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<title>Re: [cisco-voip] sip trunk problem</title>
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<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Chris,<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Sccp Phones >CUCM >h323 gateway> router > sip trunk(outside
providers).<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>Thanks.<o:p></o:p></span></p>
<p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><o:p> </o:p></span></p>
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<p class=MsoNormal><b><span lang=EN-US style='font-size:10.0pt;font-family:
"Tahoma","sans-serif"'>From:</span></b><span lang=EN-US style='font-size:10.0pt;
font-family:"Tahoma","sans-serif"'> Chris Ward [mailto:chrward@cisco.com] <br>
<b>Sent:</b> Monday, February 02, 2009 11:01 PM<br>
<b>To:</b> Baris Gulten; cisco-voip@puck.nether.net<br>
<b>Subject:</b> Re: [cisco-voip] sip trunk problem<o:p></o:p></span></p>
</div>
</div>
<p class=MsoNormal><o:p> </o:p></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><span style='font-size:11.0pt;
font-family:"Calibri","sans-serif"'>How are calls being sent from the CUCM to
the GW?<br>
<br>
Chris Ward <br>
Cisco Systems Inc. <br>
Customer Support Engineer<br>
Unified Communication Infrastructure<br>
Boxborough, MA <br>
9:00am - 6:00pm Eastern <br>
978-936-0217<br>
<a href="chrward@cisco.com">chrward@cisco.com</a><br>
<br>
<o:p></o:p></span></p>
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style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>
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<p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>From:
</span></b><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>Baris
Gulten <<a href="barisgulten@gmail.com">barisgulten@gmail.com</a>><br>
<b>Date: </b>Mon, 2 Feb 2009 22:22:24 +0200<br>
<b>To: </b>Chris Ward <<a href="chrward@cisco.com">chrward@cisco.com</a>>,
<<a href="cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
<b>Subject: </b>RE: [cisco-voip] sip trunk problem<br>
<br>
<span style='color:#1F497D'>Chris,<br>
Call starting from CUCM sccp iphones. I am not sure exactly what is my
providers demand.<br>
Thanks.<br>
<br>
</span><br>
</span><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Chris Ward [<a
href="mailto:chrward@cisco.com">mailto:chrward@cisco.com</a>] <br>
<b>Sent:</b> Monday, February 02, 2009 10:14 PM<br>
<b>To:</b> Baris Gulten; <a href="cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] sip trunk problem<br>
</span><br>
<span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>Baris,<br>
<br>
Does your provider require early media? I do see that you are not sending SDP
in the initial INVITE.<br>
<br>
Also, how is this call being sent from CUCM? H323?<br>
<br>
Chris Ward </span><o:p></o:p></p>
<div class=MsoNormal align=center style='text-align:center'><span
style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>
<hr size=3 width="95%" align=center>
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<p><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
<b>From: </b>Baris Gulten <<a href="barisgulten@gmail.com">barisgulten@gmail.com</a>><br>
<b>Date: </b>Mon, 2 Feb 2009 21:45:55 +0200<br>
<b>To: </b>Chris Ward <<a href="chrward@cisco.com">chrward@cisco.com</a>>,
<<a href="cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
<b>Subject: </b>RE: [cisco-voip] sip trunk problem<br>
<br>
<span style='color:#1F497D'>Hi Chris,<br>
I set up g729br8 codec also ringing working but when i off-hook after
then calls dropping.<br>
Here is the below debug result.(ccsip message), thanks.<br>
<br>
</span></span><span style='font-size:8.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'>*Feb 2 21:29:51.075:
//-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Sent: <br>
