<font face="arial" size="2">Not an expert at reading Polycom config files, but guess g729 and ulaw are both preference 1 isn't it?<br /><br />
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 :)<br /><br />
set "canreinvite=yes" (or just comment it since that's the default) on both parties and try again.<br /><br />
Let me know if it works.<br /><br />
Alyed </font>
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<font face="Tahoma, Arial, Sans-Serif" size="2">
<hr align="center" size="2" width="100%" />Return-Path: <asterisk-users-bounces@lists.digium.com> Wed Sep 20 12:38:41 2006<br />Received: from digium-69-16-138-164.phx1.puregig.net [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 (localhost [127.0.0.1])<br /></font>
<br />Still having the same problem. i modified the sip.cfg in order to make<br />g729 the first choice:<br /><br /><codecs><br /><preferences voice.codecpref.g711mu="2"><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 <alyed.tzompa @simitel.com=""> 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: <asterisk-users-bounces @lists.digium.com=""> Tue<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 />> 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 />> 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 />--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 /></asterisk-users-bounces></alyed.tzompa></preferences></codecs>