<html><head><meta http-equiv="content-type" content="text/html; charset=utf-8"></head><body dir="auto"><div>What do your media resources look like?</div><div><br></div><div><br></div><div>Also can you show me a copy of your voice service voip config?<br>
<br>Sent from my iPad</div><div><br>On Jan 15, 2013, at 3:12 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com">dane.newman@gmail.com</a>> wrote:<br><br></div><blockquote type="cite"><div><div>Thanks Ryan</div>
<div> </div><div>I see I am always getting a 200 ok message after my invites from the debug</div><div> </div><div><span lang="EN"><strong>Putting a call on HOLD</strong></span></div><div><p> </p>
<p>*Jan 15 20:19:28.086: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>INVITE sip:17705439047@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 19:57:35 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Supported: timer,resource-priority,replaces</p>
<p>Min-SE: 1800</p>
<p>User-Agent: Cisco-CUCM8.6</p>
<p>Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY</p>
<p>CSeq: 102 INVITE</p>
<p>Max-Forwards: 70</p>
<p>Expires: 180</p>
<p>Allow-Events: presence</p>
<p>Supported: X-cisco-srtp-fallback</p>
<p>Supported: Geolocation</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>P-Asserted-Identity: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>></p>
<p>Remote-Party-ID: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;party=calling;screen=yes;privacy=off</p>
<p>Contact: <sip:6784563290@10.1.80.10:5060;transport=tcp>;video</p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 240</p>
<p>v=0</p>
<p>o=CiscoSystemsCCM-SIP 7322 3 IN IP4 10.1.80.10</p>
<p>s=SIP Call</p>
<p>c=IN IP4 0.0.0.0</p>
<p>b=TIAS:64000</p>
<p>b=AS:64</p>
<p>t=0 0</p>
<p>m=audio 21476 RTP/AVP 0 101</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=ptime:20</p>
<p>a=inactive</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-15</p>
<p>*Jan 15 20:19:28.094: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>INVITE <a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK691F12E0</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Supported: 100rel,timer,resource-priority,replaces,sdp-anat</p>
<p>Min-SE: 1800</p>
<p>Cisco-Guid: 3257897472-0000065536-0000000035-0173015306</p>
<p>User-Agent: Cisco-SIPGateway/IOS-12.x</p>
<p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p>
<p>CSeq: 103 INVITE</p>
<p>Max-Forwards: 70</p>
<p>Timestamp: 1358281168</p>
<p>Contact: <<a href="http://sip:6784563290@98.192.104.214:5060">sip:6784563290@98.192.104.214:5060</a>></p>
<p>Expires: 180</p>
<p>Allow-Events: telephone-event</p>
<p>Authorization: Digest username="6784563290",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 289</p>
<p>v=0</p>
<p>o=CiscoSystemsSIP-GW-UserAgent 3168 2739 IN IP4 98.192.104.214</p>
<p>s=SIP Call</p>
<p>c=IN IP4 98.192.104.214</p>
<p>t=0 0</p>
<p>m=audio 19458 RTP/AVP 0 101 19</p>
<p>c=IN IP4 98.192.104.214</p>
<p>a=inactive</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-15</p>
<p>a=rtpmap:19 CN/8000</p>
<p>a=ptime:20</p>
<p>*Jan 15 20:19:28.094: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>SIP/2.0 100 Trying</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>CSeq: 102 INVITE</p>
<p>Allow-Events: telephone-event</p>
<p>Server: Cisco-SIPGateway/IOS-12.x</p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:28.110: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>SIP/2.0 100 Trying</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK691F12E0;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>CSeq: 103 INVITE</p>
<p>Server: Asterisk PBX 1.6.2.13</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p>
<p>Require: timer</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150">sip:17705439047@64.154.41.150</a>></p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:28.110: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>SIP/2.0 200 OK</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK691F12E0;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>CSeq: 103 INVITE</p>
