in advance]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13912298</guid>
<pubDate>Sun, 17 Jul 2005 06:09:21 EDT</pubDate>
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<title>Voice encryption...</title>
<link>http://www.dslreports.com/forum/remark,13910068</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : I currently have no way of telling how good the encryption built into the Sipura adapters is (so it may be easier to "crack" then people believe).  And furthermore, the only place I've found to get the encryption keys is the Voxilla.com web site (and no matter how secure a web site is, it is still a "3rd party" that has access to your keys, and from a security/encryption standpoint that is "a bad thing").  And finally, encryption will only work with other sites/adapters that support encryption, which most of them don't (so most of your calls will still probably behave the same as if you didn't have encryption enabled).  <br><br>So all things considered, I wouldn't trust Sipura "secure calls" (encryption) to keep people from "listening in" on your calls.  However, it might "slow them down" some, and IMHO you have nothing to lose by enabling this feature.  After all, if the encryption works, you may have prevented some "eves-dropper" from intercepting you call.  But even if the encryption fails, you are no worse off then before (because the "normal case" is to send your voice "in the clear"). With that in mind, here is how I just enabled voice encryption on my Sipura adapters (tested by calling between my SPA-3000, and my older SPA-2000).<br><br>1) Sign up with a free account at &raquo;<A HREF="http://www.voxilla.com" >www.voxilla.com</A>.  This is necessary, as the only place I am currently aware of that allows you to get the encryption keys is the voxilla web site, and they require you to be a "member" to run their "wizards".<br><br>2) Go to the Voxilla Sipura encryption Wizard.  You can either find the link (on the left hand panel) at the main voxilla web site, or the current "direct link" is at this URL:  &raquo;<A HREF="http://voxilla.com/certrequest.php" >voxilla.com/certrequest.php</A><br><br>3) On the above web page, completely fill out the form.  Apparently the form will fail unless ALL of the field (including the "Your name or alias" field) are filled in.  In the case of the "Your name or alias" field entry, if you don't want to fill it in, do what I did, and just use a single space character as your "name".  This Wizard will allow you to push a set of keys (public and private) to your Sipura.  Don't forget that this Wizard needs to be run once for each "Line" that the Sipura has (for example, my SPA-2000 has two "lines", and each line needs a different "key").<br><br>4) Check to make sure that the Voxilla encryption wizard pushed encryption keys to your adapter.  You can verify this by looking at the (admin login, advanced) "Line x" tab fields:  "Mini Certificate:" and "SRTP Private Key:".  If these fields are still empty/blank (the default for Sipura adapters) than the Wizard didn't do its job.  However, if the "Mini Certificate:" has a bunch of characters in it, and the "SRTP Private Key:" shows "*************" (indicating that something hidden is in that field), than the public/private keys were entered into your Sipura (which is what you want to have happen).<br><br>5) The Voxilla wizard suggests you enter "*18" to do a "secure" call.  However, why would you want to bother with that?  Wouldn't you want the Sipura to just "default" to "secure" mode when it can?  To make "secure"/"encrypted" calls the default (while still allowing other calls when encryption isn't available), go over to the (admin login, advanced) user tab for the line, and change "Secure Call Setting:" to "yes".<br><br>6) At this point, the Sipura should work the same as it did before, EXCEPT when you call DIRECTLY (not via a 3rd party) some location that supports encrypted calls (for example, another Sipura with this feature enabled).  When encryption is supported (by both sides), the Sipura appears to take an extra second or so to initially connect, and then beeps at you three times (to let you know that the call is "secure").  I also noticed a little extra (maybe 1/3 second?) latency/lag in the call, but the sound was otherwise "clear" when I tried this on my LAN between my SPA-2000 and SPA-3000.<br><br>NOTE:  I have not yet had an opportunity to test encryption with "Free World Dialup" (so YMMV).  But according to posts I've seen in the past, FWD does support (pass though) Sipura voice encryption when all of the following are the case:  1) Both parties (the caller and the called party) in the call have Sipura adapters with encryption keys installed (and remember they are NOT installed by default, you have to use the Voxilla wizard to get them).  2) Both parties are on FWD directly, not via some "peering partner".  3) Both parties are using "fwd.pulver.com" as their proxy (i.e. neither party is using the alternate "fwdnat.pulver.com").  and 4) the party making the call has told their adapter to make a "secure call" (for example by having "Secure Call Setting: yes").<br>  ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13910068</guid>
<pubDate>Sat, 16 Jul 2005 20:47:57 EDT</pubDate>
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<item>
<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13864430</link>
<description><![CDATA[<A HREF="/useremail/u/780972"><b>WhyADuck</b></A> : I just came across a <A HREF="http://www.tomsnetworking.com/Reviews-205-ProdID-SPA2100.php"> Product Review of the Sipura Analog Telephone Adapter (SPA-2100)</A> at <A HREF="http://www.tomsnetworking.com/">Tom's Networking</A> and it contains an interesting <A HREF="http://www.tomsnetworking.com/Reviews-205-ProdID-SPA2100-4.php">evaluation of the quality of service (QoS) features</A>.  According to the test, it would appear that much better results are obtained with QoS set to TBF (Token Bucket Filter) as opposed to CBQ (Class Based Queueing) or QoS disabled. The summary states, "Enabling TBF-based QoS (Sipura's recommendation), essentially eliminates packet loss, but doesn't do much for jitter. On the other hand, switching to CBQ-based QoS appears to increase packet discards, but significantly lower jitter."  Apparently packet discards cause greater voice degradation than jitter.<br><br>Those of you who are interested in QoS on the SPA-2100 will probably find the test results and associated comments interesting.  And yes, I'm aware that the QoS can't control what happens to your packets after they get out onto the 'net, but if you do any significant amount of uploading, it could definitely make a difference.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13864430</guid>
<pubDate>Mon, 11 Jul 2005 14:46:58 EDT</pubDate>
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<item>
<title>Blocking costly area codes via your &#x22;Dial Plan&#x22;.</title>
<link>http://www.dslreports.com/forum/remark,13813346</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : Considering how easy it is to dial high cost regions, and considering that some providers (including the DialPad.com I use) do NOT have an option to block costly calls from their portal, I thought I would put some number "blocking" into my Sipura dial plan.  <br><br>While blocking "international dialing" is pretty obvious (just don't allow more than 11 digits of dialing), some countries near the USA (for example the Bahama's) are considered costly international calls, yet they dial as if they were USA numbers (ouch)!  So I googled for a list of area codes, and used that info to construct a list of dial plan patterns that my Sipura should disallow/block (Sipura adapters let you disallow a pattern by putting an "!" at the end of the pattern string).  I make no promises that this list is accurate/complete (feel free to modify it, and post those modifications to this thread, if you find something wrong).  And I also made no effort to block Canadian numbers (as many providers, including mine, charge Canada numbers the same as the USA).  <br><br>However, FWIW here is my 1st attempt at a "costly area code block list".  If you want to use this list, I would suggest putting it as the first entry in your dial plan, so that the entries in this list override any entries that allow normal 11 digit dialing.<br><div class="code"><PRE><span class="codetext">124&#91;26&#93;x.!|1441x.!|1456x.!|1473x.!|1555x.!|1590x.!|1649x.!|1758x.!|1767x.!|1809x.!|<br>126&#91;48&#93;x.!|1&#91;27&#93;84x.!|134&#91;05&#93;x.!|16&#91;68&#93;4x.!|186&#91;89&#93;x.!|1&#91;89&#93;76x.!|1&#91;79&#93;00x.!|&#91;34678&#93;11!</SPAN></PRE></DIV>NOTE:  As always, Sipura dial plan entries are on a single line.  Please remove the extra "line break" (in the plan entry above), before copying this string into the front of your Sipura dial plan.<br>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13813346</guid>
<pubDate>Mon, 04 Jul 2005 21:02:12 EDT</pubDate>
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<title>Calling SIP URIs via the &#x22;Speed Dials&#x22;...</title>