INVITE sip:0xxx@xxx0:5060 SIP/2.0<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK2B1562<br>
From: <sip: 0xxx@xxx>;tag=619AD0-1324<br>
To: <sip:0xxx@xxx><br>
Date: Mon, 02 Feb 2009 21:29:51 GMT<br>
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5@xxx<br>
Supported: 100rel,timer,replaces<br>
Min-SE: 1800<br>
Cisco-Guid: 2153288071-2118939032-437513217-2886732291<br>
User-Agent: Cisco-SIPGateway/IOS-12.x<br>
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO, UPDATE, REGISTER<br>
CSeq: 101 INVITE<br>
Max-Forwards: 70<br>
Remote-Party-ID: <sip:0xxx@xxx2>;party=calling;screen=yes;privacy=off<br>
Timestamp: 1233610191<br>
Contact: <sip:0xxx@xxx2:5060><br>
Expires: 180<br>
Allow-Events: telephone-event<br>
<br>
*Feb 2 21:29:51.147: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received: <br>
SIP/2.0 100 Trying<br>
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5@xxx<br>
From: <sip:0xxx@xxx>;tag=619AD0-1324<br>
To: <sip:0xxx@xxx>;tag=3367<br>
CSeq: 101 INVITE<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK2B1562<br>
Supported: timer,100rel<br>
Content-Length: 0<br>
<br>
*Feb 2 21:29:53.211: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received: <br>
SIP/2.0 183 Session Progress<br>
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5@xxx<br>
From: <sip:0xxx@xxx>;tag=619AD0-1324<br>
To: <sip:0xxx@xxx>;tag=3367<br>
CSeq: 101 INVITE<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK2B1562<br>
Supported: timer,100rel<br>
Content-Length: 0<br>
<br>
*Feb 2 21:29:53.243: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received: <br>
SIP/2.0 183 Session Progress<br>
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5@xxx<br>
From: <sip:0xxx@xxx>;tag=619AD0-1324<br>
To: <sip:0xxx@xxx>;tag=3367<br>
Content-Type: application/sdp<br>
CSeq: 101 INVITE<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK2B1562<br>
Supported: timer,100rel<br>
Content-Length: 258<br>
<br>
v=0<br>
o=MG4000|2.0 56404 56404 IN IP4 xxx<br>
s=-<br>
c=IN IP4 xxx<br>
t=0 0<br>
m=audio 48180 RTP/AVP 18 97 101 13<br>
a=rtpmap:97 G.729b/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=fmtp:18 annexb=yes<br>
a=ptime:10<br>
a=rtpmap:13 CN/8000<br>
<br>
*Feb 2 21:30:02.819: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Sent: <br>
CANCEL sip:0xxx@xxx:5060 SIP/2.0<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK2B1562<br>
From: <sip:0xxx@xxx>;tag=619AD0-1324<br>
To: <sip:0xxx@xxx><br>
Date: Mon, 02 Feb 2009 21:29:51 GMT<br>
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5@xxx<br>
CSeq: 101 CANCEL<br>
Max-Forwards: 70<br>
Timestamp: 1233610202<br>
Content-Length: 0<br>
<br>
*Feb 2 21:30:02.879: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received: <br>
SIP/2.0 200 OK<br>
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5@xxx<br>
From: <sip:0xxx@xxx>;tag=619AD0-1324<br>
To: <sip:0xxx@xxx>;tag=8819<br>
CSeq: 101 CANCEL<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK2B1562<br>
Contact: sip:0xxx@xxx:5060;user=phone<br>
Supported: timer,100rel<br>
Content-Length: 0<br>
<br>
*Feb 2 21:30:02.887: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received: <br>
SIP/2.0 487 Request Terminated<br>
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5@xxx<br>
From: <sip:0xxx@xxx2>;tag=619AD0-1324<br>
To: <sip:0xxx@xxx>;tag=3367<br>
CSeq: 101 INVITE<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK2B1562<br>
Supported: timer,100rel<br>
Content-Length: 0<br>
<br>
*Feb 2 21:30:02.887: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Sent: <br>
ACK sip:0xxx@xxx:5060 SIP/2.0<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK2B1562<br>
From: <sip:0xxx@xxx>;tag=619AD0-1324<br>
To: <sip:0xxx@xxx>;tag=3367<br>
Date: Mon, 02 Feb 2009 21:29:51 GMT<br>
Call-ID: 7565385F-F0A711DD-803E8100-58B252E5@xxx<br>
Max-Forwards: 70<br>
CSeq: 101 ACK<br>