<p>Server: Asterisk PBX 1.6.2.13</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p>
<p>Require: timer</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150">sip:17705439047@64.154.41.150</a>></p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 239</p>
<p>v=0</p>
<p>o=root 1685873050 1685873052 IN IP4 64.154.41.150</p>
<p>s=Asterisk PBX 1.6.2.13</p>
<p>c=IN IP4 64.154.41.150</p>
<p>t=0 0</p>
<p>m=audio 13014 RTP/AVP 0 101</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-16</p>
<p>a=ptime:20</p>
<p>a=inactive</p>
<p>*Jan 15 20:19:28.118: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>SIP/2.0 200 OK</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK2891b89f8fa</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>CSeq: 102 INVITE</p>
<p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p>
<p>Allow-Events: telephone-event</p>
<p>Remote-Party-ID: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;party=called;screen=no;privacy=off</p>
<p>Contact: <sip:17705439047@10.1.200.1:5060;transport=tcp></p>
<p>Supported: replaces</p>
<p>Supported: sdp-anat</p>
<p>Server: Cisco-SIPGateway/IOS-12.x</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Require: timer</p>
<p>Supported: timer</p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 253</p>
<p>v=0</p>
<p>o=CiscoSystemsSIP-GW-UserAgent 4444 5479 IN IP4 10.1.200.1</p>
<p>s=SIP Call</p>
<p>c=IN IP4 10.1.200.1</p>
<p>t=0 0</p>
<p>m=audio 19514 RTP/AVP 0 101</p>
<p>c=IN IP4 10.1.200.1</p>
<p>a=inactive</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-16</p>
<p>a=ptime:20</p>
<p>*Jan 15 20:19:28.118: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>ACK <a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK6920266D</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Max-Forwards: 70</p>
<p>CSeq: 103 ACK</p>
<p>Authorization: Digest username="6784563290",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Allow-Events: telephone-event</p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:28.122: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>ACK sip:17705439047@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28b4b1305a0</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 19:57:35 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Max-Forwards: 70</p>
<p>CSeq: 102 ACK</p>
<p>Allow-Events: presence</p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:28.122: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>INVITE sip:17705439047@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 19:57:35 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Supported: timer,resource-priority,replaces</p>
<p>Min-SE: 1800</p>
<p>User-Agent: Cisco-CUCM8.6</p>
<p>Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY</p>
<p>CSeq: 103 INVITE</p>
<p>Max-Forwards: 70</p>
<p>Expires: 180</p>
<p>Allow-Events: presence</p>
<p>Supported: X-cisco-srtp-fallback</p>
<p>Supported: Geolocation</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>P-Asserted-Identity: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>></p>
<p>Remote-Party-ID: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;party=calling;screen=yes;privacy=off</p>
<p>Contact: <sip:6784563290@10.1.80.10:5060;transport=tcp>;video</p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:28.126: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>INVITE <a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69211AB3</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Supported: timer,resource-priority,replaces,sdp-anat</p>
<p>Min-SE: 1800</p>
<p>Cisco-Guid: 3257897472-0000065536-0000000035-0173015306</p>
<p>User-Agent: Cisco-SIPGateway/IOS-12.x</p>
<p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p>
<p>CSeq: 104 INVITE</p>
<p>Max-Forwards: 70</p>
<p>Timestamp: 1358281168</p>
<p>Contact: <<a href="http://sip:6784563290@98.192.104.214:5060">sip:6784563290@98.192.104.214:5060</a>></p>