<link>http://www.dslreports.com/forum/remark,13811591</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <B>Fun and games with "speed dials":</B><br><br>Here's another one I just figured out.  The Sipura adapters have 8 "speed dial" entries (numbered 2 to 9) per line (they are on the advanced settings of the "user" tab).  These settings can be set from the web interface, and then called up from your "phone" (you do need "Speed Dial Serv: yes" on the line you wish to use the Speed Dials on) by pressing a digit followed by the "#" key (for example to use speed dial 2, press "2#" on the phone).<br><br>NOTE:  The "digit #" seems to work even if/when your "Dial Plan" doesn't have a translation for that pattern.  Apparently, just enabling the speed dial service, is enough to let the Sipura implicitly interpret the "digit #" sequence as a request for using a speed dial.<br><br>And the really nice thing about these "speed dials", is that they not only work with real numbers (that you could key in on the phone), but also with SIP URIs (full internet addresses for a SIP phone)!  For example, if you setup "Speed Dial 3:" to "613@fwd.pulver.com", then pressing "3#" on the phone will immediately give you the free world dialup "echo test", even if/when FWD isn't the provider programmed into that line!<br><br>And while the speed dials should work (to easily call a SIP URI) with pretty much any Sipura adapter, you appear to have even more options with an SPA-3000.  When I tested it, I was able to use the "@GWx" syntax with the Speed dials on my SPA-3000 (on the advanced "User 1" tab of my adapter).  For example, I put "18005696972@GW2" into my "Speed Dial 4:" entry (as a test), and when I pressed "4#" on the phone, the call (to an AT&T calling card access number) went through VIA my "gateway 2" provider (in my case Teliax.com).  So the speed dial not only worked to call 1-800-569-6972, but it even (correctly) did the call via my "gateway 2" VoIP provider!<br><br>So while there are only 8 of them per line, do remember the Sipura "speed dials", for when you have a specific SIP URI (internet address) you would like to call.  Not only is it easier to put a specific SIP URI into a speed dial then it is to make a "Dial Plan" translation for that URI, but it also doesn't clutter up your "Dial Plan" with translations for single SIP URIs.  <br><br>And even if your Sipura is "locked", the provider may let you into the "Speed Dial" entries of the "User x" tab (even if/when the provider doesn't let you modify the "Dial Plan").  If so, you might be able to enter in your own speed dials, and thereby call any SIP URI you like, even on a supposedly "locked" Sipura adapter...<br><br><B>Voice chatting/conferencing with other BBR VoIP members:</B><br><br>I was thinking.  Is anyone interested in having a BroadBandReports VoIP "conference room", for when we want to chat with each other (instead of just typing at each other)?  <br><br>It occurred to me that SIPphone allows you to setup your own voice conferencing for free, you just have to pick your own 7 digit conference room number (and then use it, no signup required).  And since SIPphone lets you call any of their numbers (including their conference rooms) via SIP URI, you don't even need to sign up with a free SIPphone account to join in the fun.  So what do people think about meeting at SIPphone conference room 227-8647 (aka BBR-VOIP)?  All anyone would need to do to "join in", is to make a VoIP call to SIP URI: 12222278647@proxy01.sipphone.com<br><br>Of course, the easiest way for us Sipura owners to call a SIP URI, is to just put that URI into a "Speed Dial" entry (and make sure the speed dial service is enabled).  For example, if you have "Speed Dial 2:  12222278647@proxy01.sipphone.com", then merely pressing "2#" on your phone would allow you to join the conference.]]></description>
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<pubDate>Mon, 04 Jul 2005 15:38:48 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13805111</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : <div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR><div class="bquote"><SMALL>said by MichiganTelephone:</SMALL><BR><BR>&#8226;Left a phone off hook accidentally? Found that the Sipura's off hook warning tone borders on pathetic? Here's a much better one. This gives you 30 seconds of warning warble tone followed by 30 seconds of the genuine off hook warning tone used by most phone companies: Change Call Progress Tones | Off Hook Warning Tone to <B>480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);30(.1/.1/3+4+5+6)</B><br></DIV>Very nice setting.  But why end the "off hook warning tone" at 30 seconds past the "warning warble"?  For example, what happens if the cat nock's the phone off the hook, while you are at the store?  Wouldn't you want to know about that ASAP when you get home (instead of having the "warning tone" time out after 30 seconds of really annoying sound)?  After all, you can always end the annoying sound by hanging up the phone!  <br><br>So I modified your suggestion, to keep the annoying tone up for 32000 seconds (a little under 9 hours of the loud annoying "you left the phone off the hook" sounds):<br><div class="code"><PRE><span class="codetext">480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);32000(.1/.1/3+4+5+6)</SPAN></PRE></DIV></DIV>I got to thinking about this and realized that you can have a truly indefinite tone if you want - just replace the "32000" in your string above with * (a single asterisk), so it would look like this:<br><div class="code"><PRE><span class="codetext">480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);*(.1/.1/3+4+5+6)</SPAN></PRE></DIV><br>I added a notation about this on <A HREF="http://michigantelephone.mi.org/distribute.html">my web page</A>, and also a section about how to increase the volume of the dial tone and other tones.  I would suggest that under the Regional tab, you look in the "Change Call Progress Tones" section and if you see any volume levels of <B>@-19</B> (or anything less than @-16) you try changing them to <B>@-16</B> - note there are usually two or more such values in each line.  Only do that if you think the present tone volumes are a bit low (I actually do think they are a bit lower than the standard, but that's just a subjective observation) and make sure you change all the volume levels shown in each line if you change any of them.<br><br>Since readers of this forum are a bit more technically-mined than most, I'll just point out that you can probably do a lot of specific tricks with the tones if you like.  For example, if you have multi-line phones in a small-office setting, you could change the frequency of the dial tone coming out of the Sipura to a much higher-pitched tone (try frequencies in the 600 to 800 Hz range, for example you could try something like 620 and 660 Hz) and train people that the line(s) with the "normal" dial tone are to be used for local calls only and all 911 calls, and the line(s) with the strange, high-pitched dial tone are to be used for all toll calls.  Be careful when picking frequencies so that you don't inadvertently pick one that's too close to a frequency found in one of the <A HREF="http://www.shout.net/~wildixon/telecom/dtmf/dtmf.html">touch-tone pairs</A>, such as 697 Hz or 770 Hz.]]></description>
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<pubDate>Sun, 03 Jul 2005 15:21:42 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13804927</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  ags <A HREF="/useremail/u/463209"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>I have a Sipura 2002 behind a router and get an internal IP<br>(192.168.0.2) from that router.<br><br>I have a friend in another city who has service from Vonage. His vonage box is behind a router.<br><br>How can I call him by dialing his external IP and have his Vonage box attached phone ring and start a conversation.</DIV>Vonage used to allow people to call their customers by SIP URIs (internet addresses).  However, I think they may have turned that feature off at their end, at the same time they dropped the FWD=>Vonage gateway.  And since Vonage "locks" their adapters, there is no way for your friend to get into his/her adapter and override Vonage's settings.<br><br>[Edit]Does anyone with Vonage know if Vonage still allows SIP URI's for calling their subscribers?  If so, what is the format of the needed SIP URI for calling a Vonage subscriber?<br><br>In the off chance that Vonage still allows SIP URI calls to their customers, and you can figure out what the proper SIP URI is for your friend's Vonage number/account, it would be pretty trivial to modify your "dial plan" (in a way similar to my FWD calling trick) to call your friend's Vonage SIP URI.  But this will ONLY work if Vonage still allows this feature.  If I'm correct that Vonage has turned this feature off at their end, than there is nothing you can do to override Vonage's settings (as SIP to SIP calls require both sides to allow them)!<br><br>NOTE:  If Vonage has blocked things on their end, you could always try convincing your friend to sign up with FWD directly.  If your friend was willing to spend $100 for a Sipura SPA-3000, (s)he could even hook up the Vonage adapter to the "Line" port of the SPA-3000.  With proper SPA-3000 configuration, this arrangement (Vonage adapter off the SPA-3000's line port, and then the "phone" plugged into the SPA-3000) should allow your friend to use the same "phone" to  make/receive normal LD calls via the Vonage adapter (because an SPA-3000 can "pass through" the "Line" port to the "phone") AND also make/receive calls via VoIP accounts programmed directly into the SPA-3000.  Of course, once this setup is ready, it's trivial for your friend to signup with FWD, and program his/her FWD account as one of the VoIP providers in his/her SPA-3000 (allowing the two of you to chat as long as you want FWD account to FWD account)!]]></description>
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<pubDate>Sun, 03 Jul 2005 14:51:17 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13804018</link>
<description><![CDATA[<A HREF="/useremail/u/463209"><b>ags</b></A> : I have a Sipura 2002 behind a router and get an internal IP<br>(192.168.0.2) from that router.<br><br>I have a friend in another city who has service from Vonage. His vonage box is behind a router.<br><br>How can I call him by dialing his external IP and have his Vonage box attached phone ring and start a conversation.<br><br>To recap, How can I dial from my Sipura 2002 to connect with his Vonage box connected phone by dialing his external IP.<br><br>I do have FWD and my friend does not. It use to be FWD and Vonage has peering agrrement and I can call him using my FWD account. Unfortunately the peering agrement between FWD and Vonage has been terminated.<br><br>Any suggestion will be highly appreciated.]]></description>
