[asterisk-users] media not accpetable with outgoing call on cisco

[Alex Balashov]

Laurent,

   You should be able to set it with the 'codec' subcommand on the outgoing 
dial peer as well.  'codec g711ulaw' or similar.

-- Alex

On Tue, 17 Jul 2007, laurent schweizer wrote:

> Hello,
>
> I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec
> in my ata the the GW return a media not acceptable error.
>
> but If i add the g729 codec the all is ok.
> I see in the config of the cisco where to define codec for imcoming call but
> not for outgoing
>
> *Jul 17 15:57:02.604: Received:
> INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0
> To: <sip:0041787518551 at 192.168.0.110>
> From: 021111111 <sip:021111111 at peoplefone.ch
>> ;tag=27B98752-469CEA8A0002F2E4-5F903B30
> CSeq: 10 INVITE
> Call-ID: 1973211C-469CEA8A0002F2EA-5F903B30 at 212.203.123.82
> Content-Length: 250
> User-Agent: OpenSER (1.2.1-notls (i386/linux))
> Contact: <sip:sems at 192.168.0.107:5070>
> P-MsgFlags: 0
> billingid: 106
> accountid: 28928
> Remote-Party-ID: <sip:0445532001 at 192.168.0.106
>> ;party=calling;id-type=subscriber;screen=yes
> Content-Type: application/sdp
>
> v=0
> o=MxSIP 0 198 IN IP4 192.168.0.249
> s=SIP Call
> c=IN IP4 200.200.100.106
> t=0 0
> m=audio 39318 RTP/AVP 8 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=direction:active
> a=nortpproxy:yes
>
> *Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_NONE,
> SUBSTATE_NONE)  to (STATE_IDLE, SUBSTATE_NONE)
> *Jul 17 15:57:02.608: sipSPIStreamTypeAndDtmfRelay: ERROR - no voice codec
> and no dtmf-relay match
> *Jul 17 15:57:02.608: sipSPIDoAudioNegotiation: Media negotiation failed for
> m-line 1
>
> *Jul 17 15:57:02.608: sipSPIDoMediaNegotiation: ERROR - no valid fax or
> audio streams
> *Jul 17 15:57:02.608: sipSPIHandleInviteMedia: Media Negotiation failed for
> an incoming call - Sending 488
>
> *Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_IDLE,
> SUBSTATE_NONE)  to (STATE_DISCONNECTING, SUBSTATE_NONE)
> *Jul 17 15:57:02.608: Sent:
> SIP/2.0 488 Not Acceptable Media
> Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0
> From: 021111111 <sip:021111111 at peoplefone.ch
>> ;tag=27B98752-469CEA8A0002F2E4-5F903B30
> To: <sip:0041787518551 at 192.168.0.110>;tag=C0E57710-2347
> Date: Tue, 17 Jul 2007 15:57:02 GMT
> Call-ID: 1973211C-469CEA8A0002F2EA-5F903B30 at 212.203.123.82
> Server: Cisco-SIPGateway/IOS-12.x
> CSeq: 10 INVITE
> Allow-Events: telephone-event
> Content-Length: 0
>

--
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : +1-678-954-0670
Direct : +1-678-954-0671