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DracoFelis
Premium
join:2003-06-15
| reply to DracoFelis
Re: [Equipment] Useful Sipura tricks...
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).
Call one line of the SPA from the other line:
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:
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".
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":
<#1:> S0 <:testing@127.0.0.1:5063>
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)!
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.
Here are the needed pieces:
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!
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".
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).
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 »www.no-ip.com ). 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.
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").
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".
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)!
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!!!
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.... | |
tlpintpe
join:2002-09-13
Spicewood, TX
| 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.
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.
Is there are 255 character limit per dial plan?
Can I move the voxee line to the PSTN Voip line and thus have access to lots more dial plan lines? | |
DracoFelis
Premium
join:2003-06-15
1 edit | said by tlpintpe  :Is there are 255 character limit per dial plan? 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:
Notes: - The dial plan length limit for <Dial Plan 1> through <Dial Plan 8>
is 511 characters. This is less than that for the <Dial Plan> under [Line 1],
which is 2047 characters. | |
tlpintpe
join:2002-09-13
Spicewood, TX
| said by DracoFelis :said by tlpintpe :Is there are 255 character limit per dial plan? 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: 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. I am using Firefox on Linux, but I'll give it a go with Konqourer and see if that helps. | |
DracoFelis
Premium
join:2003-06-15
| said by tlpintpe :I am using Firefox on Linux, but I'll give it a go with Konqourer and see if that helps. 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... | |
tlpintpe
join:2002-09-13
Spicewood, TX
| reply to DracoFelis
said by DracoFelis :said by tlpintpe :Is there are 255 character limit per dial plan? 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: Notes: - The dial plan length limit for <Dial Plan 1> through <Dial Plan 8>
is 511 characters. This is less than that for the <Dial Plan> under [Line 1],
which is 2047 characters. I finally got it.
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.
Thanks for the great tips! | |
DracoFelis
Premium
join:2003-06-15
1 edit | said by tlpintpe :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. Yep. A "dial plan" for a Sipura should all be on one line. 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...
While we are on the subject of "dial plans", here are a couple of cute ones. Since SIPphone.com 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):
1 747 xxx xxxx <:@proxy01.sipphone.com>
1 222 xxx xxxx <:@proxy01.sipphone.com> | |
igi
join:2002-04-21
Oceanport, NJ
| reply to DracoFelis
Hi DracoFelis,
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?
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?
Thanks,
I. | |
DracoFelis
Premium
join:2003-06-15
| said by igi :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? 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.
As a result, the predicted behavior of my forwarding "trick" is this:
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).
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.
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".
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! | |
gnexus
join:2005-06-24
1 edit | said by DracoFelis :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! 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". 
I was going to post it, but then I had problems:
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.
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. . .
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. 
Edit: Still no luck. . .One way audio, fleeting two way audio. I did find out something interesting in the SIP logs, however:
SIPphone server response:
Server: Sip EXpress router (0.8.12-tcp_nonb (i386/linux))
That means the SIPphone proxy server is using the same code as the SER being added to DD-WRT! | |
DKreil
@surfer.at
| reply to DracoFelis
This is fantastic and exactly what I'd need!!
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.
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):
- localhost & port
4832525@127.0.0.1:5061
- SIP account at provider (sipgate, Austria):
4832525@sipgate.at
- external IP & port
4832525@voice.kreil.net:5061
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.
What else could I try?
Is it likely that it's a difference between the SPA2100 and the 3000? Is it worth going for the 3000 instead?
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).
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
- require a password
- have a different user ID?
In my case, I need an account in the UK and one in Austria, currently using sipgate (sort of).
I'd be grateful for your thoughts!
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 all the countries my life has touched: US, UK, Germany, Austria) but I doubt you want to hear that
With many thanks again for your help
and best regards,
David. | |
DracoFelis
Premium
join:2003-06-15
| said by DKreil:
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. 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.
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).
said by DKreil:
I have tried to learn from your examples but I cannot get the call forwarding to work on my SPA-2100. 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:
said by DKreil:
- localhost & port
4832525@127.0.0.1:5061
This may work for an INTERNAL call within your LAN. But it will NOT work for "forwarding".
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.
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.
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)!
said by DKreil:
- external IP & port
4832525@voice.kreil.net:5061
The external IP and port seems to work only if all of the following are true:
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.
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!
3) You remembered to set "Ans Call Without Reg: yes" on the line receiving the forwarded call.
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!
said by DKreil:
Is it likely that it's a difference between the SPA2100 and the 3000? Is it worth going for the 3000 instead? "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).
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.
IMHO here are the places the SPA-3000 seems to do better in:
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!
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.
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).
said by DKreil:
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
- require a password
- have a different user ID?
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.
To use the "gateway" slots, enter things like this:
"Gateway x: " gets "userid@sip_proxy" (i.e. your user account at your provider).
"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...
"GWx NAT Mapping Enable: yes" (assuming you are behind NAT, which most of us are).
"GWx Password: " is your SIP password with that provider (obviously).
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):
1[2-9]xx[2-9]xxxxxxS0 <:@GW1> | |
kreil
join:2005-08-20
Austria
| Dear DracoFelis,
Thank you so much for your fast and friendly reply!
Some thoughts and further questions:
said by DracoFelis :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 :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 :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. | |
JakesOnline
join:2005-08-05
Huntington Beach, CA
|  reply to DracoFelis
said by DracoFelis :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? | |
DracoFelis
Premium
join:2003-06-15
| said by JakesOnline :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"). | |
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 | |
aup2
join:2003-06-05
| 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. | |
DracoFelis
Premium
join:2003-06-15
1 edit | said by aup2 :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"! | |
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. | |
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 | |
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