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[General] PAP2 v2 and syslog »
« [VoiceStick] SIP Ports  
page: 1 · 2
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jeanclaudenow

@sympatico.ca

reply to kreil
telephone adapter

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.
Please advise me if you have this product available otherwise please refer me to a place which can provide me such an adapter.
Thanks.

rizzo2dial
Premium
join:2004-08-05

reply to rizzo2dial
Re: [Equipment] Useful Sipura tricks...

said by rizzo2dial See Profile :

Here's how to configure LINE1 of the SPA-1001 to dial "911" on LINE2:

1) In the "Phone" tab, set the "Line 2 Select Code" to be the default value of "#"

2) In the "User 1" TAB, configure a "Speed Dial [2-9]" entry of your choice as "#911" (for dialing "911" on LINE2).

3) Assuming you configured entry "Speed Dial 2" in the previous step, insert into the beginning of the LINE1 Dialing plan the following:
<911:2>S0|
If you configured a different "Speed Dial [2-9]" entry, replace the ":2" above with ":x"
(where x is the configured "Speed Dial [2-9]" entry #).

When dialing 911, you'll hear a LINE2 dialtone for a split second followed by the true 911 call on LINE2.

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).

With this solution, you retain full LINE2 functionality should you want/need to manually dial out on that line.

Rizzo
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.

Here's your choice of modified LINE1 dialing plans prefixes 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:

Option #1: Directly dialing "2#" results in a FAST BUSY SIGNAL
<911:2>S0|<2#:>S0|

Option #2: Directly dialing "2#" gives the LINE2 DIAL-TONE
<911:2>S0|<2#:#>S0|

Option #3: Directly dialing "2#" dials a phone number of your choice
<911:2>S0|<2#:xxxxxxxxxxx>S0|
Replace 'xxxxxxxxxxx' w/ the phone number of your choice.

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).

Rizzo


DracoFelis
Premium
join:2003-06-15

reply to rizzo2dial
said by rizzo2dial See Profile :

Here's how to configure LINE1 of the SPA-1001 to dial "911" on LINE2:

1) In the "Phone" tab, set the "Line 2 Select Code" to be the default value of "#"

2) In the "User 1" TAB, configure a "Speed Dial [2-9]" entry of your choice as "#911" (for dialing "911" on LINE2).

3) Assuming you configured entry "Speed Dial 2" in the previous step, insert into the beginning of the LINE1 Dialing plan the following:
<911:2>S0|
If you configured a different "Speed Dial [2-9]" entry, replace the ":2" above with ":x"
(where x is the configured "Speed Dial [2-9]" entry #).
Hey, cool!

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.

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!

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:
<12:2>S0

Hmmm... Let's see...
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...

rizzo2dial
Premium
join:2004-08-05


2 edits
reply to rizzo2dial
said by rizzo2dial See Profile :

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).

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:
<911:2#>S0
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.
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.

Here's how to configure LINE1 of the SPA-1001 to dial "911" on LINE2:

1) In the "Phone" tab, set the "Line 2 Select Code" to be the default value of "#"

2) In the "User 1" TAB, configure a "Speed Dial [2-9]" entry of your choice as "#911" (for dialing "911" on LINE2).

3) Assuming you configured entry "Speed Dial 2" in the previous step, insert into the beginning of the LINE1 Dialing plan the following:
<911:2>S0|
If you configured a different "Speed Dial [2-9]" entry, replace the ":2" above with ":x"
(where x is the configured "Speed Dial [2-9]" entry #).

When dialing 911, you'll hear a LINE2 dialtone for a split second followed by the true 911 call on LINE2.

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).

With this solution, you retain full LINE2 functionality should you want/need to manually dial out on that line.

Rizzo

rizzo2dial
Premium
join:2004-08-05

reply to DracoFelis
said by DracoFelis See Profile :

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.
I agree that an SPA-3000 would work perfectly.

said by DracoFelis See Profile :

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!
For the situation I'm working w/ at the moment, an SPA-3000 is not available.

said by DracoFelis See Profile :

BTW:
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"!
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).

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:
<911:2#>S0
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.

said by DracoFelis See Profile :

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:
<911:15555551234>S0
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.

Rizzo


DracoFelis
Premium
join:2003-06-15


1 edit
reply to rizzo2dial
said by rizzo2dial See Profile :

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).

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.
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.

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!

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!

BTW:
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"!

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:
<911:15555551234>S0

Edit:
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).

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)!

rizzo2dial
Premium
join:2004-08-05


2 edits
reply to p2pvoice
said by p2pvoice See Profile :

DracoFelis:

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.

Here is the "trick: I need to implement:

My business number is an IPKall number, which "points" to my "business" extension on my Asterisk server.

My home number is "ported" to an ITSP and it "points" my "home" extension.

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.

Will this "trick" (inspired by your SIPPhone post) work?

911

I don't know where to put UN/PW?

