<div>I have already setup a list of prefered codec , but it's only for incoming call, not outgoing</div>
<div> </div>
<div>Laurent<br><br> </div>
<div><span class="gmail_quote">2007/7/17, Alex Balashov <<a href="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</a>>:</span>
<blockquote class="gmail_quote" style="PADDING-LEFT: 1ex; MARGIN: 0px 0px 0px 0.8ex; BORDER-LEFT: #ccc 1px solid">Laurent,<br><br> You should be able to set it with the 'codec' subcommand on the outgoing<br>dial peer as well. 'codec g711ulaw' or similar.
<br><br>-- Alex<br><br>On Tue, 17 Jul 2007, laurent schweizer wrote:<br><br>> Hello,<br>><br>> I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec<br>> in my ata the the GW return a media not acceptable error.
<br>><br>> but If i add the g729 codec the all is ok.<br>> I see in the config of the cisco where to define codec for imcoming call but<br>> not for outgoing<br>><br>> *Jul 17 15:57:02.604: Received:<br>
> INVITE <a href="mailto:sip:0041787518551@192.168.0.110">sip:0041787518551@192.168.0.110</a> SIP/2.0<br>> Via: SIP/2.0/UDP <a href="http://192.168.0.107:5070">192.168.0.107:5070</a>;branch=z9hG4bK5f66.fc82e301.0<br>
> To: <<a href="mailto:sip:0041787518551@192.168.0.110">sip:0041787518551@192.168.0.110</a>><br>> From: 021111111 <<a href="mailto:sip:021111111@peoplefone.ch">sip:021111111@peoplefone.ch</a><br>>> ;tag=27B98752-469CEA8A0002F2E4-5F903B30
<br>> CSeq: 10 INVITE<br>> Call-ID: <a href="mailto:1973211C-469CEA8A0002F2EA-5F903B30@212.203.123.82">1973211C-469CEA8A0002F2EA-5F903B30@212.203.123.82</a><br>> Content-Length: 250<br>> User-Agent: OpenSER (1.2.1-notls
(i386/linux))<br>> Contact: <sip:sems@192.168.0.107:5070><br>> P-MsgFlags: 0<br>> billingid: 106<br>> accountid: 28928<br>> Remote-Party-ID: <<a href="mailto:sip:0445532001@192.168.0.106">sip:0445532001@192.168.0.106
</a><br>>> ;party=calling;id-type=subscriber;screen=yes<br>> Content-Type: application/sdp<br>><br>> v=0<br>> o=MxSIP 0 198 IN IP4 <a href="http://192.168.0.249">192.168.0.249</a><br>> s=SIP Call<br>> c=IN IP4
<a href="http://200.200.100.106">200.200.100.106</a><br>> t=0 0<br>> m=audio 39318 RTP/AVP 8 0 101<br>> a=rtpmap:8 PCMA/8000<br>> a=rtpmap:0 PCMU/8000<br>> a=rtpmap:101 telephone-event/8000<br>> a=fmtp:101 0-15
<br>> a=sendrecv<br>> a=direction:active<br>> a=nortpproxy:yes<br>><br>> *Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_NONE,<br>> SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)<br>> *Jul 17 15:57:
02.608: sipSPIStreamTypeAndDtmfRelay: ERROR - no voice codec<br>> and no dtmf-relay match<br>> *Jul 17 15:57:02.608: sipSPIDoAudioNegotiation: Media negotiation failed for<br>> m-line 1<br>><br>> *Jul 17 15:57:
02.608: sipSPIDoMediaNegotiation: ERROR - no valid fax or<br>> audio streams<br>> *Jul 17 15:57:02.608: sipSPIHandleInviteMedia: Media Negotiation failed for<br>> an incoming call - Sending 488<br>><br>> *Jul 17 15:57:
02.608: 0x64C01D00 : State change from (STATE_IDLE,<br>> SUBSTATE_NONE) to (STATE_DISCONNECTING, SUBSTATE_NONE)<br>> *Jul 17 15:57:02.608: Sent:<br>> SIP/2.0 488 Not Acceptable Media<br>> Via: SIP/2.0/UDP <a href="http://192.168.0.107:5070">
192.168.0.107:5070</a>;branch=z9hG4bK5f66.fc82e301.0<br>> From: 021111111 <<a href="mailto:sip:021111111@peoplefone.ch">sip:021111111@peoplefone.ch</a><br>>> ;tag=27B98752-469CEA8A0002F2E4-5F903B30<br>> To: <
<a href="mailto:sip:0041787518551@192.168.0.110">sip:0041787518551@192.168.0.110</a>>;tag=C0E57710-2347<br>> Date: Tue, 17 Jul 2007 15:57:02 GMT<br>> Call-ID: <a href="mailto:1973211C-469CEA8A0002F2EA-5F903B30@212.203.123.82">
1973211C-469CEA8A0002F2EA-5F903B30@212.203.123.82</a><br>> Server: Cisco-SIPGateway/IOS-12.x<br>> CSeq: 10 INVITE<br>> Allow-Events: telephone-event<br>> Content-Length: 0<br>><br><br>--<br>Alex Balashov<br>
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