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<pubDate>Sun, 03 Jul 2005 11:58:12 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13792326</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Regarding the dial tone volume:  There is a volume level (I think it's in decibels, but don't hold me to that) associated with each frequency of a tone.  For example, I had suggested using a dial tone value of:<br>350@-19,440@-19;20(*/0/1+2)<br>The "-19" that appears (twice) in the above string is the volume.  Normally people can't hear a volume change of less than about 3 db, so you could try changing both instances of "-19" to "-16", or "-13", or any higher volume level.<br><br>Now, regarding how these strings are constructed.  As best I can determine, here's how it works.  Take the off hook tone I suggested (and by the way, I limited the loud warning to 30 seconds because that's what phone companies usually do these days - I think the theory is that sometimes people take the phone off-hook on purpose and they don't want to be annoyed by the tone for the entire time.  After all, certain activities for which you might spontaneously take the phone off the hook might be somewhat inhibited by a loud off-hook warning going off in the background) ;) - recall it was this:<br>480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);30(.1/.1/3+4+5+6)<br><br>So to dissect this:<br>480@-10 is one of the tones of the initial warning warble (480 Hz at -10dB??? volume)<br>620@-16 is the other tone of the initial warning warble<br>(the volume levels are different because the human ear needs the lower tone to have more energy to perceive it as being at about the same level as the higher tone).<br><br>The remaining four values are the frequencies and volumes of the loud off-hook warning.  I used 0 dB because we want them to be LOUD.  Possibly they could be made even louder by using positive dB values, but I didn't want to blow out a phone's earpiece (or anyone's eardrum) and 0 dB just felt and sounded like the right value.<br><br>Now, note that the given order of the tones corresponds to the number we use to indicate them later in the string.  For example the 480 Hz tone appears first in the string so it is "tone #1", the 620 Hz tone indicates "tone #2", etc.  The first semicolon marks the end of the tone definitions and moves us to the first of the duration specifications.<br><br>Next we have 30(.2/0/1,.2/0/2) - the 30 is the total duration in seconds.  The .2/0/1 means .2 seconds on, zero seconds silence (we want the next tone to follow immediately) of tone #1 (the 480 Hz tone).  The .2/0/2 means .2 seconds on, zero seconds silence, of tone #2 (the 620 Hz tone).  480 and 620 just happen to be standard frequencies used in telephony, though you could certainly use others.<br><br>Next another semicolon marks the end of that duration specification.  Next we have the final duration specification, for the real on-hook blast:<br>30(.1/.1/3+4+5+6)<br><br>And this means we want thirty seconds (and yes, you can change that) of the four tones (#3, #4, #5, and #6), which is to say, the 1400 Hz, 2060 Hz, 2450 Hz, and 2600 Hz tones, with duration of .1 seconds on followed by .1 seconds of silence.<br><br>Note that in the case of a continuous tone, such as a dial tone that we simply want to run for 20 seconds, it's specified like this:<br><br>20(*/0/1+2)<br><br>The asterisk simply means "on for the full specified duration, zero silence"<br><br>I do not know if there is a maximum length to the string, nor if there is a maximum number of frequencies that can be specified in one command.  Maybe you could make the dial tone play a little tune if you were sufficiently motivated, but my goal was to get these tones to the standards used in the U.S.A. and Canada.<br><br>I hope this helps those of you who want to customize your tones.  Just remember not to get so clever that you confuse others living in, or visiting your home!]]></description>
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<pubDate>Fri, 01 Jul 2005 17:17:46 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13788700</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  random1739 <A HREF="/useremail/u/1125260"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>I've got a SPA-2000. Is there a way to increase ring tone/dialling volume from the voice box? Or doesn't the voice box have a volume control?<br><br>Currently, the dial tone is extremely quiet and difficult to hear :huh:<br> </DIV>Probably, but we would have to figure out the correct settings.<br><br>Based upon MichiganTelephone's post, it appears that all the "tones" are very configurable (pattern, volume, and duration) from the "Call Progress Tones" section of the "Regional" tab.  The problem is, what is the format of those tone strings (and how long can the tone string fields be)?  I didn't see that format info anywhere in my Sipura manual, so at the moment I can only guess.  <br><br>With the proper format for that "tone info", we should (at least in theory) be able to change the tones pretty much however we want.  So if anyone finds a reference to the format used by Sipura adapters to describe the "tone" entries, please post a link to that reference here.  <br><br>[Edit]<br>Check the note (by MichiganTelephone) below!  It looks like we now have the details necessary to construct the "Call Progress Tones" strings.  So all that should be necessary to have a louder dial-tone, is to adjust the volume setting of the tones in the "Dial Tone:" string.<br><br>[Edit2]<br>It works.  Use the description in the note below to find the tone volume settings (expressed as negative numbers) within the strings, and raise the volume (by lowering the negative numbers).  For example, I went though and changed many of the -19 volume tone entries to -17 (on my SPA-3000), to raise the volume "just a little".  Obviously you can continue to raise the volume (by using progressively smaller negative numbers) to whatever volume level you find pleasing.]]></description>
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<pubDate>Fri, 01 Jul 2005 09:21:58 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13787612</link>
<description><![CDATA[<A HREF="/useremail/u/1125260"><b>random1739</b></A> : I've got a SPA-2000. Is there a way to increase ring tone/dialling volume from the voice box? Or doesn't the voice box have a volume control?<br><br>Currently, the dial tone is extremely quiet and difficult to hear :huh:]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13787612</guid>
<pubDate>Fri, 01 Jul 2005 02:31:14 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13787447</link>
<description><![CDATA[<A HREF="/useremail/u/1223723"><b>gnexus</b></A> : <div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>The fact that you can talk out, but you can't hear, often means a "firewall" (router) issue. What often happens, is that the router's firewall (and NAT) "blocks" (doesn't forward to the Sipura) inbound voice packets. Since inbound voice packets are blocked at your router, you can't hear the other party. But since most routers (by default) let any packet go out, your voice packets make it though your router to the other party (and they can therefore hear you). The classic "one way audio".<br></DIV>This is another reason, besides QOS, why I was thinking of getting an unlocked Linksys router. I even had this exact problem with the Vonage Linksys and my XTen softphone and had to map ports.<br><br>I suppose you don't know which version of Sipura ATA the Linksys have? I know it's certainly not SP-3000 functionality, but is it maybe SP-2001? Any help would be greatly appreciated. <br><br>Thanks again for your awesome posts. I plan to use this info when I get my ATA. ]]></description>
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<pubDate>Fri, 01 Jul 2005 01:45:18 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13785624</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by MichiganTelephone:</SMALL><BR><BR>I have a few Sipura tricks listed on the page, <A HREF="http://michigantelephone.mi.org/distribute.html">How to Distribute VoIP Throughout a Home</A></DIV>Wow!  Those are some nice (and really NOT obvious) settings.  I've already added several of those tips to my SPA-3000.  Thanks for posting.<br><br><div class="bquote"><SMALL>said by MichiganTelephone:</SMALL><BR><BR>&#8226;Left a phone off hook accidentally? Found that the Sipura's off hook warning tone borders on pathetic? Here's a much better one. This gives you 30 seconds of warning warble tone followed by 30 seconds of the genuine off hook warning tone used by most phone companies: Change Call Progress Tones | Off Hook Warning Tone to <B>480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);30(.1/.1/3+4+5+6)</B><br></DIV>Very nice setting.  But why end the "off hook warning tone" at 30 seconds past the "warning warble"?  For example, what happens if the cat nock's the phone off the hook, while you are at the store?  Wouldn't you want to know about that ASAP when you get home (instead of having the "warning tone" time out after 30 seconds of really annoying sound)?  After all, you can always end the annoying sound by hanging up the phone!  <br><br>So I modified your suggestion, to keep the annoying tone up for 32000 seconds (a little under 9 hours of the loud annoying "you left the phone off the hook" sounds):<br><div class="code"><PRE><span class="codetext">480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);32000(.1/.1/3+4+5+6)</SPAN></PRE></DIV><br><div class="bquote"><SMALL>said by MichiganTelephone:</SMALL><BR><BR>Sorry if this is a bit long, but it fits in this thread and I thought it might be useful for someone.<br> </DIV>IMHO your post fits this thread very well.  Thanks for tips.]]></description>
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<pubDate>Thu, 30 Jun 2005 21:31:39 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13785319</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by balta:</SMALL><BR><BR>Well it almost work :-) Thanks a lot for the advance...<br><br>As per your instructions, I managed to get media through : unfortunately only one way : from home to the outside. From the phone I don't seem to send the udp packets for the sipura.</DIV>The fact that you can talk out, but you can't hear, often means a "firewall" (router) issue.  What often happens, is that the router's firewall (and NAT) "blocks" (doesn't forward to the Sipura) inbound voice packets.  Since inbound voice packets are blocked at your router, you can't hear the other party.  But since most routers (by default) let any packet go out, your voice packets make it though your router to the other party (and they can therefore hear you).  The classic "one way audio".<br><br>First check for the obvious, and triple check your Sipura settings (including making sure that the Sipura Nat settings are "yes" for the line/gateway entries, assuming you are behind a NAT router).  I remember I was getting "one way audio" (even after setting up STUN), until I discovered that I had forgot to set "NAT Mapping Enable: yes" on the "Line 1" tab!  So double-check your Sipura settings first!<br><br>However, if the NAT/STUN settings are proper in the Sipura, the most likely culprit is your router's handling of inbound packets on the the SIP ports.  Assuming it's a router issue, you may be able to figure out what is wrong by looking at the router logs.  Does your router allow "syslog"?  If so, turn on the router's log feature, and then see which packets are being dropped at the router's firewall (when you are getting the one way audio).  Chances are, those are the ports you need to forward to your Sipura, to get things working...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13785319</guid>