<p>Expires: 180</p>
<p>Allow-Events: telephone-event</p>
<p>Authorization: Digest username="6784563290",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:28.126: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>SIP/2.0 100 Trying</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>CSeq: 103 INVITE</p>
<p>Allow-Events: telephone-event</p>
<p>Server: Cisco-SIPGateway/IOS-12.x</p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:28.146: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>SIP/2.0 100 Trying</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69211AB3;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>CSeq: 104 INVITE</p>
<p>Server: Asterisk PBX 1.6.2.13</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p>
<p>Require: timer</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150">sip:17705439047@64.154.41.150</a>></p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:28.146: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>SIP/2.0 200 OK</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69211AB3;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>CSeq: 104 INVITE</p>
<p>Server: Asterisk PBX 1.6.2.13</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p>
<p>Require: timer</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150">sip:17705439047@64.154.41.150</a>></p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 333</p>
<p>v=0</p>
<p>o=root 1685873050 1685873053 IN IP4 64.154.41.150</p>
<p>s=Asterisk PBX 1.6.2.13</p>
<p>c=IN IP4 64.154.41.150</p>
<p>t=0 0</p>
<p>m=audio 13014 RTP/AVP 3 8 0 18 101</p>
<p>a=rtpmap:3 GSM/8000</p>
<p>a=rtpmap:8 PCMA/8000</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:18 G729/8000</p>
<p>a=fmtp:18 annexb=no</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-16</p>
<p>a=ptime:20</p>
<p>a=inactive</p>
<p>*Jan 15 20:19:28.150: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>SIP/2.0 200 OK</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28c43b47e7e</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>CSeq: 103 INVITE</p>
<p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p>
<p>Allow-Events: telephone-event</p>
<p>Remote-Party-ID: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;party=called;screen=no;privacy=off</p>
<p>Contact: <sip:17705439047@10.1.200.1:5060;transport=tcp></p>
<p>Supported: replaces</p>
<p>Supported: sdp-anat</p>
<p>Server: Cisco-SIPGateway/IOS-12.x</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Require: timer</p>
<p>Supported: timer</p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 277</p>
<p>v=0</p>
<p>o=CiscoSystemsSIP-GW-UserAgent 4444 5480 IN IP4 10.1.200.1</p>
<p>s=SIP Call</p>
<p>c=IN IP4 10.1.200.1</p>
<p>t=0 0</p>
<p>m=audio 19514 RTP/AVP 0 101 19</p>
<p>c=IN IP4 10.1.200.1</p>
<p>a=inactive</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-16</p>
<p>a=rtpmap:19 CN/8000</p>
<p>a=ptime:20</p>
<p>*Jan 15 20:19:28.162: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>ACK sip:17705439047@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28d3eadaab3</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 19:57:35 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Max-Forwards: 70</p>
<p>CSeq: 103 ACK</p>
<p>Allow-Events: presence</p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 209</p>
<p>v=0</p>
<p>o=CiscoSystemsCCM-SIP 7322 4 IN IP4 10.1.80.10</p>
<p>s=SIP Call</p>
<p>c=IN IP4 0.0.0.0</p>
<p>b=TIAS:64000</p>
<p>b=AS:64</p>
<p>t=0 0</p>
<p>m=audio 21476 RTP/AVP 0</p>
<p>a=X-cisco-media:nomedia</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=ptime:20</p>
<p>a=inactive</p>
<p>*Jan 15 20:19:28.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>ACK <a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK692226EA</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Date: Tue, 15 Jan 2013 20:19:28 GMT</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Max-Forwards: 70</p>
<p>CSeq: 104 ACK</p>
