Hi, I'm experiencing some problems with polycom phones, asterisk and g729 codec. As I understand, between polycom and polycom i can use g729 without license at all as long as I'm using codec_g729.so module (i'm using the Open Source Implementation ( http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ ) because it's pure pass-thru and there's no transcoding). My sip.conf has the following options: [general] disallow=all allow=g729 allow=ulaw [voipuser] type=friend username=user host=dynamic callerid=user <202> mailbox=202@default secret=gbvVf423 canreinvite=no insecure=yes disallow=all allow=g729 so i force the voipuser to use g729 as main codec. The problem comes when i try to connect to other polycom phone with the same config as voipuser. The CLI shows the following: Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible codecs! show modules doesnt show codec_g729.so but if i try to load it i get this: Unable to load module codec_g729.so Sep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module 'codec_g729.so' already exists Anyone had this issue? If you need more information, feel fre to ask for it :) Thanks a lot! Santiago
Make sure the codec used by the Polycom will be only g729 via the phone's web interface, as far as I remember Polycom will try always to use ulaw or alaw first unless it is configured to use only or as first choice the g729 codec. Alyed ---------------------------------------- Return-Path: <asterisk-users-bounces@lists.digium.com> Tue Sep 19 14:47:54 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Tue, 19 Sep 2006 14:47:54 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) by lists.digium.com (Postfix) with ESMTP id AB0F03C1F4; Hi, I'm experiencing some problems with polycom phones, asterisk and g729 codec. As I understand, between polycom and polycom i can use g729 without license at all as long as I'm using codec_g729.so module (i'm using the Open Source Implementation ( http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ ) because it's pure pass-thru and there's no transcoding). My sip.conf has the following options: [general] disallow=all allow=g729 allow=ulaw [voipuser] type=friend username=user host=dynamic callerid=user <202> mailbox=202@default secret=gbvVf423 canreinvite=no insecure=yes disallow=all allow=g729 so i force the voipuser to use g729 as main codec. The problem comes when i try to connect to other polycom phone with the same config as voipuser. The CLI shows the following: Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible codecs! show modules doesnt show codec_g729.so but if i try to load it i get this: Unable to load module codec_g729.so Sep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module 'codec_g729.so' already exists Anyone had this issue? If you need more information, feel fre to ask for it :) Thanks a lot! Santiago _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060919/87f1cdbd/attachment.htm
Not an expert at reading Polycom config files, but guess g729 and ulaw are both preference 1 isn't it? hey... you have in your sip.conf configuration "canreinvite=no"... think this may be a problem: since Asterisk will always stay in the path of the RTPs, I think it might need to have the proper transcoder, as it does not, then the error arises... at least that's what I think :) set "canreinvite=yes" (or just comment it since that's the default) on both parties and try again. Let me know if it works. Alyed ---------------------------------------- Return-Path: <asterisk-users-bounces@lists.digium.com> Wed Sep 20 12:38:41 2006 Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; Wed, 20 Sep 2006 12:38:41 -0700 Received: from digium-69-16-138-164.phx1.puregig.net (localhost [127.0.0.1]) Still having the same problem. i modified the sip.cfg in order to make g729 the first choice: voice.codecPref.G711A="3" voice.codecPref.G729AB="1" voice.codecPref.IP_4000.G711Mu="1" voice.codecPref.IP_4000.G711A="2" voice.codecPref.IP_4000.G729AB=""/> Cheers, Santiago On 9/19/06, Alyed Tzompa wrote:> Make sure the codec used by the Polycom will be only g729 via the phone's > web interface, as far as I remember Polycom will try always to use ulaw or > alaw first unless it is configured to use only or as first choice the g729 > codec. > > Alyed > > ________________________________ > Return-Path: Tue > Sep 19 14:47:54 2006 > Received: from digium-69-16-138-164.phx1.puregig.net > [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; > Tue, 19 Sep 2006 14:47:54 -0700 > Received: from digium-69-16-138-164.phx1.puregig.net > (localhost [127.0.0.1]) > by lists.digium.com (Postfix) with ESMTP id AB0F03C1F4; > > Hi, I'm experiencing some problems with polycom phones, asterisk and g729 > codec. > > As I understand, between polycom and polycom i can use g729 without > license at all as long as I'm using codec_g729.so module (i'm using > the Open Source Implementation ( > http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ > ) > because it's