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