I have already setup a list of prefered codec , but it's only for incoming call, not outgoing
Laurent 2007/7/17, Alex Balashov <[EMAIL PROTECTED]>:
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:[EMAIL PROTECTED] SIP/2.0 > Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0 > To: <sip:[EMAIL PROTECTED]> > From: 021111111 <sip:[EMAIL PROTECTED] >> ;tag=27B98752-469CEA8A0002F2E4-5F903B30 > CSeq: 10 INVITE > Call-ID: [EMAIL PROTECTED] > Content-Length: 250 > User-Agent: OpenSER (1.2.1-notls (i386/linux)) > Contact: <sip:[EMAIL PROTECTED]:5070> > P-MsgFlags: 0 > billingid: 106 > accountid: 28928 > Remote-Party-ID: <sip:[EMAIL PROTECTED] >> ;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:[EMAIL PROTECTED] >> ;tag=27B98752-469CEA8A0002F2E4-5F903B30 > To: <sip:[EMAIL PROTECTED]>;tag=C0E57710-2347 > Date: Tue, 17 Jul 2007 15:57:02 GMT > Call-ID: [EMAIL PROTECTED] > 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 _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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