<pubDate>Thu, 30 Jun 2005 20:55:52 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13784800</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Well it almost work :-) Thanks a lot for the advance...<br><br>As per your instructions, I managed to get media through : unfortunately only one way : from home to the outside. From the phone I don't seem to send the udp packets for the sipura.<br>Everything works when its the main provider, so it must be some trick with the receiving port. Although I have the SIP syslogd files  I can't get it clear what it might be. If you're available to help me, send me an email and I can show you the log snippet where the reason may be.<br><br>Thanks again]]></description>
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<pubDate>Thu, 30 Jun 2005 19:50:25 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13784296</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by balta:</SMALL><BR><BR>I've two providers :one [Provider A] gives me a PSTN number, the other [Provider B] doesn't. Nonetheless provider B has better rates, but unfortunatelly seems to require an outbound proxy.<br><br>The problem I get at this stage is that GTWn dial plan aliases do not allow oubound proxies (only the proxy) part, so I can't set Provider A on line 1, and and Provider B as a dialplan gateway.</DIV>Hmmm...  That's a tuff one.  So far the "outbound proxy" is optional with all my providers.  Some of them offer it, but you don't "need it" if/when STUN is setup correctly.  So I can usefully use the "gateway" options, since I leave "outbound proxy" (not to be confused with the SIP gateway) blank for all my providers.  <br><br>You might try setting up your "provider B" without an "outbound proxy" (even though "provider B" tells you that you "need" an outbound proxy), but with the following STUN settings (which I use), and see if it works (worth a try, in any event):  "STUN Server:  stun.fwdnet.net", "STUN Enable: yes", "Substitute VIA Addr: yes", "Send Resp To Src Port: yes", and "NAT Mapping Enable: yes" on the line1 tab (with all other STUN settings off).  It also wouldn't hurt to tell your router to forward your SIP ports to your SPA-3000.  These are the settings I use, and they essentially make it look like (to the remote providers) like your Sipura is not behind a NAT router (even when it is).  With these settings I have only needed the normal SIP proxy (not the optional "outbound proxy") with all the providers I use (which doesn't mean that your "provider B" might not be the exception to this rule).  Don't know if it will work in your case or not, but it is worth a try "just in case", since if the experiment does work you have a way to put "provider B" on one of the 4 "gateway" entries (and thereby making it trivial to do what you desire)!  <br><br>If the above (valid STUN settings, but no "outbound proxy") doesn't work with "provider B", my next question is:  Does "provider A" give you the option to forward inbound calls to a SIP URI (internet address)?  If so, I have a theory about how you might be able to use this "forwarding" to let "provider A" ring "Line 1", even if/when you put "provider B" (and it's outbound proxy) on the "Line 1" settings.  Again I'm not sure if this will work or not (I can't test it by myself, so PM me if you are interested in setting up some tests for this), and it will (in any event) only work if/when "provider A" allows SIP forwarding.<br><br>Finally, you might want to look at my previously posted SPA-2000 trick for forwarding one line to the other.  I have NOT tested the "forward one line to the other" trick on my SPA-3000, so YMMV.  However, if that trick can be made to work on an SPA-3000, you could (in theory) put "provider A" (the inbound one) on the "PSTN" line, forward the "PSTN" line to "Line 1" (of the same adapter), and then put "provider B" (complete with it's outbound proxy) on "Line 1".  I have no idea if this convoluted setup will work or not (without running some experiments), but it would (in theory) be a way to have both providers on the same adapter with different "outbound proxy" settings.<br><br>And if all else fails, you might also try asking for help in the Voxilla.com Sipura forum here: &raquo;<A HREF="http://voxilla.com/forum-viewforum-f-14.html" >voxilla.com/forum-viewforum-f-14.html</A><br><br><B>NOTE:  If you do find a solution to the issue of different "outbound proxy" settings on an SPA-3000, please post it here.  I'm sure other people would like to know what "trick" is needed to make things work in that case!</B>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13784296</guid>
<pubDate>Thu, 30 Jun 2005 18:36:08 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13784062</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : I have a few Sipura tricks listed on the page, <A HREF="http://michigantelephone.mi.org/distribute.html">How to Distribute VoIP Throughout a Home</A>, which I will copy here so you don't have to search through the page to find them (they are near the bottom of the page).  This is copied verbatim from the page so ignore phrases like "as mentioned earlier" since they don't apply here:<br><br><BLOCKQUOTE><B>Additional hints for emulating "real" telephone service (<U>Sipura</U> <U>adapters</U> <U>only</U>, may also apply to Linksys adapters that use Sipura technology)</B><br><br>In this section we will just mention some Sipura adapter defaults that you or your service provider may want to change in order to provide service that better emulates regular wireline telephone service. Except perhaps for the first, these are NOT essential settings, and none of them in any way affect call quality. To change these settings, go to the web interface for your Sipura adapter (enter the local IP address for the Sipura in web browser), then click on "advanced" in the upper right corner of the screen, then click on the "Regional" tab. If this tab is not visible you may need to do an "Admin Login", which requires a password, or you may need to ask your VoIP provider to make the desired changes for you. Please note that the tone settings shown below are for the United States and Canada only.<br><UL><br>&#8226;In case you missed it above, in the U.S. and Canada, under Ring and Call Waiting Tone Spec, the Ring Voltage should be set to <B>90</B> and (this is most important) the Ring Frequency to <B>20</B> &#151; this not only allows older phones with mechanical bells to work, but it just might help in a few odd cases where Caller ID doesn't seem to work properly on a particular phone. In fact, if you have any weird problems with equipment that worked fine with traditional phone service not working with VoIP, and that equipment is activated by a ring signal, this may be the problem. Sipura adapters default to a ring frequency of 25 Hz, which is NOT the frequency usually used in the United States and Canada.<br>&#8226;Also as mentioned earlier, under the Control Timer Values (sec) section, we suggest setting CPC Delay to <B>10</B> and CPC duration to <B>1</B>, because if you have one or more phones with a "hold" button and you ever put a call on hold and then no one picks it up, this will release the hold (freeing the phone line) when the caller hangs up. Note this will not help if you accidentally leave an outgoing call on hold &#151; at present the Sipura doesn't have any good way to release outgoing calls accidentially left on hold automatically (perhaps Sipura might consider adding this in a future firmware release &#151; it would great if they would add an "Off Hook Warning Disconnect" signal, which would be like a CPC disconnect, except that it would activate just before the Off Hook Warning Tone plays).<br>&#8226;Lengthen the dial tone to 20 seconds (some people find the default 10 seconds too short): Change Call Progress Tones | Dial Tone to <B>350@-19,440@-19;20(*/0/1+2)</B><br>&#8226;Lengthen Second Dial Tone, Outside Dial Tone, Prompt Tone, Busy Tone, Reorder Tone, MWI Dial Tone, and Cfwd Dial Tone to 20 seconds: These settings, like the basic dial tone mentioned above, are under Call Progress Tones - in all the existing strings find <B>;10(</B> and change it to <B>;20(</B><br>&#8226;Left a phone off hook accidentally? Found that the Sipura's off hook warning tone borders on pathetic? Here's a much better one. This gives you 30 seconds of warning warble tone followed by 30 seconds of the genuine off hook warning tone used by most phone companies: Change Call Progress Tones | Off Hook Warning Tone to <B>480@-10,620@-16,1400@0,2060@0,2450@0,2600@0;30(.2/0/1,.2/0/2);30(.1/.1/3+4+5+6)</B><BR>Note that if for some reason you don't want the warning warble (which we highly recommend because it will cause most people to hang up before being blasted with the off-hook warning), you can use <B>1400@0,2060@0,2450@0,2600@0;30(.1/.1/1+2+3+4)</B><BR>Obviously, you should not make either of these changes if you made the change to generate the "D" touch tone prior to the Off Hook Warning Tone, <STRIKE>as described under "Special considerations for business customers" above.</STRIKE> <I>[The pertinent section is copied below]</I><br>&#8226;If you have an answering machine or similar device that accepts touch tones for control functions, and you find that when you call in and try to use tones to activate the unit it does not respond properly, check under the Miscellaneous section to see what the DTMF Playback Level and the DTMF Playback Length are set to. The default DTMF Playback Level is <B>-10.0</B> which is often too low, while the default DTMF Playback Length is <B>.1</B> which is very often too short.<br>&#8226;A few people may wish to go into the individual "Line" tabs ("Line 1" and "Line 2") and go to the FXS Port Polarity Configuration section, and set Caller Conn Polarity to <B>Reverse</B>. All this will do for most people is give you an audible "click" when a call you place is connected, and again when the called party hangs up. What it actually happening is that the polatity of the line is reversed when the outgoing call is connected. Some advanced phone systems may be able to use this information to avoid the "line left on hold" problem, but most "hold" buttons on telephones will NOT release just because line polarity reverses (you may be able to build a circuit that responds to polarity reversals and generates a CPC disconnect signal after a polarity reversal). Note that if for some reason you want the line polarity to reverse when you are the called party, then you will need to set the Callee Conn Polarity to <B>Reverse</B>.