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

[laurent schweizer]

I have already setup a list of prefered codec , but it's only for incoming
call, not outgoing

Laurent


2007/7/17, Alex Balashov <abalashov at evaristesys.com>:
>
> 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
>
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