<font size="3"><span style="color: rgb(0, 0, 127); font-family: times new roman;">Hi,</span><br style="color: rgb(0, 0, 127); font-family: times new roman;"><span style="color: rgb(0, 0, 127); font-family: times new roman;">Your invite is going with ulaw and alaw.</span><br style="color: rgb(0, 0, 127); font-family: times new roman;"><span style="color: rgb(0, 0, 127); font-family: times new roman;">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.</span><br style="color: rgb(0, 0, 127); font-family: times new roman;"><br style="color: rgb(0, 0, 127); font-family: times new roman;"><span style="color: rgb(0, 0, 127); font-family: times new roman;">Keshav</span></font><br><br><b><i>laurent schweizer <laurent.schweizer@gmail.com></i></b> wrote:<blockquote class="replbq" style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px;
padding-left: 5px;"> <div>Hello,</div> <div> </div> <div>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.</div> <div> </div> <div>but If i add the g729 codec the all is ok.</div> <div>I see in the config of the cisco where to define codec for imcoming call but not for outgoing</div> <div> </div> <div>*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> >;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>>;party=calling;id-type=subscriber;screen=yes<br>Content-Type: application/sdp</div> <div>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</div> <div>*Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_NONE, SUBSTATE_NONE) to (STATE_IDLE, SUBSTATE_NONE)<br>*Jul
17 15:57:02.608: sipSPIStreamTypeAndDtmfRelay: ERROR - no voice codec and no dtmf-relay match<br> *Jul 17 15:57:02.608: sipSPIDoAudioNegotiation: Media negotiation failed for m-line 1</div> <div>*Jul 17 15:57:02.608: sipSPIDoMediaNegotiation: ERROR - no valid fax or audio streams<br>*Jul 17 15:57:02.608: sipSPIHandleInviteMedia: Media Negotiation failed for an incoming call - Sending 488 </div> <div>*Jul 17 15:57:02.608: 0x64C01D00 : State change from (STATE_IDLE, 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>>;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</div> _______________________________________________<br>--Bandwidth and Colocation Provided by http://www.api-digital.com--<br><br>asterisk-users mailing list<br>To UNSUBSCRIBE or update options visit:<br> http://lists.digium.com/mailman/listinfo/asterisk-users</blockquote><br><p> 
<hr size=1>Shape Yahoo! in your own image.
<a href="http://us.rd.yahoo.com/evt=48517/*http://surveylink.yahoo.com/gmrs/yahoo_panel_invite.asp?a=7">Join our Network Research Panel today!</a>