<br>&#8226;Finally, if you have changed any of the above settings (as opposed to your service provider), you will want to make sure that you go to the "Provisioning" tab and set Provision Enable to <B>no</B>. However, most commercial service providers will not allow you to do this, since it prevents them from making changes to your service (which could overwrite the changes you have made above). That is why, if you want any of the above settings changed and you are a commercial VoIP service customer, it is better to get your service provider to make the changes in their system so that any future updates will not overwrite the changes.</UL>By the way, one way to avoid a line left on hold is to not use a hold button on a telephone, but instead use "flash hold" &#151; hit the flash button (or flash the hookswitch), then when you hear the second dial tone, hang up. Your phones will emit a short ring every few seconds until you pick the line back up, so unless you have the ringers turned off, it will be pretty hard to forget about the call on hold. This gives you a way to put a call on hold on one phone, and then pick it up at another. Some VoIP providers may disable this capability, though we don't know of any that do, and also we do not know if this works with any brand of VoIP adapter other than Sipura.</BLOCKQUOTE><br><br>One other hint appeared earlier on the page, but it's it's for business customers trying to use VoIP with a small PBX.  This is slightly abbreviated from what's on the page (residential customers can probably skip this entirely, unless you have a phone with a "hold" button and someone keeps leaving the line on hold):<br><BLOCKQUOTE>..... certain VoIP adapters do not generate what is known as the CPC signal. Without getting into a long technical dissertation, this is a momentary drop in power on the line (as if the line were completely disconnected for a few moments), or a polarity reversal on the line. Either of these signals can be used to cause telephone hold circuits to release automatically. The CPC signal (but not the polarity reversal) can also be useful with some consumer grade equipment, including telephones with "hold" buttons and some types of answering machines. However, most residential customers won't notice much of an impact from the lack of the CPC signal.<br><br>On a "normal" telephone line, this voltage drop or polarity reversal usually occurs if you hold the line open after the other party has hung up (to a person listening on the line, it sounds like either just a single click, or a click, a VERY short pause of "dead air", and another click. But in either case, it happens very quickly so if you're not paying close attention you could easily miss it!).<br><br>As mentioned, the purpose of this signal is to release a line inadvertently left on "hold." So, when connecting a small business phone system that is designed to work with regular business phone lines to a VoIP adapter that does not supply the CPC signal, the problem appears to be that if a call is accidentally left on hold, the line will never release. In some slightly more expensive systems, it's possible to "park" a call while waiting for another extension to pick it up, and a "parked" call might <I>never</I> be released without manual intervention. Thus, probably with distressing regularity, the line gets held open because someone put a call on hold and forgot about it, and with no CPC signal the line was never released.<br><br>If you think you have this problem, contact your VoIP provider &#151; they may be able to enable CPC on your adapter. For example, Sipura adapters have CPC settings under the "Regional" tab &#151; try setting "CPC Delay" to 10 and "CPC Duration" to 1 &#151; your VoIP provider may have locked out these settings; if so, they would have to change these settings for you. But note that Sipura's CPC functions only on <B>incoming</B> calls; it will do nothing to help a situation where an <B>outgoing</B> call has been abandoned on hold. If you really need a CPC-like disconnect following outgoing calls using a Sipura adapter, there is one thing you could try, if you have access to the Sipura's advanced settings. Go to the "Regional" tab, under Call Progress Tones, and set the Off Hook Warning Tone to <B>941@-16,1633@-16,1400@0,2060@0,2450@0,2600@0;1(*/0/1+2);30(.1/.1/3+4+5+6)</B><BR>This will generate one second of a "D" touch tone (941 Hz + 1633 Hz) just prior to playing the Off Hook Warning Tone. The "D" tone cannot be generated from a standard 12-button touch tone pad, but can be used directly by some types of equipment as a disconnect signal. If your equipment cannot be directly programmed to release a line upon hearing the "D" tone, you can buy a "DTMF Flash Generator" that should disconnect the line upon hearing the "D" tone (see next paragraph).<br><br>Assuming that none of the above suggestions work, we can point you to a place where you can probably find a solution to this problem, but please be aware that we've not personally tested any of this equipment. Reports from actual users of this equipment would be welcome. Go to Mike Sandman's <A HREF="http://www.sandman.com/wizard.html">"Wizard's Tool Box" page</A> and scroll about &frac14; of the way down the page &#151; search for the items entitled "CPC GENERATOR", "MAKE A SILENCE DETECTOR, TO CREATE A CPC SIGNAL", and "CallSaver: Disconnects a Phone Line that's Left Off Hook!" ONE of these devices will probably solve your problem, depending on the VoIP adapter in use and how it actually reacts when a phone is left off-hook after the call is ended (for example, VoicePulse returns a fast busy signal when the other end disconnects, and Vonage <B>sometimes</B> does this as well, so from the descriptions given it is likely that the "CPC GENERATOR", which detects a dial tone or a busy signal, would be the device to use to create the CPC signal). If you have a Sipura adapter and can reprogram it to generate the "D" touch tone as described in the above paragraph, then the "DTMF Flash Generator" (not quite halfway down Mike's page) may be able to listen for it and generate a useable CPC signal.</BLOCKQUOTE><br>Sorry if this is a bit long, but it fits in this thread and I thought it might be useful for someone.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13784062</guid>
<pubDate>Thu, 30 Jun 2005 18:01:24 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13783646</link>
<description><![CDATA[<A HREF="/useremail/u/0"><b>anon</b></A> : Hi.Just received my SPA-3000 today and need somehelp to configure my scenario.<br><br>I've two providers :one [Provider A] gives me a PSTN number, the other [Provider B] doesn't. Nonetheless provider B has better rates, but unfortunatelly seems to require an outbound proxy.<br><br>The problem I get at this stage is that GTWn dial plan aliases do not allow oubound proxies (only the proxy) part, so I can't set Provider A on line 1, and and Provider B as a dialplan gateway.<br><br>On the other hand I don't seem to be able to forward all VOIP2 calls to the FXS phones. Any help ?<br><br>Thanks]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13783646</guid>
<pubDate>Thu, 30 Jun 2005 17:08:36 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13781494</link>
<description><![CDATA[<A HREF="/useremail/u/651075"><b>devil24</b></A> : Thanks a lot for your reply :).]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13781494</guid>
<pubDate>Thu, 30 Jun 2005 12:38:33 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13780644</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  devil24 <A HREF="/useremail/u/651075"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>Now, can this 'multiple providers in one line' trick work on a SPA-2002??? if so, can you please post a more detailed example of how to set it up???.<br><br>I'm currently subscribed to 3 services, all running from my SPA-2002 and having to erase/edit any of the 2 active ones every time I want to use the other one is a PITA,</DIV>I don't have an SPA-2002, but as far as I know the SPA-2002 has pretty much the same features as the SPA-2000 (which I do have).  <br><br>So I think the answer is that the SPA-2002 can do about as much as the SPA-2000 can.  Specifically, that would mean that providers that let you call their subscribers without authorization (such as FWD or SIPphone) can be used with appropriate "Dial Plan" tricks (shown above).  But as far as I'm aware, you need an SPA-3000 to use any of the "gateway" tricks that allow multiple providers that require a username/password on the same "phone".<br><br>BTW:  It was the "gateway" (multiple providers WITH authorization, on the same "phone") features, that were a strong reason why I recently purchased my SPA-3000 (from voxilla.com), despite the fact that I already owned a properly functioning SPA-2000.  The SPA-3000 really does give you more options than many of the other Sipura models.  OTOH it's not as if my investment in my SPA-2000 is totally wasted either.  I can still use the SPA-2000 if/when I ever want to setup a "remote extension", and I will probably also make use of it if/when I ever setup an * box.  And I may even experiment (at some point) with hooking the SPA-2000's ports into the "Line port" of the newer SPA-3000 (doing so should "in theory" allow for even more providers on the same "phone", than you can do with just the SPA-3000 by itself).<br><br>[EDIT]:  As to your example, here goes.  Just keep in mind that the "gateway" features are SPA-3000 specific (and won't work on say an SPA-2000)!<br><br>You first setup your primary "Line 1" provider normally (including "registration" with their SIP proxy).  Choose your primary provider carefully, as that is the only provider your SPA-3000 will allow inbound calls from (unless you do clever tricks with forwarding and/or hooking up another ATA to the "Line port").  In my case, I use FWD as my "primary", as all my pay providers are currently not supplying inbound DIDs (and so FWD is the only provider I currently need to "ring" the phone).