<div>I only have MTP configured on the CUCM server as part of the Device Pool. Do I need separate MTP configured on the router?</div>
<div> </div>
<div>Here is my debug. 7715 is the DID which I have configured on my soft phone. I used RDM to create base config with SIP trunks to gateways. Trunk is in the same CSS and Device Pool as the phone. </div>
<div> </div>
<div> </div>
<div>002000: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: RX <- SETUP pd = 8 callref = 0x0001<br> Bearer Capability i = 0x8090A2<br> Standard = CCITT<br> Transfer Capability = Speech<br>
Transfer Mode = Circuit<br> Transfer Rate = 64 kbit/s<br> Channel ID i = 0xA98381<br> Exclusive, Channel 1<br> Calling Party Number i = 0x2183, '9528183360'<br>
Plan:ISDN, Type:National<br> Called Party Number i = 0xC1, '7715'<br> Plan:ISDN, Type:Subscriber(local)<br>002001: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: Received SETUP callref = 0x8001 callID = 0x0042 switch = primary-5ess interface = User<br>
002002: Apr 26 16:31:12 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Sent:<br>INVITE <a href="http://sip:7715@172.21.20.10:5060">sip:7715@172.21.20.10:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5DDED4<br>
Remote-Party-ID: <<a href="mailto:sip%3A9528183360@172.21.20.254">sip:9528183360@172.21.20.254</a>>;party=calling;screen=yes;privacy=off<br>From: <<a href="mailto:sip%3A9528183360@172.21.20.254">sip:9528183360@172.21.20.254</a>>;tag=33617790-169F<br>
To: <<a href="mailto:sip%3A7715@172.21.20.10">sip:7715@172.21.20.10</a>><br>Date: Mon, 26 Apr 2010 21:31:12 GMT<br>Call-ID: <a href="mailto:DF28E48D-50B111DF-814BFADB-42390B@172.21.20.254">DF28E48D-50B111DF-814BFADB-42390B@172.21.20.254</a><br>
Supported: 100rel,timer,resource-priority,replaces,sdp-anat<br>Min-SE: 1800<br>Cisco-Guid: 3743959142-1353781727-2150402115-3779644176<br>User-Agent: Cisco-SIPGateway/IOS-12.x<br>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER<br>
CSeq: 101 INVITE<br>Max-Forwards: 70<br>Timestamp: 1272317472<br>Contact: <<a href="http://sip:9528183360@172.21.20.254:5060">sip:9528183360@172.21.20.254:5060</a>><br>Expires: 180<br>Allow-Events: telephone-event<br>
Content-Type: application/sdp<br>Content-Disposition: session;handling=required<br>Content-Length: 285</div>
<div>v=0<br>o=CiscoSystemsSIP-GW-UserAgent 9351 4649 IN IP4 172.21.20.254<br>s=SIP Call<br>c=IN IP4 172.21.20.254<br>t=0 0<br>m=audio 25022 RTP/AVP 0 18 101<br>c=IN IP4 172.21.20.254<br>a=rtpmap:0 PCMU/8000<br>a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16</div>
<div>002003: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0x8001<br> Channel ID i = 0xA98381<br> Exclusive, Channel 1<br>002004: Apr 26 16:31:12 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Received:<br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5DDED4<br>From: <<a href="mailto:sip%3A9528183360@172.21.20.254">sip:9528183360@172.21.20.254</a>>;tag=33617790-169F<br>To: <<a href="mailto:sip%3A7715@172.21.20.10">sip:7715@172.21.20.10</a>><br>
Date: Mon, 26 Apr 2010 21:31:31 GMT<br>Call-ID: <a href="mailto:DF28E48D-50B111DF-814BFADB-42390B@172.21.20.254">DF28E48D-50B111DF-814BFADB-42390B@172.21.20.254</a><br>CSeq: 101 INVITE<br>Allow-Events: presence<br>Content-Length: 0</div>
<div><br>002005: Apr 26 16:31:12 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5DDED4<br>From: <<a href="mailto:sip%3A9528183360@172.21.20.254">sip:9528183360@172.21.20.254</a>>;tag=33617790-169F<br>
To: <<a href="mailto:sip%3A7715@172.21.20.10">sip:7715@172.21.20.10</a>>;tag=205608936<br>Date: Mon, 26 Apr 2010 21:31:31 GMT<br>Call-ID: <a href="mailto:DF28E48D-50B111DF-814BFADB-42390B@172.21.20.254">DF28E48D-50B111DF-814BFADB-42390B@172.21.20.254</a><br>
