<div>I'm going to start a new post since this is a different problem...</div>
<div> </div>
<div>Thanks,</div>
<div>Brian<br><br></div>
<div class="gmail_quote">On Mon, Apr 26, 2010 at 5:16 PM, Brian Schultz <span dir="ltr"><<a href="mailto:bms314@gmail.com">bms314@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<div>That was it, thanks Justin. Somehow digest authentication was checked on the SIP security profile.</div>
<div> </div>
<div>Now, I have a new problem. So far this problem has been intermittent. On some calls, the SIP setup messages don't start until 7 seconds after the call hits the gateway. I see the ISDN debug and hear a message from the provider that the call couldn't be completed as dialed. Then ISDN releases the call and I immediately see a SIP invite and my phone starts to ring and completes as normal. I've reproduced it several times already. Very strange. Here is an example:</div>
<div> </div>
<div> </div>
<div>002234: Apr 26 17:13:54 Central: ISDN Se0/0/0:23 Q931: TX -> CALL_PROC pd = 8 callref = 0x8001
<div class="im"><br> Channel ID i = 0xA98381<br> Exclusive, Channel 1<br></div>002235: Apr 26 17:13:54 Central: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x00A0
<div class="im"><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></div>002236: Apr 26 17:13:54 Central: ISDN Se0/0/0:23 Q931: RX <- CALL_PROC pd = 8 callref = 0x80A0
<div class="im"><br> Channel ID i = 0xA98397<br> Exclusive, Channel 23<br>UHD.EAST.RTR1#<br></div>002237: Apr 26 17:13:54 Central: ISDN Se0/0/0:23 Q931: RX <- PROGRESS pd = 8 callref = 0x80A0
<div class="im"><br> Cause i = 0x8283 - No route to destination<br> Progress Ind i = 0x8288 - In-band info or appropriate now available<br></div>002238: Apr 26 17:13:54 Central: ISDN Se0/0/0:23 Q931: TX -> ALERTING pd = 8 callref = 0x8001
<div class="im"><br> Progress Ind i = 0x8288 - In-band info or appropriate now available<br>UHD.EAST.RTR1#<br></div>002239: Apr 26 17:14:01 Central: ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x80A0
<div class="im"><br> Cause i = 0x8290 - Normal call clearing<br></div>002240: Apr 26 17:14:01 Central: ISDN Se0/0/0:23 Q931: TX -> RELEASE pd = 8 callref = 0x00A0<br>002241: Apr 26 17:14:01 Central: ISDN Se0/0/0:23 Q931: RX <- RELEASE_COMP pd = 8 callref = 0x80A0<br>
002242: Apr 26 17:14:01 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
<div class="im"><br>Sent:<br>INVITE <a href="http://sip:7715@172.21.20.10:5060" target="_blank">sip:7715@172.21.20.10:5060</a> SIP/2.0<br></div>Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5ED1B0D
<div class="im"><br>Remote-Party-ID: <<a href="mailto:sip%3A9528183360@172.21.20.254" target="_blank">sip:9528183360@172.21.20.254</a>>;party=calling;screen=yes;privacy=off<br></div>From: <<a href="mailto:sip%3A9528183360@172.21.20.254" target="_blank">sip:9528183360@172.21.20.254</a>>;tag=3388AB18-55
<div class="im"><br>To: <<a href="mailto:sip%3A7715@172.21.20.10" target="_blank">sip:7715@172.21.20.10</a>><br></div>Date: Mon, 26 Apr 2010 22:14:01 GMT<br>Call-ID: <a href="mailto:DA74D879-50B711DF-819BFADB-42390B@172.21.20.254" target="_blank">DA74D879-50B711DF-819BFADB-42390B@172.21.20.254</a>
<div class="im"><br>Supported: 100rel,timer,resource-priority,replaces,sdp-anat<br>Min-SE: 1800<br></div>Cisco-Guid: 3595010882-1354174943-2151188547-3779644176
<div class="im"><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></div>Timestamp: 1272320041
<div class="im"><br>Contact: <<a href="http://sip:9528183360@172.21.20.254:5060" target="_blank">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></div>Content-Length: 330</div>
<div>v=0<br>o=CiscoSystemsSIP-GW-UserAgent 5784 5366 IN IP4 172.21.20.254
<div class="im"><br>s=SIP Call<br>c=IN IP4 172.21.20.254<br>t=0 0<br></div>m=audio 26090 RTP/AVP 0 18 9 101
<div class="im"><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></div>a=rtpmap:9 G722/8000<br>a=fmtp:9 bitrate=64