<br><br>For the other 4 outbound providers, you setup the "Line 1" (again SPA-3000 only) providers as follows:<br>The "Gateway x:" field gets "userid@proxy" (NOTE:  It's non-obvious from the docs, but you really need the "userid @ the_proxy_address" in the gateway field, not just the proxy address!).  The "GWx Auth ID:" ID gets your userid (yes, you need it in this field by itself, and part of the gateway field info).  The "GWx NAT Mapping Enable:" field gets whatever your desired NAT setting is (in my case "Yes").  And the "GWx Password:" field obviously gets the password for that provider.<br><br>Once you have the Gateway fields (there are 4 of them, allowing for up to 4 additional "authorized" outbound providers) filled in, the only other step is to modify the "Dial Plan" to pick when you want to use that "gateway".  For example, I have DialPad.com in my "gateway 1", so I use the following to send most LD calls to DialPad:<br><div class="code"><PRE><span class="codetext"> 1&#91;2-9&#93;xx&#91;2-9&#93;xxxxxxS0 &lt;:@GW1&gt;</SPAN></PRE></DIV>Likewise, Telix has really good quality phone calls, and they don't charge for "toll free" calls (on my "pay as you go" plan).  So I auto-route 800/888/877/866 calls to Teliax (which is on "gateway 2") by the following:<br><div class="code"><PRE><span class="codetext">1 800 &#91;2-9&#93;xxxxxx S0 &lt;:@GW2&gt; | 1 888 &#91;2-9&#93;xxxxxx S0 &lt;:@GW2&gt; | 1 877 &#91;2-9&#93;xxxxxx S0 &lt;:@GW2&gt;<br> * | 1 866 &#91;2-9&#93;xxxxxx S0 &lt;:@GW2&gt;<br><br>(*) WARNING 1 long line(s) split</SPAN></PRE></DIV><br>BTW:  While I haven't played with it yet, the "Line port" is considered "gateway 0".  So if you wanted to send 911 calls to the "Line port" (for example, if you hooked up the line port to a POTS line), you could probably use the following in your "Dial Plan":<br><div class="code"><PRE><span class="codetext">911 S0 &lt;:@GW0&gt;</SPAN></PRE></DIV>NOTE:  I think the 911 example is correct, but as I've already mentioned, I haven't tested it yet!<br>]]></description>
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<pubDate>Thu, 30 Jun 2005 10:26:47 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13779073</link>
<description><![CDATA[<A HREF="/useremail/u/651075"><b>devil24</b></A> : <div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>I just got my SPA-3000 today, and (while the process was more painful than expected) I've already figured out how to have both my DialPad.com ($11.99/month) and my Teliax.com (pay as you go) accounts running on "Line 1" (and both accounts have different SIP passwords).<br><br>Now if I could just find a reliable way to override the "User ID:" field (the "GWx Auth ID:" fields don't seem to behave as expected in this regard), I would be able to include my other SIP accounts that were done under a different userid (in addition to accounts with a different proxy and password)...<br><br>[edit] Just got the solution to my SPA-3000 problem from the Voxilla forums.  Instead of putting in my provider's "proxy_address" (i.e. 66.35.222.58 for DialPad.com) in for the value of "Gateway x:" and my userid for the value of "GWx Auth ID:", you put in "userid@proxy_address" (i.e. myaccount@66.35.222.58 for DialPad) for the "Gateway x:" field, and then also put in your "userid" for the "GWx Auth ID:" field.  You apparently also need a "new enough" version of the firmware for this to work (my SPA-3000 has version "2.0.13(GWg)", which works fine with this "trick").  <br><br>But once you get this "trick" to work, you can use the 4 "gateway" fields to totally override your userid/password/proxy settings of "Line 1" (of an SPA-3000).  This allows you to easily have line 1 setup for 5 different VoIP providers (the default "Line 1" one being by-directional, and the 4 "gateway" ones being "outbound only").  And those 5 providers are on top of any providers (such as FWD) that you can call out to without "authorizing" (i.e. using the previous described "@proxy" trick to the dial-plan).<br><br>BTW:  At the moment I only have 3 providers, so I have some room for expansion.  My "Line 1" default is currently setup for FWD (this is only so that I can receive inbound calls by FWD, otherwise I would have used the previous FWD outbound "trick"), my GW1 is setup for DialPad.com ($11.99/month "unlimited", and also the provider my dial plan selects for LD calls), and GW2 is my "pay as you go" Teliax account (default dialing for toll free numbers, and can be explicitly used by dialing "# 8 call_digits #").<br> </DIV>Hey DracoFelis, great job, pal!<br><br>Now, can this 'multiple providers in one line' trick work on a SPA-2002??? if so, can you please post a more detailed example of how to set it up???.<br><br>I'm currently subscribed to 3 services, all running from my SPA-2002 and having to erase/edit any of the 2 active ones every time I want to use the other one is a PITA, so, if this is able to work on my device, it'll make things much much easier for me :).<br><br>Thanks in advance for your help. ]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13779073</guid>
<pubDate>Thu, 30 Jun 2005 02:13:33 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13778989</link>
<description><![CDATA[<A HREF="/useremail/u/1223723"><b>gnexus</b></A> : This stuff is awesome Draco. I never knew a Sipura could do all this! <br><br>I'm wondering whether the unlocked Linksys routers would have this kind of capability, since the ATA is a Sipura?<br><br>I think really it would be better having a separate ATA, however. That way you can upgrade the router or the ATA to add more features. The only thing is, I want a new 802.11g router. Then I can use my iPaq with an XTen softphone as a TV remote <I>and</I> a cordless. :D The unlocked linksys has QOS and it is the cheapest way to go. Provided, of course, if I can get away with buying one wholesale. Linksys shouldn't be allowed to make that such a PITA. Especially since they just bought Sipura.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13778989</guid>
<pubDate>Thu, 30 Jun 2005 01:50:06 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13777933</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  tlpintpe <A HREF="/useremail/u/687883"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>I think it was me trying to just cut and paste from your entries (dialing toll free numbers via FWD) that was the culprit. There is a carrage return in there, and when I edited the dial plan in a text editor, and removed the carrage return, then reentered the dial plan, all worked as it should.</DIV>Yep.  A "dial plan" for a Sipura should all be on <B>one line</B>.  Please ignore any extra "line breaks" in the examples.  They are simply because BBR doesn't support 2K long text lines (without wrapping), whereas the Sipura dial plan does...<br><br>While we are on the subject of "dial plans", here are a couple of cute ones.  Since <A HREF="http://www.sipphone.com/">SIPphone.com</A> accepts inbound "peering", you can directly call any SIPphone.com account (which all have numbers in the form:  1 747 xxx-xxxx) by adding the following to your Dial Plan (before your normal LD pattern):<br><div class="code"><PRE><span class="codetext">1 747 xxx xxxx &lt;:@proxy01.sipphone.com&gt;</SPAN></PRE></DIV>And potentially even more interesting, is that you can use the <A HREF="http://sipphone.com/conference/">"phone conferencing"</A> ability of SIPphone (even if you don't have a SIPphone account), by adding the following to your Dial Plan (again before your normal LD pattern):<br><div class="code"><PRE><span class="codetext">1 222 xxx xxxx &lt;:@proxy01.sipphone.com&gt;</SPAN></PRE></DIV>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13777933</guid>
<pubDate>Wed, 29 Jun 2005 22:55:06 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13752778</link>
<description><![CDATA[<A HREF="/useremail/u/687883"><b>tlpintpe</b></A> : <div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR><div class="bquote"><SMALL>said by  tlpintpe <A HREF="/useremail/u/687883"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>Is there are 255 character limit per dial plan?</DIV>I'm wondering if it is your browser doing the truncation on the fields, as my current "Dial Plan" is already well over 255 chars.  And from the Sipura SPA-3000 manual, we have this comment:<br><br><div class="code"><PRE><span class="codetext">Notes:  - The dial plan length limit for &lt;Dial Plan 1&gt; through &lt;Dial Plan 8&gt; <br>is 511 characters. This is less than that for the &lt;Dial Plan&gt; under &#91;Line 1&#93;, <br>which is 2047 characters. </SPAN></PRE></DIV> </DIV>I finally got it.  <br><br>I think it was me trying to just cut and paste from your entries (dialing toll free numbers via FWD) that was the culprit. There is a carrage return in there, and when I edited the dial plan in a text editor, and removed the carrage return, then reentered the dial plan, all worked as it should.<br><br>Thanks for the great tips!<br>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13752778</guid>
<pubDate>Sun, 26 Jun 2005 21:42:07 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13750225</link>
<description><![CDATA[<A HREF="/useremail/u/523877"><b>yakmandu</b></A> : Although, these are not actually any "tricks", here is a link to some IVR response codes and also some blink codes that could come in handy.<br>&raquo;<A HREF="http://www.voipmechanic.com/networkingandlanissues.htm" >www.voipmechanic.com/networkinga&middot;&middot;&middot;sues.htm</A>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13750225</guid>
<pubDate>Sun, 26 Jun 2005 14:04:02 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13747391</link>
<description><![CDATA[<A HREF="/useremail/u/639783"><b>digiblur</b></A> : My Sipura trick isn't really like yours but here is mine:<br><br>&raquo;<A HREF="/forum/remark,13371620">[Sipura] Make your Sipura Speak! - GetSipura Guide</A><br><br>I haven't had time to work on the project for quite some time but I do plan on working in within a month.  I going to be adding some call logging.  Hopefully be able to show you your outbound, inbound, and missed calls with name, number, and time stamp.   Possibly show the duration of the calls too.  <br><br>I created a separate plugin for the 841 but I haven't totally finished it due to my 841 being out for repair.  <br><SMALL>--<br>FWD#64466(6PM-11PM GMT-5) <BR> &raquo;<A HREF="/forum/remark,13371620">[Sipura] Make your Sipura Speak! - GetSipura Guide</A><BR> Drop me a PM if you'd like a custom Samurize plugin for your device.</SMALL>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13747391</guid>
