Hi, I enabled sip debug and i get the following when i am trying to
call a polycom phone with the same sip.cfg I sent before (with g729 as
the primary codec):
--- (15 headers 9 lines)---
Using INVITE request as basis request - [EMAIL PROTECTED]
Sending to 10.14.34.130 : 5060 (NAT)
Found user '202'
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 10.14.34.130:10008
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x100 (g729), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
Sep 20 16:52:57 NOTICE[8226]: chan_sip.c:3691 process_sdp: No compatible codecs!
Transmitting (NAT) to 10.14.34.130:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP
10.14.34.130;branch=z9hG4bKa595a1daA5CED30F;received=10.14.34.130
From: "Santiago del Castillo" <sip:[EMAIL PROTECTED]>;tag=FF5B5B3D-F2E725F8
To: <sip:[EMAIL PROTECTED];user=phone>;tag=as5e963058
Call-ID: [EMAIL PROTECTED]
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[EMAIL PROTECTED]>
Content-Length: 0
Capabilities: us - 0x100 (g729), peer - audio=0xc
(ulaw|alaw)/video=0x0 (nothing), combined - 0x0 (nothing)
This line looks a little weird. As i understand peer should be the
other phone and the other phone has g729 enabled at sip.conf (asterisk
side) and sip.cfg (polycom phone side) And the line after that is what
i get without sip debug
Cheers!
Santiago
On 9/20/06, Delca <[EMAIL PROTECTED]> wrote:
Still having the same problem. i modified the sip.cfg in order to make
g729 the first choice:
<codecs>
<preferences voice.codecPref.G711Mu="2"
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 <[EMAIL PROTECTED]> 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: <[EMAIL PROTECTED]> 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
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