Content-Length: 0<br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Chris Ward [<a
href="mailto:chrward@cisco.com">mailto:chrward@cisco.com</a>] <br>
<b>Sent:</b> Monday, February 02, 2009 9:03 PM<br>
<b>To:</b> Baris Gulten; <a href="cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] sip trunk problem<br>
</span><br>
<span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>Hi Baris,<br>
<br>
As a test, can you try and remove the voice-class codec from the dial-peer and
add a specific codec?<br>
<br>
Try this:<br>
<br>
Codec g729br8<br>
<br>
Looks like this is the codec the provider is wanting. I know its in your
voice-class codec list, but I would still try it.<br>
<br>
Chris Ward </span><o:p></o:p></p>
<div class=MsoNormal align=center style='text-align:center'><span
style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>
<hr size=3 width="95%" align=center>
</span></div>
<p><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
<br>
<b>From: </b>Baris Gulten <<a href="barisgulten@gmail.com">barisgulten@gmail.com</a>><br>
<b>Date: </b>Mon, 2 Feb 2009 20:58:45 +0200<br>
<b>To: </b>Chris Ward <<a href="chrward@cisco.com">chrward@cisco.com</a>>,
<<a href="cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>><br>
<b>Subject: </b>RE: [cisco-voip] sip trunk problem<br>
<br>
<span style='color:#1F497D'>Call working one time ring after then dropping. <br>
Here is the below “debug ccsip message”<br>
</span></span><span style='font-size:8.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><br>
*Feb 2 20:52:18.951: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Sent: <br>
INVITE sip:0xxxxxx@xxx:5060 SIP/2.0<br>
Via: SIP/2.0/UDP xxxx:5060;branch=z9hG4bK151BC<br>
From: <sip: 0xxxxxx@xxx>;tag=3F3D70-212E<br>
To: <sip: 0xxxxxx@xxx><br>
Date: Mon, 02 Feb 2009 20:52:18 GMT<br>
Call-ID: <a href="3705C6EF-F0A211DD-802E8100-58B252E5@84.44.99.162">3705C6EF-F0A211DD-802E8100-58B252E5@84.44.99.162</a><br>
Supported: 100rel,timer,replaces<br>
Min-SE: 1800<br>
Cisco-Guid: 2154448201-2990764440-101963777-2886732291<br>
User-Agent: Cisco-SIPGateway/IOS-12.x<br>
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO, UPDATE, REGISTER<br>
CSeq: 101 INVITE<br>
Max-Forwards: 70<br>
Remote-Party-ID: <sip: 0xxxxxx@xxx>;party=calling;screen=yes;privacy=off<br>
Timestamp: 1233607938<br>
Contact: <sip: 0xxxxxx@xxx:5060><br>
Expires: 180<br>
Allow-Events: telephone-event<br>
<br>
*Feb 2 20:52:19.031: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received: <br>
SIP/2.0 100 Trying<br>
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5@xxx<br>
From: <sip: 0xxxxxx@xxx>;tag=3F3D70-212E<br>
To: <sip: 0xxxxxx@xxx>;tag=28846<br>
CSeq: 101 INVITE<br>
Via: SIP/2.0/UDP xxxx:5060;branch=z9hG4bK151BC<br>
Supported: timer,100rel<br>
Content-Length: 0<br>
<br>
*Feb 2 20:52:21.207: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received: <br>
SIP/2.0 183 Session Progress<br>
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5@xxx<br>
From: <sip: 0xxxxxx@xxx>;tag=3F3D70-212E<br>
To: <sip: 0xxxxxx@xxx>;tag=28846<br>
CSeq: 101 INVITE<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK151BC<br>
Supported: timer,100rel<br>
Content-Length: 0<br>
<br>
*Feb 2 20:52:21.243: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received: <br>
SIP/2.0 183 Session Progress<br>
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5@xxx<br>
From: <sip: 0xxxxxx@xxx>;tag=3F3D70-212E<br>
To: <sip: 0xxxxxx@xxx>;tag=28846<br>
Content-Type: application/sdp<br>
CSeq: 101 INVITE<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK151BC<br>
Supported: timer,100rel<br>
Content-Length: 258<br>
<br>
v=0<br>
o=MG4000|2.0 99854 99854 IN IP4 xxx<br>
s=-<br>
c=IN IP4 62.244.254.131<br>
t=0 0<br>
m=audio 48296 RTP/AVP 18 97 101 13<br>
a=rtpmap:97 G.729b/8000<br>
a=rtpmap:101 telephone-event/8000<br>
a=fmtp:101 0-15<br>
a=fmtp:18 annexb=yes<br>