<pubDate>Sat, 25 Jun 2005 23:52:29 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13747116</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  tlpintpe <A HREF="/useremail/u/687883"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>I am using Firefox on Linux, but I'll give it a go with Konqourer and see if that helps.<br> </DIV>That's odd then, because I do my dial plans with FireFox on Windows (and FF usually behaves pretty much the same cross-platform).  Since I'm not getting Dial Plan truncations, I don't know why you would.  As I said, odd...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13747116</guid>
<pubDate>Sat, 25 Jun 2005 23:05:14 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13747099</link>
<description><![CDATA[<A HREF="/useremail/u/687883"><b>tlpintpe</b></A> : <div class="bquote"><SMALL>said by  DracoFelis <A HREF="/useremail/u/826863"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR><div class="bquote"><SMALL>said by  tlpintpe <A HREF="/useremail/u/687883"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>Is there are 255 character limit per dial plan?</DIV>I'm wondering if it is your browser doing the truncation on the fields, as my current "Dial Plan" is already well over 255 chars.  And from the Sipura SPA-3000 manual, we have this comment:<br><br>Notes:  - The dial plan length limit for  through  is 511 characters. This is less than that for the  under [Line 1], which is 2047 characters. <br> </DIV>I am using Firefox on Linux, but I'll give it a go with Konqourer and see if that helps.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13747099</guid>
<pubDate>Sat, 25 Jun 2005 23:02:40 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13747077</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  tlpintpe <A HREF="/useremail/u/687883"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>Is there are 255 character limit per dial plan?</DIV>I'm wondering if it is your browser doing the truncation on the fields, as my current "Dial Plan" is already well over 255 chars.  And from the Sipura SPA-3000 manual, we have this comment:<br><br><div class="code"><PRE><span class="codetext">Notes:  - The dial plan length limit for &lt;Dial Plan 1&gt; through &lt;Dial Plan 8&gt; <br>is 511 characters. This is less than that for the &lt;Dial Plan&gt; under &#91;Line 1&#93;, <br>which is 2047 characters. </SPAN></PRE></DIV>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13747077</guid>
<pubDate>Sat, 25 Jun 2005 22:58:04 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13747035</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  tlpintpe <A HREF="/useremail/u/687883"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>Thanks!  This is very useful stuff!  I just used the variations for calling FWD with my Sipura 3000 (line 1 is configured with Voxee), and it works very well!</DIV>On an SPA-2000 you can override the proxy via the "dial plan", and thereby allow dialing to open proxies (such as FWD or SIPphone).  But don't forget that the SPA-3000 also lets you override the password, and thereby use multiple providers with different passwords (as well as different proxies).  <br><br>For example:  I just got my SPA-3000 today, and (while the process was more painful than expected) I've already figured out how to have both my DialPad.com ($11.99/month) and my Teliax.com (pay as you go) accounts running on "Line 1" (and both accounts have different SIP passwords).<br><br>Now if I could just find a reliable way to override the "User ID:" field (the "GWx Auth ID:" fields don't seem to behave as expected in this regard), I would be able to include my other SIP accounts that were done under a different userid (in addition to accounts with a different proxy and password)...<br><br>[edit] Just got the solution to my SPA-3000 problem from the Voxilla forums.  Instead of putting in my provider's "proxy_address" (i.e. 66.35.222.58 for DialPad.com) in for the value of "Gateway x:" and my userid for the value of "GWx Auth ID:", you put in "userid@proxy_address" (i.e. myaccount@66.35.222.58 for DialPad) for the "Gateway x:" field, and then also put in your "userid" for the "GWx Auth ID:" field.  You apparently also need a "new enough" version of the firmware for this to work (my SPA-3000 has version "2.0.13(GWg)", which works fine with this "trick").  <br><br>But once you get this "trick" to work, you can use the 4 "gateway" fields to totally override your userid/password/proxy settings of "Line 1" (of an SPA-3000).  This allows you to easily have line 1 setup for 5 different VoIP providers (the default "Line 1" one being by-directional, and the 4 "gateway" ones being "outbound only").  And those 5 providers are on top of any providers (such as FWD) that you can call out to without "authorizing" (i.e. using the previous described "@proxy" trick to the dial-plan).<br><br>BTW:  At the moment I only have 3 providers, so I have some room for expansion.  My "Line 1" default is currently setup for FWD (this is only so that I can receive inbound calls by FWD, otherwise I would have used the previous FWD outbound "trick"), my GW1 is setup for DialPad.com ($11.99/month "unlimited", and also the provider my dial plan selects for LD calls), and GW2 is my "pay as you go" Teliax account (default dialing for toll free numbers, and can be explicitly used by dialing "# 8 call_digits #").]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13747035</guid>
<pubDate>Sat, 25 Jun 2005 22:51:58 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13746937</link>
<description><![CDATA[<A HREF="/useremail/u/687883"><b>tlpintpe</b></A> : I have my 3000 using voxee on line 1.  I used the voxilla.com SPA 3000 wizard to configure the spa, leaving their dialplan unchanged.<br><br>When I try to add the "tricks" to force dialing toll free numbers over the FWD method, the dialplan is truncated and nothing then works.<br><br>Is there are 255 character limit per dial plan? <br><br>Can I move the voxee line to the PSTN Voip line and thus have access to lots more dial plan lines?]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13746937</guid>
<pubDate>Sat, 25 Jun 2005 22:33:19 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13746890</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  digiblur <A HREF="/useremail/u/639783"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>I made my Sipura's do some tricks too ;)</DIV>So please post those "tricks". ;)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13746890</guid>
<pubDate>Sat, 25 Jun 2005 22:23:40 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13746525</link>
<description><![CDATA[<A HREF="/useremail/u/687883"><b>tlpintpe</b></A> : Thanks!  This is very useful stuff!  I just used the variations for calling FWD with my Sipura 3000 (line 1 is configured with Voxee), and it works very well!]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13746525</guid>
<pubDate>Sat, 25 Jun 2005 21:17:57 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13746321</link>
<description><![CDATA[<A HREF="/useremail/u/639783"><b>digiblur</b></A> : I've got a VoicePulse locked 2000 (line2 unlocked).  Unlocked 3000 and 1001 to play with.<br><br>I made my Sipura's do some tricks too ;)]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13746321</guid>
<pubDate>Sat, 25 Jun 2005 20:44:00 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13746028</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : FWIW:  I just got a SPA-3000 (I already had an SPA-2000), and I'm running it though its paces.  However, many of the "tricks" I would like to do (such as forwarding of calls between Sipuras, direct URL dialing, etc), can't easily be tested by me alone (as they require someone with an unlocked Sipura adapter outside my LAN to test against).  <br><br>So if anyone is interested in helping me test when I have some free time (and you have an "unlocked" Sipura adapter), could you please PM me with a number you can be reached at (either a normal USA number, or a FWD number will do), and a convenient time to have me call?  I'll then give you a ring, and we'll see if we can figure out how to get any of the "more confusing" parts of the Sipura features to work...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13746028</guid>
<pubDate>Sat, 25 Jun 2005 19:48:03 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13686393</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : And I found a few useful (and non-obvious) settings in this BBR thread:  &raquo;<A HREF="/forum/remark,13074245">[Equipment] Found Critical Sipura Settings</A>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13686393</guid>
<pubDate>Sat, 18 Jun 2005 02:44:39 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13685573</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : Here's a couple more that I was able to figure out today.  I was experimenting to see if there was some way I could forward incoming calls on my SPA-2000 "Line 1" (FWD), automatically to "Line 2" (my DialPad.com line).  I wanted to do this, so that I didn't need to have a phone hooked up to the Line1 port of the SPA, in order to receive FWD calls (remember, I can already DIAL FWD calls from "Line 2", that "trick" is shown in a previous note in this thread).<br><br><B>Call one line of the SPA from the other line:</B><br>This is not especially useful (more of a "parlor trick"), but it was a good "learning experience" in trying to get "forwarding" to work.  Believe it or not, the SPA-2000 can directly call one line from the other (by using the internal "loop back" IP address)!  Here's how:<br><br>First make sure that each "Line" on the SPA is setup with a unique SIP port, and a unique userid (i.e. make the settings for these two values different for each line).  You will also need to set "Make call without reg: Yes" on the calling line, and "Ans Call Without Reg: Yes" on the line receiving.  And you will also have to make sure that the lines have at least one CODEC in common (it might as well be "G711u", since the "call" is "internal" to the SPA).  Finally, you will have to setup a "dial plan" to call the other line, at "userid@127.0.0.1:sip_port".  <br><br>For example, if the other line is on SIP port 5063, and is userid "testing", than you can call that line (from the other one) by pressing #1 if you have the following as part of your "dial plan":<br><div class="code"><PRE><span class="codetext">&lt;#1:&gt; S0 &lt;:testing@127.0.0.1:5063&gt;</SPAN></PRE></DIV><br><B>And now (drum roll), how to forward all inbound calls to the OTHER line:</B><br>This is VERY USEFUL, because it either lets you have a TWO VoIP accounts that both "ring" the same phone, OR lets you use one account for all incoming, and a 2nd account for all outgoing (by putting the "phone" on the line with the outgoing VoIP service, and then forwarding all incoming calls on that other VoIP line to that one)!