<HTML>
<HEAD>
<TITLE>Re: [cisco-voip] sip trunk problem</TITLE>
</HEAD>
<BODY>
<FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>Looks like a media negotiation failure. The only thing that I can see is that your voice-class doesn’t have any G711 in it. Are we sure your provider isn’t looking for G711?<BR>
<BR>
It may be helpful to get a “debug ccsip message”.<BR>
<BR>
Chris Ward <BR>
<BR>
<BR>
<HR ALIGN=CENTER SIZE="3" WIDTH="95%"><B>From: </B>Baris Gulten <<a href="barisgulten@gmail.com">barisgulten@gmail.com</a>><BR>
<B>Date: </B>Mon, 2 Feb 2009 20:39:01 +0200<BR>
<B>To: </B><<a href="cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><BR>
<B>Subject: </B>[cisco-voip] sip trunk problem<BR>
<BR>
Hi everybody,<BR>
I have ccm 6.1.1, also 2851 router. <BR>
I define sip trunk on 2851 router. I have trouble when i decide make a call. Lets hold my hand </SPAN></FONT><SPAN STYLE='font-size:11pt'><FONT FACE="Wingdings">J<BR>
</FONT><FONT FACE="Calibri, Verdana, Helvetica, Arial"> <BR>
<BR>
<B>Here is the debug result<BR>
</B></FONT></SPAN><FONT FACE="Calibri, Verdana, Helvetica, Arial"><FONT SIZE="1"><SPAN STYLE='font-size:8pt'>vgw#sh debug<BR>
CCSIP SPI: SIP Call Events tracing is enabled (filter is OFF)<BR>
CCSIP SPI: SIP error debug tracing is enabled (filter is OFF<BR>
</SPAN></FONT><SPAN STYLE='font-size:11pt'>-------------<BR>
</SPAN><FONT SIZE="1"><SPAN STYLE='font-size:8pt'>#<BR>
*Feb 2 20:40:30.779: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_SETUP<BR>
*Feb 2 20:40:30.783: //109/803045A3FD12/SIP/Event/sipSPICreateRpid: Received Octet3A=0x83 -> Setting ;screen=yes ;privacy=off<BR>
*Feb 2 20:40:33.295: //109/803045A3FD12/SIP/Error/sipSPIDoAudioNegotiation: Media negotiation failed for m-line 1<BR>
*Feb 2 20:40:33.295: //109/803045A3FD12/SIP/Error/sipSPIDoMediaNegotiation: <BR>
no valid fax or audio streams<BR>
*Feb 2 20:40:33.295: //109/803045A3FD12/SIP/Error/ccsip_api_call_cut_progress: MediaNegotiation Failure - Send Cancel<BR>
*Feb 2 20:40:33.295: //-1/xxxxxxxxxxxx/SIP/Event/sipSPIEventInfo: Queued event from SIP SPI : SIPSPI_EV_CC_CALL_DISCONNECT<BR>
<BR>
</SPAN></FONT><SPAN STYLE='font-size:11pt'> <BR>
<B>Here is the config below (i did before cme router with this config)<BR>
</B></SPAN><FONT SIZE="1"><SPAN STYLE='font-size:8pt'>!<BR>
voice rtp send-recv<BR>
!<BR>
voice service voip <BR>
allow-connections h323 to h323<BR>
allow-connections h323 to sip<BR>
allow-connections sip to h323<BR>
allow-connections sip to sip<BR>
supplementary-service h450.12<BR>
h323<BR>
session transport tcp calls-per-connection 200<BR>
sip<BR>
bind control source-interface GigabitEthernet0/1<BR>
bind media source-interface GigabitEthernet0/1<BR>
registrar server expires max 3600 min 3600<BR>
no call service stop<BR>
!<BR>
voice class codec 1<BR>
codec preference 1 g729r8<BR>
codec preference 2 g729br8<BR>
codec preference 3 g723r63<BR>
codec preference 4 g723r53<BR>
codec preference 5 g726r24<BR>
codec preference 6 g726r16<BR>
codec preference 7 g726r32<BR>
codec preference 8 g723ar53<BR>
codec preference 9 g723ar63<BR>
! <BR>
interface GigabitEthernet0/1<BR>
ip address xxx.xxx.xxx.xxx xxx.xxx.xxx.xxx (defined real ip by sip provider)<BR>
duplex auto<BR>
speed auto<BR>
!<BR>
<BR>
!<BR>
dial-peer voice 100 voip<BR>
preference 1<BR>
voice-class codec 1<BR>
destination-pattern 053T<BR>
session protocol sipv2<BR>
session target ipv4:xxx.xxx.xxx.xxx (sip server ip)<BR>
dtmf-relay rtp-nte<BR>
clid network-number xxxxxxxxx<BR>
!<BR>
<BR>
<BR>
</SPAN></FONT><SPAN STYLE='font-size:11pt'><BR>
<HR ALIGN=CENTER SIZE="3" WIDTH="95%"></SPAN></FONT><FONT SIZE="2"><FONT FACE="Consolas, Courier New, Courier"><SPAN STYLE='font-size:10pt'>_______________________________________________<BR>
cisco-voip mailing list<BR>
<a href="cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><BR>
<a href="https://puck.nether.net/mailman/listinfo/cisco-voip">https://puck.nether.net/mailman/listinfo/cisco-voip</a><BR>
</SPAN></FONT></FONT>
</BODY>
</HTML>