<p>Authorization: Digest username="6784563290",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Allow-Events: telephone-event</p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 251</p>
<p>v=0</p>
<p>o=CiscoSystemsSIP-GW-UserAgent 3168 2740 IN IP4 98.192.104.214</p>
<p>s=SIP Call</p>
<p>c=IN IP4 0.0.0.0</p>
<p>t=0 0</p>
<p>m=audio 19458 RTP/AVP 0 101</p>
<p>c=IN IP4 0.0.0.0</p>
<p>a=inactive</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-16</p>
<p>a=ptime:20</p>
<p> </p><p><strong>Unholding the call the MOH continues on the previously held caller while the user hears nothing</strong></p><p><strong></strong> </p><p> </p>
<p>*Jan 15 20:19:35.166: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>INVITE sip:17705439047@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 19:57:42 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Supported: timer,resource-priority,replaces</p>
<p>Min-SE: 1800</p>
<p>User-Agent: Cisco-CUCM8.6</p>
<p>Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY</p>
<p>CSeq: 104 INVITE</p>
<p>Max-Forwards: 70</p>
<p>Expires: 180</p>
<p>Allow-Events: presence</p>
<p>Supported: X-cisco-srtp-fallback</p>
<p>Supported: Geolocation</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>P-Asserted-Identity: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>></p>
<p>Remote-Party-ID: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;party=calling;screen=yes;privacy=off</p>
<p>Contact: <sip:6784563290@10.1.80.10:5060;transport=tcp>;video;audio;video</p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:35.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>INVITE <a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Supported: timer,resource-priority,replaces,sdp-anat</p>
<p>Min-SE: 1800</p>
<p>Cisco-Guid: 3257897472-0000065536-0000000035-0173015306</p>
<p>User-Agent: Cisco-SIPGateway/IOS-12.x</p>
<p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p>
<p>CSeq: 105 INVITE</p>
<p>Max-Forwards: 70</p>
<p>Timestamp: 1358281175</p>
<p>Contact: <<a href="http://sip:6784563290@98.192.104.214:5060">sip:6784563290@98.192.104.214:5060</a>></p>
<p>Expires: 180</p>
<p>Allow-Events: telephone-event</p>
<p>Authorization: Digest username="6784563290",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:35.190: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>SIP/2.0 100 Trying</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>CSeq: 104 INVITE</p>
<p>Allow-Events: telephone-event</p>
<p>Server: Cisco-SIPGateway/IOS-12.x</p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>SIP/2.0 100 Trying</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>CSeq: 105 INVITE</p>
<p>Server: Asterisk PBX 1.6.2.13</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p>
<p>Require: timer</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150">sip:17705439047@64.154.41.150</a>></p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>SIP/2.0 200 OK</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>CSeq: 105 INVITE</p>
<p>Server: Asterisk PBX 1.6.2.13</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p>
<p>Require: timer</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150">sip:17705439047@64.154.41.150</a>></p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 333</p>
<p>v=0</p>
<p>o=root 1685873050 1685873054 IN IP4 64.154.41.150</p>
<p>s=Asterisk PBX 1.6.2.13</p>
<p>c=IN IP4 64.154.41.150</p>
<p>t=0 0</p>
<p>m=audio 13014 RTP/AVP 3 8 0 18 101</p>
<p>a=rtpmap:3 GSM/8000</p>
<p>a=rtpmap:8 PCMA/8000</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:18 G729/8000</p>
<p>a=fmtp:18 annexb=no</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-16</p>
<p>a=ptime:20</p>
<p>a=inactive</p>
<p>*Jan 15 20:19:35.198: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>SIP/2.0 200 OK</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>CSeq: 104 INVITE</p>
<p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p>
<p>Allow-Events: telephone-event</p>