BTW, my Asterisk server is registered to My911ITSP.com

Thanks
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:

Equipment: PAP2 (running SPA1001 3.1.8 firmware)
LINE1: Outgoing VoIP line (from a provider w/o any 911/E911 support)
LINE2: Incoming VoIP line (from a provider w/ unlimited incoming, metered outgoing, but w/ true E911 support)

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 "555" (so that you can configure the solution and see how it works):

1) In the SPA1001's PHONE tab, configure the "Line 2 Select Code" to be "555" instead of "#"

2) In the LINE1 tab, add the following to the beginning of the Dialing Plan: "555S0|" (this will ensure that 555 gets detected immediately on the LINE1 side)

3) On the LINE2 tab, set the entire dialing plan to be a HOTLINE:
(S1 <:18005551212>)
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").

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.

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).

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.

Rizzo


DracoFelis
Premium
join:2003-06-15

reply to p2pvoice
said by p2pvoice See Profile :

Will this "trick" (inspired by your SIPPhone post) work?

911 S0 <:@My911ITSP.com>

I don't know where to put UN/PW?
In order to use an alternate (not your "registered") VoIP account for 911 (only), you would need one of the following options:

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.

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.

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:
911 S0 <:@GW1>

Warning:
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...

p2pvoice

join:2003-10-28
Dover, NH

 reply to p2pvoice
In my previuos post, "cut and paste" didn't work properly.
Here is the corrected section:

Will this "trick" (inspired by your SIPPhone post) work?

911 S0 <:@My911ITSP.com>

I don't know where to put UN/PW?

Thanks

p2pvoice

join:2003-10-28
Dover, NH

reply to DracoFelis
DracoFelis:

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.

Here is the "trick: I need to implement:

My business number is an IPKall number, which "points" to my "business" extension on my Asterisk server.

My home number is "ported" to an ITSP and it "points" my "home" extension.

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.

Will this "trick" (inspired by your SIPPhone post) work?

911

I don't know where to put UN/PW?

BTW, my Asterisk server is registered to My911ITSP.com

Thanks


t_rajan_m

join:2005-12-23
Rochester, NY
 reply to DracoFelis
Thanks for the forwarding post. This is very useful. I just wanted to know whether this forwarding works in SPA 3000?

Lefteris9

join:2005-12-22
Greece

reply to Bollie Bolst
Hi,

As we saw here thanks to DracoFelis, we can forward incoming calls from one line to the other, but how about forwarding outgoing calls?!?

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).
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.

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)?

So far I was not able to do this, so I'm seeking for help....

Bollie Bolst

join:2005-12-21
reply to DracoFelis
@dracefelis

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..

I use the Sipura 2002


amdrkenny

@ntl.com

reply to DracoFelis
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.


DracoFelis
Premium
join:2003-06-15


1 edit
reply to aup2
said by aup2 See Profile :

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 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".

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...

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.

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"!

aup2

join:2003-06-05

reply to venk25
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?
Thanks.

venk25

join:2004-05-12
Nashua, NH
·VoicePulse

reply to JakesOnline
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.

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


DracoFelis
Premium
join:2003-06-15

reply to JakesOnline
said by JakesOnline See Profile :

I tried forwarding to line1id@127.0.0.1:5060, line1@myDynDNSname.com:5060 and line1id@myinternetip:5060.

i'm pretty sure i followed every step.

ports are forwarded properly on the nat as well.

Any ideas?
Even though I posted that "trick", it appears to be a YMMV thing (works for some, and not for others).

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".

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).

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").

JakesOnline

join:2005-08-05
Huntington Beach, CA

 reply to DracoFelis
said by DracoFelis See Profile :

And now (drum roll), how to forward all inbound calls to the OTHER line:
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)!
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.

I tried forwarding to line1id@127.0.0.1:5060, line1@myDynDNSname.com:5060 and line1id@myinternetip:5060.

i'm pretty sure i followed every step.

ports are forwarded properly on the nat as well.

Any ideas?

kreil

join:2005-08-20
Austria

reply to DracoFelis
Dear DracoFelis,

Thank you so much for your fast and friendly reply!

Some thoughts and further questions:

said by DracoFelis See Profile:

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).
Ah, thanks!!

said by DracoFelis See Profile:

said by DKreil:

- SIP account at provider (sipgate, Austria):
4832525@sipgate.at
This one should work if BOTH of the following are true:

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.
Yes, this is my "external" SIP number, and anyone should be able to call it.

said by DracoFelis See Profile:

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)!
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
They do have voicemail though, so I wonder whether they do that via a SIP reinvite message or with a different internal method...

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.)

Else, do you perhaps know anyone using a provider offering UK numbers and supporting SIP reinvite?

What I am basically looking for is placing and receiving calls on one phone to and from two different SIP provider accounts.

From what you say, I probably need to swap my SPA2100 for an SPA3000. While at the moment my setup is

Cablemodem
|
SPA2100 (w/NAT) -- phone
|
USR8054 router (w/NAT, again) and WiFi access point
|
PCs

this means

Cablemodem
|
USR8054 router (w/NAT) and WiFi access point
| |
| SPA3000
|
PCs

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?

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?

Many thanks again for your kind help!

With best regards,

David.
Forums » VOIP etc » Voice Over IP - VOIP » VOIP Tech Chat[General] PAP2 v2 and syslog »
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