didn't work :(
Regards, Santiago On 9/20/06, Alyed Tzompa <[EMAIL PROTECTED]> 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: <[EMAIL PROTECTED]> 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> > [EMAIL PROTECTED] > 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
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