<p>Remote-Party-ID: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;party=called;screen=no;privacy=off</p>
<p>Contact: <sip:17705439047@10.1.200.1:5060;transport=tcp></p>
<p>Supported: replaces</p>
<p>Supported: sdp-anat</p>
<p>Server: Cisco-SIPGateway/IOS-12.x</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Require: timer</p>
<p>Supported: timer</p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 277</p>
<p>v=0</p>
<p>o=CiscoSystemsSIP-GW-UserAgent 4444 5481 IN IP4 10.1.200.1</p>
<p>s=SIP Call</p>
<p>c=IN IP4 10.1.200.1</p>
<p>t=0 0</p>
<p>m=audio 19514 RTP/AVP 0 101 19</p>
<p>c=IN IP4 10.1.200.1</p>
<p>a=inactive</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-16</p>
<p>a=rtpmap:19 CN/8000</p>
<p>a=ptime:20</p>
<p>*Jan 15 20:19:35.206: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>ACK sip:17705439047@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28f6dca6616</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 19:57:42 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Max-Forwards: 70</p>
<p>CSeq: 104 ACK</p>
<p>Allow-Events: presence, kpml</p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 243</p>
<p>v=0</p>
<p>o=CiscoSystemsCCM-SIP 7322 5 IN IP4 10.1.80.10</p>
<p>s=SIP Call</p>
<p>c=IN IP4 10.1.10.18</p>
<p>b=TIAS:64000</p>
<p>b=AS:64</p>
<p>t=0 0</p>
<p>m=audio 21476 RTP/AVP 0 101</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=ptime:20</p>
<p>a=inactive</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-15</p>
<p>*Jan 15 20:19:35.210: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>ACK <a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69246AB</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Max-Forwards: 70</p>
<p>CSeq: 105 ACK</p>
<p>Authorization: Digest username="6784563290",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Allow-Events: telephone-event</p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 265</p>
<p>v=0</p>
<p>o=CiscoSystemsSIP-GW-UserAgent 3168 2741 IN IP4 98.192.104.214</p>
<p>s=SIP Call</p>
<p>c=IN IP4 98.192.104.214</p>
<p>t=0 0</p>
<p>m=audio 19458 RTP/AVP 0 101</p>
<p>c=IN IP4 98.192.104.214</p>
<p>a=inactive</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-16</p>
<p>a=ptime:20</p>
<p>Cisco3825#</p>
<p>Cisco3825#</p>
<p> </p>
<p>Cisco3825#</p><span lang="EN"></span></div><div><p> </p>
<p>INVITE sip:17705439047@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 19:57:42 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Supported: timer,resource-priority,replaces</p>
<p>Min-SE: 1800</p>
<p>User-Agent: Cisco-CUCM8.6</p>
<p>Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY</p>
<p>CSeq: 104 INVITE</p>
<p>Max-Forwards: 70</p>
<p>Expires: 180</p>
<p>Allow-Events: presence</p>
<p>Supported: X-cisco-srtp-fallback</p>
<p>Supported: Geolocation</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>P-Asserted-Identity: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>></p>
<p>Remote-Party-ID: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;party=calling;screen=yes;privacy=off</p>
<p>Contact: <sip:6784563290@10.1.80.10:5060;transport=tcp>;video;audio;video</p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:35.170: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>INVITE <a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Supported: timer,resource-priority,replaces,sdp-anat</p>
<p>Min-SE: 1800</p>
<p>Cisco-Guid: 3257897472-0000065536-0000000035-0173015306</p>
<p>User-Agent: Cisco-SIPGateway/IOS-12.x</p>
<p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p>
<p>CSeq: 105 INVITE</p>
<p>Max-Forwards: 70</p>
<p>Timestamp: 1358281175</p>
<p>Contact: <<a href="http://sip:6784563290@98.192.104.214:5060">sip:6784563290@98.192.104.214:5060</a>></p>
<p>Expires: 180</p>
<p>Allow-Events: telephone-event</p>
<p>Authorization: Digest username="6784563290",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:35.190: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>SIP/2.0 100 Trying</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>CSeq: 104 INVITE</p>
