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

[Keshav K.]

Hi,
Your invite is going with ulaw and alaw.
need to check that what are the entries of codecs in your sip.conf, have you allowed there ulaw and alaw or not, and next thing is if your gateway accepting, these codecs or not.

Keshav

laurent schweizer <laurent.schweizer at gmail.com> 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
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