Re: [asterisk-users] G729 Passthrough How To
You ok sir? Are you going to make it? N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
Sent from my Verizon Wireless 4G LTE DROID Eric Wieling wrote: >If you want to do pass-thu you need to make sure Asterisk NEVER EVER needs to >transcode. This means all calls must use only g729, sound files must be in >g729 format and no early audio, inband ringing or anything else which might >cause Asterisk to require a temp transcoding path. > >In my experience it never works right.The most you should expect to be >able to do is reduce the need for transcoding by doing the above steps. > >-Original Message- >From: asterisk-users-boun...@lists.digium.com >[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis >Sent: Wednesday, August 14, 2013 10:20 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [asterisk-users] G729 Passthrough How To > >Anyone? :) > >N. > >On 8/13/13, Nick Khamis wrote: >> Hello Everyone, >> >> We are currently experiencing some higher load on our servers, and >> since signaling comes into our servers on G729, we would like to >> implement G729 pass-through. A few questions arise, do we need to >> convert all the recording to the codec, and what about voicemail? >> >> We are also using A2Billing (hope I am not violating any thread >> rules), and would like to convert all that recording to G729 as well. >> >> Any help is greatly appreciated. >> >> Kind Regards, >> >> Nick from Toronto. >> > >-- >_ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to >Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- >_ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- >New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
Sent from my Verizon Wireless 4G LTE DROID Eric Wieling wrote: >If you want to do pass-thu you need to make sure Asterisk NEVER EVER needs to >transcode. This means all calls must use only g729, sound files must be in >g729 format and no early audio, inband ringing or anything else which might >cause Asterisk to require a temp transcoding path. > >In my experience it never works right.The most you should expect to be >able to do is reduce the need for transcoding by doing the above steps. > >-Original Message- >From: asterisk-users-boun...@lists.digium.com >[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis >Sent: Wednesday, August 14, 2013 10:20 AM >To: Asterisk Users Mailing List - Non-Commercial Discussion >Subject: Re: [asterisk-users] G729 Passthrough How To > >Anyone? :) > >N. > >On 8/13/13, Nick Khamis wrote: >> Hello Everyone, >> >> We are currently experiencing some higher load on our servers, and >> since signaling comes into our servers on G729, we would like to >> implement G729 pass-through. A few questions arise, do we need to >> convert all the recording to the codec, and what about voicemail? >> >> We are also using A2Billing (hope I am not violating any thread >> rules), and would like to convert all that recording to G729 as well. >> >> Any help is greatly appreciated. >> >> Kind Regards, >> >> Nick from Toronto. >> > >-- >_ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to >Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >-- >_ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- >New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
Asterisk does not "generate" prompts. You force G729 in VM by only allowing g729 in voicemail.conf.Try reading the Asterisk book. Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ https://www.google.com/search?q=asterisk+%22file+convert%22 -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Wednesday, August 14, 2013 11:25 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 Passthrough How To Not really no... And how do I make sure Asterisk always generates prompts and VM recordings in G729 from now on. This is also hard to find information.. N. On 8/14/13, Eric Wieling wrote: > I have no idea, though Google might. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick > Khamis > Sent: Wednesday, August 14, 2013 11:16 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] G729 Passthrough How To > > Hey Eric, I do have the codec installed, and I remember hearing about > the CLI command to convert. Is there a recent how-to of blog already > discussing this somewhere? > > N. > > On 8/14/13, Nick Khamis wrote: >> I wanted to mention that I do not mind posting the converted files on >> this list for future individuals, given that I am not doing anything >> illegal... >> >> N. >> > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
Not really no... And how do I make sure Asterisk always generates prompts and VM recordings in G729 from now on. This is also hard to find information.. N. On 8/14/13, Eric Wieling wrote: > I have no idea, though Google might. > > -Original Message- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis > Sent: Wednesday, August 14, 2013 11:16 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] G729 Passthrough How To > > Hey Eric, I do have the codec installed, and I remember hearing about the > CLI command to convert. Is there a recent how-to of blog already discussing > this somewhere? > > N. > > On 8/14/13, Nick Khamis wrote: >> I wanted to mention that I do not mind posting the converted files on >> this list for future individuals, given that I am not doing anything >> illegal... >> >> N. >> > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
I wanted to mention that I do not mind posting the converted files on this list for future individuals, given that I am not doing anything illegal... N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