<br><br>NOTE:  This theory was tested earlier this evening, by forwarding my SPA-2000's "Line 1" (setup for FWD) to "Line 2" (setup for DialPad.com), and then calling my FWD number from Packet8.  After I finally got all the pieces in place, my "Line 2" was happy to "ring", and when I picked up that phone 2-way talking worked fine!  So this appears to work (at least for me).  But naturally YMMV.<br><br>Here are the needed pieces:<br><br>1) As in the previous "trick", you need unique SIP ports and unique userids for the two lines.  NOTE:  It's quite OK to use whatever "userid" the provider on that line supplied (for logging into their SIP proxy).  You don't need the UserId set to any specific value, just something unique!<br><br>2) Again, the line you are forwarding from will need "Make call without reg: Yes", and the line you want to forward to will need "Ans Call Without Reg: Yes".<br><br>3) If you are behind a router (I am), you will need to forward the SIP port of the line you want to ring (the line you are forwarding to) to the SPA.  This is probably much easier if you program the SPA for a "static LAN IP" (instead of using DHCP).<br><br>4) Your external address will need to either be "static", _OR_ you will need to use a dynamic DNS service (btw:  I'm happy with the free dynamic DNS service from &raquo;<A HREF="http://www.no-ip.com" >www.no-ip.com</A> ).  This is necessary, as you will need to always know the internet address of your SPA-2000 (not the LAN address, the "external address") for forwarding to work.<br><br>5) Turn on "Cfwd All Serv: yes" on the line you are forwarding "from" (i.e. if you want calls to the VoIP on "Line 1" to ring "Line 2", than you set this on "Line 1").<br><br>6) Go over to the "user" tab for the line you are forwarding from, and setup the "Cfwd All Dest:" field as "userid@external_address:sip_port".  For example, if your dynamic DNS entry is "dummy.no-ip.com", your target line's userid is "testing", and the target line's SIP port is 5063, than you would want to "Cfwd All Dest:" to "testing@dummy.no-ip.com:5063".  <br><br>NOTE:  I was NOT successful in getting the loopback address (127.0.0.1) working for call forwarding (even though it worked for calling one line from the other, above).  I had to use the "external address" for the SPA, to get forwarding to work (even between one line and the other on the same Sipura adapter)!<br><br>7) Test the setup.  The easiest way is to get a friend to call the VoIP number you are forwarding from, and see if the forwarded to line "rings".  In my case, I verified the setup by using my Packet8 account (and the Packet8 to FWD gateway) to call my FWD line (line 1 of my SPA-2000), and have the DialPad.com line (Line 2) ring!  I then picked up the phones, and verified that two-way talking was working.  Success!!!<br><br><B>BTW:  So far there has only been one other poster in this thread.  I'm sure the two of us can't be the only ones trying to figure out what our Sipura adapters are capable of!  So please join in and post your "tricks".  I'd hate to have this thread degenerate into just DracoFelis' book of Sipura tricks....</B>]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13685573</guid>
<pubDate>Fri, 17 Jun 2005 23:56:22 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13678001</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : <div class="bquote"><SMALL>said by  rjackson <A HREF="/useremail/u/610601"><IMG SRC="http://i.dslr.net/bb/profile.gif" ALT="See Profile" BORDER=0 WIDTH=16 HEIGHT=11></A>:</SMALL><BR><BR>Dialing **** takes you to the Sipura configuration menu, and then from there you can enter options. I don't know them all but 110# will read you your adapter's current IP address.</DIV>Yep, handy to remember.  <br><br>BTW:  Both the "****" sequence, and the various codes you can follow it with are documented in the Sipura user manual.  You can download the user manual (and upgraded firmware versions) from this web page:  &raquo;<A HREF="http://www.sipura.com/support/index.htm" >www.sipura.com/support/index.htm</A>  ]]></description>
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<pubDate>Fri, 17 Jun 2005 00:21:02 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13658103</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : Since I started this thread, I thought I would post a few "tricks".  All of the "tricks" in this note work by adding things to the "Dial Plan" for the line in your SPA.  If you have an "unlocked" SPA you can use these tricks.  However, if you have a "locked" SPA, the provider may prevent you from editing the "Dial Plan" (and thus prevent you from using these "tricks").<br><br><B>NOTE:  Per the Sipura SPA-xxxx manual, all dial plan patterns need to be separated from each other by a "|" (vertical bar) character.</B>  So when I say to add a pattern to your dial plan, don't forget to separate it from the rest of the dial plan by a "|" character!<br><br>BTW:  In my case, I have tested these tricks on "line 2" of my SPA-2000 (which is currently setup with the $11.99/month DialPad.com outbound VoIP service).  Since these tricks do work on my SPA-2000, and they appear to follow the syntax that is common to the Sipura adapters, they should work with pretty much any SPA model.  However, I have only tested them on my SPA-2000, so YMMV.<br><br><B>Trick 1:  Do your own "911" support.</B><br>This can be useful if/when using a provider that doesn't provide 911 (for example dialpad.com).  This does NOT give you E911 (just normal "speed dialing"), but it is better than nothing.  You first decide which number you would like to "speed dial" when 911 is entered on a phone connected to that SPA-xxxx line.  IMHO the "best way" to get the number, is to call the non-emergency number for your area (usually listed in the front a phone book), and ask them what number to call in an emergency when using a phone that can't directly dial 911.  You can then make an automatic "translation" in the dial plan, so that 911 is turned into your own custom "speed dial".  For example, if your emergency number is 319-555-2222 (no that's not a real emergency number, use the real number for your area), than the following added to your "Dial Plan" will enable custom 911 services: <br><div class="code"><PRE><span class="codetext">&lt;911:13195552222&gt;S0 </SPAN></PRE></DIV>NOTE:  Replace the dummy number after the : (above), with your real emergency "speed dial" number (before adding this code to your SPA's "Dial Plan")!<br><br><B>Trick 2:  Calling Free World Dialup from a line provisioned to another provider.</B><br>The Sipura has a little understood "IP dialing" feature, that can be usefully combined with the FWD SIP gateway (and no you do NOT have to turn on "IP dialing" in the Sipura to use this trick).  When you have a line provisioned to another provider (for example my "line 2" is setup for DialPad.com), you can still make outbound FWD calls (including calling the FWD "Service numbers", and calling "FWD partners") from that line for only 3 extra keystrokes.  Once the following code is added to your line's "Dial Plan", FWD calls are as simple as pressing: #3 fwd_number # (for example, the FWD time number would be reached by calling #3612#).<br><div class="code"><PRE><span class="codetext">&lt;#3:&gt;&#91;x*&#93;.&lt;#:&gt;S0 &lt;:@fwd.pulver.com&gt; </SPAN></PRE></DIV><br><B>Trick 3: Transparently use FWD for USA "Toll Free" numbers, instead of using your provider:</B><br>This is a variation of trick 2 (above).  Some "pay as you go" VoIP providers not only charge for normal LD calls, but also charge for "toll free" calls.  And other providers (such as DialPad.com), don't officially "support" calling toll free number via their service.  In such cases, it could be useful to have toll free numbers automatically go via FWD, instead of dialing through your VoIP provider.  You can easily force 1 (800/888/877/866) xxx-xxxx numbers to use FWD, by putting the following (long string) in your "Dial Plan" BEFORE (to the left of) the pattern that would normally dial LD numbers:<br><div class="code"><PRE><span class="codetext">&lt;:*&gt;1800xxxxxxxS0 &lt;:@fwd.pulver.com&gt; | &lt;:*&gt;1888xxxxxxxS0 &lt;:@fwd.pulver.com&gt; | <br>&lt;:*&gt;1877xxxxxxxS0 &lt;:@fwd.pulver.com&gt; | &lt;:*&gt;1866xxxxxxxS0 &lt;:@fwd.pulver.com&gt;</SPAN></PRE></DIV>]]></description>
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<pubDate>Tue, 14 Jun 2005 19:17:00 EDT</pubDate>
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<title>Re: [Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13657922</link>
<description><![CDATA[<A HREF="/useremail/u/610601"><b>rjackson</b></A> : This works on my SPA-2000, but I would assume other models as well.<br><br>Dialing **** takes you to the Sipura configuration menu, and then from there you can enter options. I don't know them all but 110# will read you your adapter's current IP address. Handy when troubleshooting networking issues.]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13657922</guid>
<pubDate>Tue, 14 Jun 2005 18:55:07 EDT</pubDate>
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<title>[Equipment] Useful Sipura tricks...</title>
<link>http://www.dslreports.com/forum/remark,13657675</link>
<description><![CDATA[<A HREF="/useremail/u/826863"><b>DracoFelis</b></A> : While the Sipura SPA-xxxx VoIP adapters appear to be very powerful (and inexpensive), getting the most out of them is non-obvious.  So I thought I would start a thread where we could post various "tricks" (non-obvious ways to enable useful features), that we have working on our Sipura adapters.  <br><br>If you have a useful Sipura "trick", please post what it does and how to do it here.  If there are any restrictions on your "trick" (for example, if your "trick" only works on a specific model of SPA adapter), please also include that info.  Hopefully, by building up a list of Sipura "tricks", we can all get the most out of our adapters...]]></description>
<guid isPermaLink="true">http://www.dslreports.com/forum/remark,13657675</guid>
<pubDate>Tue, 14 Jun 2005 18:26:24 EDT</pubDate>
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