<p>Allow-Events: telephone-event</p>
<p>Server: Cisco-SIPGateway/IOS-12.x</p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>SIP/2.0 100 Trying</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>CSeq: 105 INVITE</p>
<p>Server: Asterisk PBX 1.6.2.13</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p>
<p>Require: timer</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150">sip:17705439047@64.154.41.150</a>></p>
<p>Content-Length: 0</p>
<p> </p>
<p>*Jan 15 20:19:35.194: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>SIP/2.0 200 OK</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69232672;received=98.192.104.214</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>CSeq: 105 INVITE</p>
<p>Server: Asterisk PBX 1.6.2.13</p>
<p>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO</p>
<p>Supported: replaces, timer</p>
<p>Require: timer</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Contact: <<a href="mailto:sip%3A17705439047@64.154.41.150">sip:17705439047@64.154.41.150</a>></p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 333</p>
<p>v=0</p>
<p>o=root 1685873050 1685873054 IN IP4 64.154.41.150</p>
<p>s=Asterisk PBX 1.6.2.13</p>
<p>c=IN IP4 64.154.41.150</p>
<p>t=0 0</p>
<p>m=audio 13014 RTP/AVP 3 8 0 18 101</p>
<p>a=rtpmap:3 GSM/8000</p>
<p>a=rtpmap:8 PCMA/8000</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:18 G729/8000</p>
<p>a=fmtp:18 annexb=no</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-16</p>
<p>a=ptime:20</p>
<p>a=inactive</p>
<p>*Jan 15 20:19:35.198: //14122/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>SIP/2.0 200 OK</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28e6157f41c</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>CSeq: 104 INVITE</p>
<p>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER</p>
<p>Allow-Events: telephone-event</p>
<p>Remote-Party-ID: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;party=called;screen=no;privacy=off</p>
<p>Contact: <sip:17705439047@10.1.200.1:5060;transport=tcp></p>
<p>Supported: replaces</p>
<p>Supported: sdp-anat</p>
<p>Server: Cisco-SIPGateway/IOS-12.x</p>
<p>Session-Expires: 1800;refresher=uas</p>
<p>Require: timer</p>
<p>Supported: timer</p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 277</p>
<p>v=0</p>
<p>o=CiscoSystemsSIP-GW-UserAgent 4444 5481 IN IP4 10.1.200.1</p>
<p>s=SIP Call</p>
<p>c=IN IP4 10.1.200.1</p>
<p>t=0 0</p>
<p>m=audio 19514 RTP/AVP 0 101 19</p>
<p>c=IN IP4 10.1.200.1</p>
<p>a=inactive</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-16</p>
<p>a=rtpmap:19 CN/8000</p>
<p>a=ptime:20</p>
<p>*Jan 15 20:19:35.206: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:</p>
<p>Received: </p>
<p>ACK sip:17705439047@10.1.200.1:5060;transport=tcp SIP/2.0</p>
<p>Via: SIP/2.0/TCP 10.1.80.10:5060;branch=z9hG4bK28f6dca6616</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@10.1.80.10">sip:6784563290@10.1.80.10</a>>;tag=7322~d8eefedd-7473-4e00-a4a0-ce8f65d30766-30500957</p>
<p>To: <<a href="mailto:sip%3A17705439047@10.1.200.1">sip:17705439047@10.1.200.1</a>>;tag=2E6BC6F0-1E22</p>
<p>Date: Tue, 15 Jan 2013 19:57:42 GMT</p>
<p>Call-ID: <a href="mailto:c22f9200-f51b49d-c5-a50010a@10.1.80.10">c22f9200-f51b49d-c5-a50010a@10.1.80.10</a></p>
<p>Max-Forwards: 70</p>
<p>CSeq: 104 ACK</p>
<p>Allow-Events: presence, kpml</p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 243</p>
<p>v=0</p>
<p>o=CiscoSystemsCCM-SIP 7322 5 IN IP4 10.1.80.10</p>
<p>s=SIP Call</p>
<p>c=IN IP4 10.1.10.18</p>
<p>b=TIAS:64000</p>
<p>b=AS:64</p>
<p>t=0 0</p>
<p>m=audio 21476 RTP/AVP 0 101</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=ptime:20</p>
<p>a=inactive</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-15</p>
<p>*Jan 15 20:19:35.210: //14124/C22F92000000/SIP/Msg/ccsipDisplayMsg:</p>
<p>Sent: </p>
<p>ACK <a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a> SIP/2.0</p>
<p>Via: SIP/2.0/UDP 98.192.104.214:5060;branch=z9hG4bK69246AB</p>