<div class="im"><br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16</div></div>
<div>002243: Apr 26 17:14:01 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:
<div class="im"><br>Received:<br>SIP/2.0 100 Trying<br></div>Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5ED1B0D<br>From: <<a href="mailto:sip%3A9528183360@172.21.20.254" target="_blank">sip:9528183360@172.21.20.254</a>>;tag=3388AB18-55
<div class="im"><br>To: <<a href="mailto:sip%3A7715@172.21.20.10" target="_blank">sip:7715@172.21.20.10</a>><br></div>Date: Mon, 26 Apr 2010 22:14:20 GMT<br>Call-ID: <a href="mailto:DA74D879-50B711DF-819BFADB-42390B@172.21.20.254" target="_blank">DA74D879-50B711DF-819BFADB-42390B@172.21.20.254</a>
<div class="im"><br>CSeq: 101 INVITE<br>Allow-Events: presence<br>Content-Length: 0</div></div>
<div><br>002244: Apr 26 17:14:01 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>SIP/2.0 180 Ringing<br>Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5ED1B0D<br>From: <<a href="mailto:sip%3A9528183360@172.21.20.254" target="_blank">sip:9528183360@172.21.20.254</a>>;tag=3388AB18-55<br>
To: <<a href="mailto:sip%3A7715@172.21.20.10" target="_blank">sip:7715@172.21.20.10</a>>;tag=fc6aef00-fcb6-4e1c-a8c0-e38242535154-20774950<br>Date: Mon, 26 Apr 2010 22:14:20 GMT<br>Call-ID: <a href="mailto:DA74D879-50B711DF-819BFADB-42390B@172.21.20.254" target="_blank">DA74D879-50B711DF-819BFADB-42390B@172.21.20.254</a><br>
CSeq: 101 INVITE<br>Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY<br>Allow-Events: presence<br>Contact: <<a href="http://sip:7715@172.21.20.10:5060" target="_blank">sip:7715@172.21.20.10:5060</a>><br>
Supported: X-cisco-srtp-fallback</div>
<div>UHD.EAST.RTR1#Supported: Geolocation<br>P-Asserted-Identity: "Phenomenal Networks" <<a href="mailto:sip%3A7715@172.21.20.10" target="_blank">sip:7715@172.21.20.10</a>><br>Remote-Party-ID: "Phenomenal Networks" <<a href="mailto:sip%3A7715@172.21.20.10" target="_blank">sip:7715@172.21.20.10</a>>;party=called;screen=yes;privacy=off<br>
Content-Length: 0</div>
<div><br>UHD.EAST.RTR1#<br>002245: Apr 26 17:14:06 Central: ISDN Se0/0/0:23 Q931: RX <- DISCONNECT pd = 8 callref = 0x0001
<div class="im"><br> Cause i = 0x8290 - Normal call clearing<br></div>002246: Apr 26 17:14:06 Central: ISDN Se0/0/0:23 Q931: TX -> RELEASE pd = 8 callref = 0x8001<br>002247: Apr 26 17:14:06 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>
Sent:<br>CANCEL <a href="http://sip:7715@172.21.20.10:5060" target="_blank">sip:7715@172.21.20.10:5060</a> SIP/2.0<br>Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5ED1B0D<br>From: <<a href="mailto:sip%3A9528183360@172.21.20.254" target="_blank">sip:9528183360@172.21.20.254</a>>;tag=3388AB18-55
<div class="im"><br>To: <<a href="mailto:sip%3A7715@172.21.20.10" target="_blank">sip:7715@172.21.20.10</a>><br></div>Date: Mon, 26 Apr 2010 22:14:01 GMT<br>Call-ID: <a href="mailto:DA74D879-50B711DF-819BFADB-42390B@172.21.20.254" target="_blank">DA74D879-50B711DF-819BFADB-42390B@172.21.20.254</a><br>
CSeq: 101 CANCEL<br>Max-Forwards: 70<br>Timestamp: 1272320046<br>Reason: Q.850;cause=16<br>Content-Length: 0</div>
<div><br>002248: Apr 26 17:14:06 Central: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>Received:<br>SIP/2.0 200 OK<br>Via: SIP/2.0/UDP 172.21.20.254:5060;branch=z9hG4bK5ED1B0D<br>From: <<a href="mailto:sip%3A9528183360@172.21.20.254" target="_blank">sip:9528183360@172.21.20.254</a>>;tag=3388AB18-55
<div class="im"><br>To: <<a href="mailto:sip%3A7715@172.21.20.10" target="_blank">sip:7715@172.21.20.10</a>><br></div>Date: Mon, 26 Apr 2010 22:14:25 GMT<br>Call-ID: <a href="mailto:DA74D879-50B711DF-819BFADB-42390B@172.21.20.254" target="_blank">DA74D879-50B711DF-819BFADB-42390B@172.21.20.254</a><br>
CSeq: 101 CANCEL<br>Content-Length: 0<br><br><br></div>
<div>
<div></div>
<div class="h5">