a=ptime:10<br>
a=rtpmap:13 CN/8000<br>
<br>
*Feb 2 20:52:21.247: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Sent: <br>
CANCEL sip: 0xxxxxx@xxx:5060 SIP/2.0<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK151BC<br>
From: <sip: 0xxxxxx@xxx>;tag=3F3D70-212E<br>
To: <sip: 0xxxxxx@xxx><br>
Date: Mon, 02 Feb 2009 20:52:18 GMT<br>
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5@xxx<br>
CSeq: 101 CANCEL<br>
Max-Forwards: 70<br>
Timestamp: 1233607941<br>
Content-Length: 0<br>
<br>
*Feb 2 20:52:21.307: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received: <br>
SIP/2.0 200 OK<br>
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5@xxx<br>
From: <sip: 0xxxxxx@xxx>;tag=3F3D70-212E<br>
To: <sip: 0xxxxxx@xxx>;tag=29754<br>
CSeq: 101 CANCEL<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK151BC<br>
Contact: sip: 0xxxxxx@xxx:5060;user=phone<br>
Supported: timer,100rel<br>
Content-Length: 0<br>
<br>
*Feb 2 20:52:21.315: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received: <br>
SIP/2.0 487 Request Terminated<br>
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5@xxx<br>
From: <sip: 0xxxxxx@xxx>;tag=3F3D70-212E<br>
To: <sip: 0xxxxxx@xxx>;tag=28846<br>
CSeq: 101 INVITE<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK151BC<br>
Supported: timer,100rel<br>
Content-Length: 0<br>
<br>
*Feb 2 20:52:21.319: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Sent: <br>
ACK sip: 0xxxxxx@xxx:5060 SIP/2.0<br>
Via: SIP/2.0/UDP xxx:5060;branch=z9hG4bK151BC<br>
From: <sip: 0xxxxxx@xxx >;tag=3F3D70-212E<br>
To: <sip: 0xxxxxx@xxx >;tag=28846<br>
Date: Mon, 02 Feb 2009 20:52:18 GMT<br>
Call-ID: 3705C6EF-F0A211DD-802E8100-58B252E5@xxx<br>
Max-Forwards: 70<br>
CSeq: 101 ACK<br>
Content-Length: 0<br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";
color:#1F497D'><br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><b><span style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'>From:</span></b><span
style='font-size:10.0pt;font-family:"Tahoma","sans-serif"'> Chris Ward [<a
href="mailto:chrward@cisco.com">mailto:chrward@cisco.com</a>] <br>
<b>Sent:</b> Monday, February 02, 2009 8:44 PM<br>
<b>To:</b> Baris Gulten; <a href="cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a><br>
<b>Subject:</b> Re: [cisco-voip] sip trunk problem<br>
</span><br>
<span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>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 </span><o:p></o:p></p>
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style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>
<hr size=3 width="95%" align=center>
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<p><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
<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><span style='font-size:11.0pt;font-family:Wingdings'>J<br>
</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
<br>
<b>Here is the debug result<br>
</b></span><span style='font-size:8.0pt;font-family:"Calibri","sans-serif"'>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><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>-------------<br>
</span><span style='font-size:8.0pt;font-family:"Calibri","sans-serif"'>#<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><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
<b>Here is the config below (i did before cme router with this config)<br>
</b></span><span style='font-size:8.0pt;font-family:"Calibri","sans-serif"'>!<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>
</span><o:p></o:p></p>
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<p style='margin-bottom:12.0pt'><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'><br>
</span><span style='font-size:10.0pt;font-family:Consolas'>_______________________________________________<br>
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