I have no idea, though Google might. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Wednesday, August 14, 2013 11:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 Passthrough How To Hey Eric, I do have the codec installed, and I remember hearing about the CLI command to convert. Is there a recent how-to of blog already discussing this somewhere? N. On 8/14/13, Nick Khamis wrote: > I wanted to mention that I do not mind posting the converted files on > this list for future individuals, given that I am not doing anything > illegal... > > N. > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
Hey Eric, I do have the codec installed, and I remember hearing about the CLI command to convert. Is there a recent how-to of blog already discussing this somewhere? N. On 8/14/13, Nick Khamis wrote: > I wanted to mention that I do not mind posting the converted files on > this list for future individuals, given that I am not doing anything > illegal... > > N. > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
"file convert" in the Asterisk CLI, IF you have the g729 codec installed. You need to convert every single file you may play to a caller You can't force Asterisk to never attempt transcoding, the most you can do is force all sip.conf entries to use g729. It will still transcode to play ringback to the caller. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Wednesday, August 14, 2013 10:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 Passthrough How To Hey!!! Eric thank you so much for your response. Could you guys please direct us in achieving as much as possible. For example: * What linux command can we use to convert all recording to G729 * Which files do we need to convert and there locations * For *testing* how do we make sure Asterisk NEVER EVER transcodes. Do we still need the G729 codec installed on the asterisk machine if we manage to implement pass-through that would suffice our needs. Kind Regards, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
Hello Ashgar, Thank you so much for your response. As removing A2B is not an option we would first like to begin by converting all audio files (Asterisk, VM, A2B prompts etc...) to G729 to minimize unneeded trascoding. Linux commands and the list of recording would be a great help. Sorry, not new to VoIP but new to Asterisk :). N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
As my understanding Asterisk always pass-thu g729 if both ends have this codec. But if you answer the call or play some audio before dialing to end point then asterisk stay between both legs. In case of VM. you should install g729 if your prompts are in g729 format. As a2billing play voice prompts you cannot pass-thu transparently. I think the load on you server is not for transcoding but PHP scripts. I was in this situation and reduce the upto 80% by removing A2B. On Wed, Aug 14, 2013 at 4:44 PM, Nick Khamis wrote: > Hey!!! Eric thank you so much for your response. Could you guys please > direct us in achieving as much as possible. For example: > * What linux command can we use to convert all recording to G729 > * Which files do we need to convert and there locations > * For *testing* how do we make sure Asterisk NEVER EVER transcodes. > > Do we still need the G729 codec installed on the asterisk machine if > we manage to implement pass-through that would suffice our needs. > > Kind Regards, > > Nick. > > -- > _ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: >http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
I forgot to mention that all our equipment (phones etc..) are using G729, and this is for internal use over the net. The problem, concurrent calls, and bad bandwidth at some locations... N. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
Hey!!! Eric thank you so much for your response. Could you guys please direct us in achieving as much as possible. For example: * What linux command can we use to convert all recording to G729 * Which files do we need to convert and there locations * For *testing* how do we make sure Asterisk NEVER EVER transcodes. Do we still need the G729 codec installed on the asterisk machine if we manage to implement pass-through that would suffice our needs. Kind Regards, Nick. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
If you want to do pass-thu you need to make sure Asterisk NEVER EVER needs to transcode. This means all calls must use only g729, sound files must be in g729 format and no early audio, inband ringing or anything else which might cause Asterisk to require a temp transcoding path. In my experience it never works right.The most you should expect to be able to do is reduce the need for transcoding by doing the above steps. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nick Khamis Sent: Wednesday, August 14, 2013 10:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] G729 Passthrough How To Anyone? :) N. On 8/13/13, Nick Khamis wrote: > Hello Everyone, > > We are currently experiencing some higher load on our servers, and > since signaling comes into our servers on G729, we would like to > implement G729 pass-through. A few questions arise, do we need to > convert all the recording to the codec, and what about voicemail? > > We are also using A2Billing (hope I am not violating any thread > rules), and would like to convert all that recording to G729 as well. > > Any help is greatly appreciated. > > Kind Regards, > > Nick from Toronto. > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 Passthrough How To
Anyone? :) N. On 8/13/13, Nick Khamis wrote: > Hello Everyone, > > We are currently experiencing some higher load on our servers, and > since signaling comes into our servers on G729, we would like to > implement G729 pass-through. A few questions arise, do we need to > convert all the recording to the codec, and what about voicemail? > > We are also using A2Billing (hope I am not violating any thread > rules), and would like to convert all that recording to G729 as well. > > Any help is greatly appreciated. > > Kind Regards, > > Nick from Toronto. > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g729 passthrough?