CSeq: 101 INVITE<br>Allow-Events: presence<br>WWW-Authenticate: Digest realm="StandAloneCluster", nonce="oQbua7BD9UUXn7PtEyhEPuxTU4a5UWsT", algorithm=MD5<br>Content-Length: 0</div>
<div><br>002006: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: Applying typeplan for sw-type 0x3 is 0x2 0x1, Calling num 9528183360<br>002007: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: Sending SETUP callref = 0x0097 callID = 0x8018 switch = primary-5ess interface = User<br>
002008: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x0097<br> Bearer Capability i = 0x8090A2<br> Standard = CCITT<br> Transfer Capability = Speech<br>
Transfer Mode = Circuit<br> Transfer Rate = 64 kbit/s<br> Channel ID i = 0xA98397<br> Exclusive, Channel 23<br> Calling Party Number i = 0x2183, '9528183360'<br>
Plan:ISDN, Type:National<br> Called Party Number i = 0xC1, '7715'<br> Plan:ISDN, Type:Subscriber(local)<br>002009: Apr 26 16:31:12 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Sent:<br>ACK <a href="http://sip:7715@172.21.20.10:5060">sip:7715@172.21.20.10:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5DDED4<br>From: <<a href="mailto:sip%3A9528183360@172.21.20.254">sip:9528183360@172.21.20.254</a>>;tag=33617790-169F<br>
To: <<a href="mailto:sip%3A7715@172.21.20.10">sip:7715@172.21.20.10</a>>;tag=205608936<br>Date: Mon, 26 Apr 2010 21:31:12 GMT<br>Call-ID: <a href="mailto:DF28E48D-50B111DF-814BFADB-42390B@172.21.20.254">DF28E48D-50B111DF-814BFADB-42390B@172.21.20.254</a><br>
Max-Forwards: 70<br>CSeq: 101 ACK<br>Allow-Events: telephone-event<br>Content-Length: 0</div>
<div><br>002010: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x8097<br> Channel ID i = 0xA98397<br> Exclusive, Channel 23<br>UHD.EAST.RTR1#<br>002011: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: RX <- PROGRESS pd = 8 callref = 0x8097<br>
Cause i = 0x8283 - No route to destination<br> Progress Ind i = 0x8288 - In-band info or appropriate now available<br>002012: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8 callref = 0x8001<br>
Progress Ind i = 0x8288 - In-band info or appropriate now available<br>UHD.EAST.RTR1#<br>002013: Apr 26 16:31:18 Central: ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x0001<br> Cause i = 0x8290 - Normal call clearing<br>
</div>
<div> </div>
<div> </div>
<div><br><br> </div>
<div class="gmail_quote">On Mon, Apr 26, 2010 at 4:06 PM, miken miken <span dir="ltr"><<a href="mailto:miken@sisna.com">miken@sisna.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">MTP configured and box checked mandatory in SIP trunk configuration on CUCM?<br><br>Thanks<br>MikeN<br><br>
<div class="gmail_quote">
<div>
<div></div>
<div class="h5">On Mon, Apr 26, 2010 at 3:00 PM, Brian Schultz <span dir="ltr"><<a href="mailto:bms314@gmail.com" target="_blank">bms314@gmail.com</a>></span> wrote:<br></div></div>
<blockquote style="BORDER-LEFT: rgb(204,204,204) 1px solid; MARGIN: 0pt 0pt 0pt 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div>
<div></div>
<div class="h5">
<div>Yep, have that already. Gig0/1.110 has the IP address configured on the SIP trunk in CUCM. </div>
<div> </div>
<div>voice service voip<br> sip<br> bind control source-interface GigabitEthernet0/1.110<br> bind media source-interface GigabitEthernet0/1.110<br></div>
<div> </div>
<div>I also have the following:</div>
<div> </div>
<div>voice class codec 1<br> codec preference 1 g711ulaw<br> codec preference 2 g729r8<br></div>
<div>dial-peer voice 100 voip<br> destination-pattern ....<br> session protocol sipv2<br> session target ipv4:172.21.20.10<br> voice-class codec 1<br> dtmf-relay rtp-nte<br> no vad<br>dial-peer voice 1 pots<br> incoming called-number .<br>
direct-inward-dial<br></div>
<div>