<div class="gmail_quote">On Mon, Apr 26, 2010 at 5:01 PM, Justin Steinberg <span dir="ltr"><<a href="mailto:jsteinberg@gmail.com" target="_blank">jsteinberg@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: #ccc 1px solid; MARGIN: 0px 0px 0px 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">brian,<br><br>The CM trunk is challenging the IOS for digest authentication. take a look at the SIP security profile you've assigned to the CM SIP trunk and configure it for a profile that doesn't require digest authentication.<br>
<br>
<div class="gmail_quote">
<div>On Mon, Apr 26, 2010 at 5:56 PM, Peter Slow <span dir="ltr"><<a href="mailto:peter.slow@gmail.com" target="_blank">peter.slow@gmail.com</a>></span> wrote:<br></div>
<div>
<div></div>
<div>
<blockquote style="BORDER-LEFT: rgb(204,204,204) 1px solid; MARGIN: 0pt 0pt 0pt 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">Brian,
<div><br><br>002008: Apr 26 16:31:12 Central: ISDN Se0/0/0:23 Q931: TX -> SETUP pd = 8 callref = 0x0097<br></div><snip>
<div><br> Called Party Number i = 0xC1, '7715'<br> Plan:ISDN, Type:Subscriber(local)<br><br><br></div>-Pete
<div>
<div></div>
<div><br><br>
<div class="gmail_quote">On Mon, Apr 26, 2010 at 5:35 PM, Brian Schultz <span dir="ltr"><<a href="mailto:bms314@gmail.com" target="_blank">bms314@gmail.com</a>></span> wrote:<br>
<blockquote style="BORDER-LEFT: rgb(204,204,204) 1px solid; MARGIN: 0pt 0pt 0pt 0.8ex; PADDING-LEFT: 1ex" class="gmail_quote">
<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" target="_blank">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" target="_blank">sip:9528183360@172.21.20.254</a>>;party=calling;screen=yes;privacy=off<br>From: <<a href="mailto:sip%3A9528183360@172.21.20.254" target="_blank">sip:9528183360@172.21.20.254</a>>;tag=33617790-169F<br>
To: <<a href="mailto:sip%3A7715@172.21.20.10" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">sip:9528183360@172.21.20.254</a>>;tag=33617790-169F<br>
To: <<a href="mailto:sip%3A7715@172.21.20.10" target="_blank">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" target="_blank">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" target="_blank">sip:9528183360@172.21.20.254</a>>;tag=33617790-169F<br>
To: <<a href="mailto:sip%3A7715@172.21.20.10" target="_blank">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" target="_blank">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" target="_blank">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" target="_blank">sip:9528183360@172.21.20.254</a>>;tag=33617790-169F<br>
To: <<a href="mailto:sip%3A7715@172.21.20.10" target="_blank">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" target="_blank">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> </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" target="_blank">miken@sisna.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">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>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>
<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">
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<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>
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<p class="MsoNormal"><span style="COLOR: rgb(31,73,125); FONT-SIZE: 11pt">Try the below on the gateway:</span></p>
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<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>
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<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>
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<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>
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<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>
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<p class="MsoNormal">Thanks,</p></div>
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<p class="MsoNormal">Brian</p></div>
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