Better is really not to use G723... ;) Can't you use others alternatives? Regards, -Mensagem original- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Em nome de Asterisk guy Enviada: segunda-feira, 25 de Abril de 2005 3:17 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] g729 passthrough? i am trying to get G723 passthrough get the same error. how to configure passthrough for g723/g729 ? On 4/24/05, Brian Capouch <[EMAIL PROTECTED]> wrote: > jltaylor wrote: > > > ;;; > > > > Brian, > > > > Add to the [general] section in sip.conf the following: > > > > disallow=all > > allow=g729 > > allow=ulaw > > allow=alaw > > > > > > For some reason Asterisk will not pass audio through itself without trying > > to transcode unless you have this in your config. > > Don't ask me why it will not work with allow=g729 under the individual peer. > > This has to go in the [general] section. > > > > Still no joy. Added the allow=g729 to general, too, and I still get the > same errors. > > Thanks anyways. > > B. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 passthrough?
I got some advice from Josh Colp that has helped with some of my problem: it may have a little logic flaw in the way transcoding is supposed to be done, from the way your message is I would say you are getting hit by this. (Upgrading to latest CVS head will fix it)… but one solution is to be the following in asterisk.conf in /etc/asterisk [options] transcode_via_sln = no That’ll cause it to bridge the two and not try to transcode through signed linear. Enjoy! Well that worked, after a fashion. Now AS LONG AS I ONLY USE G.729 ONLY things are fine. But the 841 does all kinds of codecs, and so I'd like it to use g.726 to talk to a provider that doesn't speak g.729. So I set the sip.conf for the phone to "disallow=all; allow=g726,g729" and then try to connect to the g726-only server: Apr 25 00:58:42 NOTICE[5839]: channel.c:1833 set_format: Unable to find a path from g729 to g726 After playing with this for as long as I could stand to, it appears that IFF I am talking to a g729-only endpoint and I set the SIP phone to use g729 only, things are fine. Once I deviate from that (unfortunately restrictive) setup, I can't seem to do anything. In other words, if g729 is in the mix it seems to always choose it despite my preferences, and things get hosed. I'd love to hear from someone who has conquered this. Thanks. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 passthrough?
i am trying to get G723 passthrough get the same error. how to configure passthrough for g723/g729 ? On 4/24/05, Brian Capouch <[EMAIL PROTECTED]> wrote: > jltaylor wrote: > > > ;;; > > > > Brian, > > > > Add to the [general] section in sip.conf the following: > > > > disallow=all > > allow=g729 > > allow=ulaw > > allow=alaw > > > > > > For some reason Asterisk will not pass audio through itself without trying > > to transcode unless you have this in your config. > > Don't ask me why it will not work with allow=g729 under the individual peer. > > This has to go in the [general] section. > > > > Still no joy. Added the allow=g729 to general, too, and I still get the > same errors. > > Thanks anyways. > > B. > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] g729 passthrough?
jltaylor wrote: ;;; Brian, Add to the [general] section in sip.conf the following: disallow=all allow=g729 allow=ulaw allow=alaw For some reason Asterisk will not pass audio through itself without trying to transcode unless you have this in your config. Don't ask me why it will not work with allow=g729 under the individual peer. This has to go in the [general] section. Still no joy. Added the allow=g729 to general, too, and I still get the same errors. Thanks anyways. B. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] g729 passthrough?
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Brian Capouch Sent: Sunday, April 24, 2005 3:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] g729 passthrough? I'm sitting here with my dunce cap on. My weak excuse is that I haven't ever played with g729 before. I have a Sipura 841. I have the phone config set to use g729. Its appropriate sip.conf entry, and the IAX stanza for my ITSP all set to disallow=all, allow=g729. But as soon as I dial, I get a complaint from the server: -- Call accepted by 66.225.202.72 (format g729) -- Format for call is g729 Apr 24 15:38:38 NOTICE[5586]: channel.c:1833 set_format: Unable to find a path from g729 to slin . . . . I get ringback from Nufone, but as soon as the call answers I get an error: Apr 24 15:43:42 NOTICE[5596]: channel.c:1833 set_format: Unable to find a path from g729 to slin . . . What am I doing wrong to cause it to want to transcode? I assume that's where the complaint is coming from. I thought Asterisk could pass through without transcoding as long as the endpoints are all g729. Thanks. B. ;;; Brian, Add to the [general] section in sip.conf the following: disallow=all allow=g729 allow=ulaw allow=alaw For some reason Asterisk will not pass audio through itself without trying to transcode unless you have this in your config. Don't ask me why it will not work with allow=g729 under the individual peer. This has to go in the [general] section. James Taylor MetroTel ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
