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DracoFelis
Premium
join:2003-06-15
| reply to DracoFelis
How to do an SPA-3000 setup like mine...
Line 1 | 
Line 1 | 
Line 1 | 
Provisioning | 
Regional | 
Regional | 
SIP | 
System | 
User 1 |
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).
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.
FWIW: Here is the rough order I would recommend for setting things up in:
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).
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).
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.
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.
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).
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.
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".
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.
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).
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).
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.
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!
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".
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.
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.
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".
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".
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).
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. | |
igi
join:2002-04-21
Oceanport, NJ
| Hi DracoFelis,
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:
1. If I call my FWD using a softphone, Line 2 rings but I can't hear. Hearing from the softphone is OK.
2. If I call from a regular phone using IPKALL, I always get the recorded IPKALL message, never rings.
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?
Sorry if some of these questions are obvious.... and thanks a lot for your postings.
I. | |
DracoFelis
Premium
join:2003-06-15
| said by igi  :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. Doing it that way, you aren't really forwarding the call to your other line per se.
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.
However:
said by igi : This works fine, and my question now is about quality and latency. 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.
said by igi :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? 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.
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).
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). | |
igi
join:2002-04-21
Oceanport, NJ
| DracoFelis,
Thanks, now I see.
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....
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.
Thanks,
I. | |
kreil
join:2005-08-20
Austria
| Hi igi,
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.
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?
Inspired by your proxy example, I tried forwarding to proxy.at.sipgate.net --> nothing.
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...
With best regards,
David. | |
DracoFelis
Premium
join:2003-06-15
| reply to igi
said by kreil :From what you say, I probably need to swap my SPA2100 for an SPA3000. said by igi :I'm starting to think that I should go ahead and replace the 2000 for a 3000, that's what you did, right? 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.
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!
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!
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").
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:
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:
»www.sandman.com/lineshar.html
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.
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). | |
igi
join:2002-04-21
Oceanport, NJ
| reply to kreil
David,
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.
And when you try the proxy option, did you use userid@proxy?
That was key in having it working.
I. | |
ejrobinson
Premium
join:2003-05-16
Miami Beach, FL
 ·magicjack.com
| reply to DracoFelis
Your explanations are really quite interesting, though somewhat over my head at this moment. I have a couple of questions.
1. Why use stun? Some people here say stun is to be avoided if possible. You seem to think the opposite.
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?
-er | |
kreil
join:2005-08-20
Austria
| reply to DracoFelis
Dear DracoFelis,
Thanks for your fast and helpful reply! 
I almost forgot: With the SPA3000 I cannot really have two incoming VoIP accounts, right, not even with forwarding tricks because there is only Line 1, correct? 
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... 
Have you ever considered / heard of people using the SPA-1001 with the *SE firmware? It sort of sounds like what I need.
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.
Alternatively, another setup using two of the devices you found to combine two phone lines might be (dots just to make stuff align)
Cablemodem
. |
SPA2100 --- Line 1 ----\_____ phone
. | \ ----- Line 2 ----/ .. /
. | ...................... /
USR8054 (or other router) /
. | | .................. /
. | PCs ............... /
. | .................. /
SPA3000 --- Line 1 -- /
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.
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?
This worry was one of the main reasons I went for the SPA2100, which I could put in front of the router.
Looking forward to hearing from you,
David. | |
kreil
join:2005-08-20
Austria
| reply to igi
Dear igi,
said by igi : 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. 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!)
said by igi :My router has a whole bunch of ports forwarded to the SPA, so check that part. And when you try the proxy option, did you use userid@proxy? That was key in having it working. 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.
With many thanks,
David. | |
DracoFelis
Premium
join:2003-06-15
| reply to ejrobinson
said by ejrobinson :1. Why use stun? Some people here say stun is to be avoided if possible. You seem to think the opposite. 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):
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)!
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.
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.
said by ejrobinson :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? I've not tried it, however it should be possible.
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".
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). | |
kreil
join:2005-08-20
Austria
| Dear DracoFelis,
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?
Best wishes,
David. | |
DracoFelis
Premium
join:2003-06-15
| reply to kreil
said by kreil :Regarding using the "line" of the SPA3000 - isn't this a POTS FXO port to go to the phone company? Yes.
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.
Here are some other uses clever people over on the Voxilla.com forums have also made of the "Line" port:
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").
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.
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 *! | |
kreil
join:2005-08-20
Austria
| Amazing! 
Just imagine they would actually configure their devices to make this easy, there would probably be a TelCo revolt! 
Many thanks again for the heads up!
With best regards,
David. | |
DracoFelis
Premium
join:2003-06-15
| reply to kreil
said by kreil : 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. The SPA-3000 has two RJ11 telco jacks labeled "Phone" and "Line" (in addition to the RJ45 jack for the ethernet).
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.
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? | |
kreil
join:2005-08-20
Austria
| 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?)
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?
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).
So I'm back to a situation where I might just as well get an SPA1001 for the little I need.
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?
With best regardd,
David. | |
DracoFelis
Premium
join:2003-06-15
| said by kreil :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? In my limited experience, Sipura adapters seem to perform about as well as any ATA would when behind a non-QOS router.
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. | |
kreil
join:2005-08-20
Austria | Thanks, DracoFelis!
That's very useful to know. Off to ponder what to do an on the hardware side...
Best regards,
David. | |
kreil
join:2005-08-20
Austria
| reply to DracoFelis
Aaaargh!
sipgate.co.uk support:
"We do not currently support call forwarding and reinvite.
There are no methods to forward your number to another Sipgate number."
Thoroughly disentchanted, I am. 
That only leaves either having two phones sitting side by side, a hardware gadget to combine two phonelines, or an SPA-1001.
Or the Fritz!Box. I'm increasingly wondering...
»www.avm.de/en/index.php3
Best wishes,
David. | |
will792
join:2003-11-18
Stamford, CT
| reply to DracoFelis
Configuration of SPA-3000 for gateways
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.
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 .
The result is the same; fast busy signal after dialing a number, i.e. 102#613 (for gateway 2, Echo test for FWD).
Any ideas what I am missing? Maybe it is not possible to configure gateway accounts with "Gateway 1..4" fields only?
My firware version is 2.0.10(GWc) and it cannot be change (VP locks it).
Thank you in advance for any information. | |
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