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@192.168.0.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0 To: sip:0041787518551@192.168.0.110 From: 021111111 <sip:021111111@peoplefone.ch
;tag=27B98752-469CEA8A0002F2E4-5F903B30
CSeq: 10 INVITE Call-ID: 1973211C-469CEA8A0002F2EA-5F903B30@212.203.123.82 Content-Length: 250 User-Agent: OpenSER (1.2.1-notls (i386/linux)) Contact: sip:sems@192.168.0.107:5070 P-MsgFlags: 0 billingid: 106 accountid: 28928 Remote-Party-ID: <sip:0445532001@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@peoplefone.ch
;tag=27B98752-469CEA8A0002F2E4-5F903B30
To: sip:0041787518551@192.168.0.110;tag=C0E57710-2347 Date: Tue, 17 Jul 2007 15:57:02 GMT Call-ID: 1973211C-469CEA8A0002F2EA-5F903B30@212.203.123.82 Server: Cisco-SIPGateway/IOS-12.x CSeq: 10 INVITE Allow-Events: telephone-event Content-Length: 0
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@192.168.0.110 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.107:5070;branch=z9hG4bK5f66.fc82e301.0 To: sip:0041787518551@192.168.0.110 From: 021111111 <sip:021111111@peoplefone.ch
;tag=27B98752-469CEA8A0002F2E4-5F903B30
CSeq: 10 INVITE Call-ID: 1973211C-469CEA8A0002F2EA-5F903B30@212.203.123.82 Content-Length: 250 User-Agent: OpenSER (1.2.1-notls (i386/linux)) Contact: sip:sems@192.168.0.107:5070 P-MsgFlags: 0 billingid: 106 accountid: 28928 Remote-Party-ID: <sip:0445532001@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@peoplefone.ch
;tag=27B98752-469CEA8A0002F2E4-5F903B30
To: sip:0041787518551@192.168.0.110;tag=C0E57710-2347 Date: Tue, 17 Jul 2007 15:57:02 GMT Call-ID: 1973211C-469CEA8A0002F2EA-5F903B30@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
-
Alex Balashov -
laurent schweizer