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<TITLE>Re: [cisco-voip] sip trunk problem</TITLE>
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<FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>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>
<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 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>
<FONT COLOR="#1F497D">Chris,<BR>
Call starting from CUCM sccp iphones. I am not sure exactly what is my providers demand.<BR>
Thanks.<BR>
<BR>
</FONT><BR>
</SPAN></FONT><FONT SIZE="2"><FONT FACE="Tahoma, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'><B>From:</B> 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></FONT></FONT><FONT FACE="Times New Roman"><SPAN STYLE='font-size:12pt'> <BR>
</SPAN></FONT><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>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
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<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>
<FONT 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>
</FONT></SPAN><FONT COLOR="#1F497D"><FONT SIZE="1"><SPAN STYLE='font-size:8pt'>*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></FONT><SPAN STYLE='font-size:11pt'><BR>
</SPAN></FONT><SPAN STYLE='font-size:11pt'><BR>
</SPAN></FONT><FONT SIZE="2"><FONT FACE="Tahoma, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'><B>From:</B> 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></FONT></FONT><FONT FACE="Times New Roman"><SPAN STYLE='font-size:12pt'><BR>
</SPAN></FONT><FONT FACE="Calibri, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:11pt'>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
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<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>
<FONT COLOR="#1F497D">Call working one time ring after then dropping. <BR>
Here is the below “debug ccsip message”<BR>
</FONT></SPAN><FONT COLOR="#1F497D"><FONT SIZE="1"><SPAN STYLE='font-size:8pt'><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></FONT><SPAN STYLE='font-size:11pt'><BR>
</SPAN></FONT><SPAN STYLE='font-size:11pt'><BR>
</SPAN></FONT><FONT SIZE="2"><FONT FACE="Tahoma, Verdana, Helvetica, Arial"><SPAN STYLE='font-size:10pt'><B>From:</B> 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></FONT></FONT><FONT FACE="Times New Roman"><SPAN STYLE='font-size:12pt'><BR>
</SPAN></FONT><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>
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<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'>
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