pure pass-thru and there's no transcoding). > > My sip.conf has the following options: > > [general] > disallow=all > allow=g729 > allow=ulaw > > > [voipuser] > type=friend > username=user > host=dynamic > callerid=user <202> > mailbox=202@default > secret=gbvVf423 > canreinvite=no > insecure=yes > disallow=all > allow=g729 > > > so i force the voipuser to use g729 as main codec. The problem comes > when i try to connect to other polycom phone with the same config as > voipuser. The CLI shows the following: > > Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible > codecs! > > show modules doesnt show codec_g729.so but if i try to load it i get this: > > Unable to load module codec_g729.so > Sep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module > 'codec_g729.so' already exists > > > Anyone had this issue? > > If you need more information, feel fre to ask for it :) > > > Thanks a lot! > > Santiago > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060920/8e6e2319/attachment.htm
Sorry but I've ran out of ideas... Anyone else out there with a successful Polycom g729 pass through-only experience? Alyed ---------------------------------------- Return-Path: <delcas@gmail.com> Thu Sep 21 11:27:21 2006 Received: from nz-out-0102.google.com [64.233.162.206] by maila11.webcontrolcenter.com with SMTP; Thu, 21 Sep 2006 11:27:21 -0700 Received: by nz-out-0102.google.com with SMTP id z6so390195nzd didn't work :( Regards, Santiago On 9/20/06, Alyed Tzompa wrote:> Not an expert at reading Polycom config files, but guess g729 and ulaw are > both preference 1 isn't it? > > hey... you have in your sip.conf configuration "canreinvite=no"... think > this may be a problem: since Asterisk will always stay in the path of the > RTPs, I think it might need to have the proper transcoder, as it does not, > then the error arises... at least that's what I think :) > > set "canreinvite=yes" (or just comment it since that's the default) on both > parties and try again. > > Let me know if it works. > > Alyed > > ________________________________ > Return-Path: Wed > Sep 20 12:38:41 2006 > Received: from digium-69-16-138-164.phx1.puregig.net > [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; > Wed, 20 Sep 2006 12:38:41 -0700 > Received: from digium-69-16-138-164.phx1.puregig.net > (localhost [127.0.0.1]) > > Still having the same problem. i modified the sip.cfg in order to make > g729 the first choice: > > > > voice.codecPref.G711A="3" voice.codecPref.G729AB="1" > voice.codecPref.IP_4000.G711Mu="1" voice.codecPref.IP_4000.G711A="2" > voice.codecPref.IP_4000.G729AB=""/> > > > Cheers, > Santiago > > On 9/19/06, Alyed Tzompa wrote: > > Make sure the codec used by the Polycom will be only g729 via the phone's > > web interface, as far as I remember Polycom will try always to use ulaw or > > alaw first unless it is configured to use only or as first choice the g729 > > codec. > > > > Alyed > > > > ________________________________ > > Return-Path: Tue > > > Sep 19 14:47:54 2006 > > Received: from digium-69-16-138-164.phx1.puregig.net > > [69.16.138.164] by maila11.webcontrolcenter.com with SMTP; > > Tue, 19 Sep 2006 14:47:54 -0700 > > Received: from digium-69-16-138-164.phx1.puregig.net > > (localhost [127.0.0.1]) > > by lists.digium.com (Postfix) with ESMTP id AB0F03C1F4; > > > > Hi, I'm experiencing some problems with polycom phones, asterisk and g729 > > codec. > > > > As I understand, between polycom and polycom i can use g729 without > > license at all as long as I'm using codec_g729.so module (i'm using > > the Open Source Implementation ( > > > http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/ > > ) > > because it's pure pass-thru and there's no transcoding). > > > > My sip.conf has the following options: > > > > [general] > > disallow=all > > allow=g729 > > allow=ulaw > > > > > > [voipuser] > > type=friend > > username=user > > host=dynamic > > callerid=user <202> > > mailbox=202@default > > secret=gbvVf423 > > canreinvite=no > > insecure=yes > > disallow=all > > allow=g729 > > > > > > so i force the voipuser to use g729 as main codec. The problem comes > > when i try to connect to other polycom phone with the same config as > > voipuser. The CLI shows the following: > > > > Sep 19 18:37:38 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible > > codecs! > > > > show modules doesnt show codec_g729.so but if i try to load it i get this: > > > > Unable to load module codec_g729.so > > Sep 19 18:39:16 WARNING[10688]: loader.c:305 __load_resource: Module > > 'codec_g729.so' already exists > > > > > > Anyone had this issue? > > > > If you need more information, feel fre to ask for it :) > > > > > > Thanks a lot! > > > > Santiago > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > > --Bandwidth and Colocation provided by Easynews.com -- > > > > asterisk-users mailing list > > To UNSUBSCRIBE or update options visit: > > > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060921/951e8ce8/attachment.htm