<div></div>
<div>
<div><br><br> </div>
<div class="gmail_quote">On Mon, Apr 26, 2010 at 3:58 PM, Ahmed Elnagar <span dir="ltr"><<a href="mailto:ahmed_elnagar@rayacorp.com" target="_blank">ahmed_elnagar@rayacorp.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: rgb(204,204,204) 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div lang="EN-US" vlink="purple" link="blue">
<div>
<p class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 11pt">You have to bind signal and media traffic out of the router to the CUCM with the IP address you have configured on CUCM “by default CUCM reject calls with source address other than the configured”</span></p>
<p class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 11pt"> </span></p>
<p class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 11pt">Try the below on the gateway:</span></p>
<p class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 11pt"> </span></p>
<p class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 11pt">voice service voip</span></p>
<p class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 11pt">sip</span></p>
<p class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 11pt">bind all source-interface “interface configured on CUCM”</span></p>
<p class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 11pt"> </span></p>
<p class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 11pt"> </span></p>
<p class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 11pt"> </span></p>
<div>
<p style="BACKGROUND: white" class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 10pt"> </span><span style="COLOR: rgb(31,73,125); FONT-SIZE: 10.5pt">Best Regards;</span></p>
<p style="BACKGROUND: white" class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 10.5pt"> Ahmed Elnagar</span></p>
<p style="BACKGROUND: white" class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 10.5pt"> Senior Network PS Engineer</span></p>
<p style="BACKGROUND: white" class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 10.5pt"> Mob: +2019-0016211</span></p>
<p style="BACKGROUND: white" class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 10.5pt"> <img alt="ccie_voice_large.gif" width="63" height="63"><img alt="ccvp_voice_large.gif" width="63" height="63"></span></p>
</div>
<p class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 11pt"> </span></p>
<div style="BORDER-BOTTOM: medium none; BORDER-LEFT: medium none; PADDING-BOTTOM: 0in; PADDING-LEFT: 0in; PADDING-RIGHT: 0in; BORDER-TOP: rgb(181,196,223) 1pt solid; BORDER-RIGHT: medium none; PADDING-TOP: 3pt">
<p class="MsoNormal"><b><span style="FONT-SIZE: 10pt">From:</span></b><span style="FONT-SIZE: 10pt"> <a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a> [mailto:<a href="mailto:cisco-voip-bounces@puck.nether.net" target="_blank">cisco-voip-bounces@puck.nether.net</a>] <b>On Behalf Of </b>Brian Schultz<br>
<b>Sent:</b> Monday, April 26, 2010 10:34 PM<br><b>To:</b> <a href="mailto:cisco-voip@puck.nether.net" target="_blank">cisco-voip@puck.nether.net</a><br><b>Subject:</b> [cisco-voip] SIP gateway config</span></p></div>
<div>
<p class="MsoNormal"> </p>
<div>
<p class="MsoNormal">Does anyone happen to have an example SIP gateway config for an ISR? CUCM 8.0(2) with a SIP trunk to a 2921 gateway (15.0.M1.12) with a standard PRI for PSTN access. I have outbound working, but inbound gives a fast busy with a 401 Unauthorized in the SIP debug.</p>
</div>
<div>
<p class="MsoNormal"> </p></div>
<div>
<p class="MsoNormal">Thanks,</p></div>
<div>
<p class="MsoNormal">Brian</p></div>
<div>
<p class="MsoNormal"> </p></div>
<div>
<p class="MsoNormal"> </p></div></div></div>
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