<p>From: "Dane Newman" <<a href="mailto:sip%3A6784563290@sipconnect.ipcomms.net">sip:6784563290@sipconnect.ipcomms.net</a>>;tag=2E6BC0B0-2268</p>
<p>To: <<a href="mailto:sip%3A17705439047@sip.talkinip.net">sip:17705439047@sip.talkinip.net</a>>;tag=as7c5ff82e</p>
<p>Date: Tue, 15 Jan 2013 20:19:35 GMT</p>
<p>Call-ID: <a href="mailto:A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214">A750328E-5E8711E2-8DC8A5CA-19425D52@98.192.104.214</a></p>
<p>Max-Forwards: 70</p>
<p>CSeq: 105 ACK</p>
<p>Authorization: Digest username="6784563290",realm="asterisk",uri="<a href="http://sip:17705439047@64.154.41.150:5060">sip:17705439047@64.154.41.150:5060</a>",response="0f6bb4f824d35b7315056c24e36a8709",nonce="206b7665",algorithm=MD5</p>
<p>Allow-Events: telephone-event</p>
<p>Content-Type: application/sdp</p>
<p>Content-Length: 265</p>
<p>v=0</p>
<p>o=CiscoSystemsSIP-GW-UserAgent 3168 2741 IN IP4 98.192.104.214</p>
<p>s=SIP Call</p>
<p>c=IN IP4 98.192.104.214</p>
<p>t=0 0</p>
<p>m=audio 19458 RTP/AVP 0 101</p>
<p>c=IN IP4 98.192.104.214</p>
<p>a=inactive</p>
<p>a=rtpmap:0 PCMU/8000</p>
<p>a=rtpmap:101 telephone-event/8000</p>
<p>a=fmtp:101 0-16</p>
<p>a=ptime:20</p>
<p>Cisco3825#</p><br><br></div><div class="gmail_quote">On Tue, Jan 15, 2013 at 2:28 PM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>></span> wrote:<br><blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote">
<div style="word-wrap:break-word">ccsip message is what I'd go with just to see the signaling with no other stuff. Depending on what that shows and what your gateway is doing to the signals you may need to expand from there.<div>
<span class="HOEnZb"><font color="#888888"><br><div>
<span style="text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;border-collapse:separate">-Ryan</span>
</div></font></span><div><div class="h5">
<br><div><div>On Jan 15, 2013, at 2:11 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:</div><br><div>Ryan</div><div> </div><div>What is the proper debug to use to caputre the useful information?</div>
<div> </div><div>Dane</div><div><br><br> </div><div class="gmail_quote">On Tue, Jan 15, 2013 at 12:42 PM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div style="word-wrap:break-word">Without sip messages I can't get any clues from that.<div>
<span><font color="#888888"><br><div>
<span style="text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;border-collapse:separate">-Ryan</span>
</div></font></span><div><div>
<br><div><div>On Jan 15, 2013, at 12:35 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:</div><br><div>Thanks Ryan for the input</div><div> </div><div> </div>
<div><strong>On the call when I hold the call the following debug pops out....</strong></div><div> </div><div><br>*Jan 15 17:56:05.246: //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add passthru hdrs to<br>
container<br>SIP: Attribute mid, level 1 instance 1 not found.<br>SIP: (13938) Group (a= group line) attribute, level 65535 instance 1 not found.<br>*Jan 15 17:56:05.274: //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add<br>
passthru headers to container<br>SIP: Attribute mid, level 1 instance 1 not found.<br>SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not found.<br>*Jan 15 17:56:05.286: //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add passthru hdrs to<br>
container<br>*Jan 15 17:56:05.302: //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add<br> passthru headers to container<br>SIP: Attribute mid, level 1 instance 1 not found.<br>
SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not found.<br>SIP: Attribute mid, level 1 instance 1 not found.<br>*Jan 15 17:56:05.322: //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could not modify QoS params for midcall INVITE<br>
</div><div> </div><div><strong>After I try to unhold the call the following debug comes out....</strong></div><div><strong></strong> </div><div><br>*Jan 15 17:56:18.874: //13939/922252E78D73/SIP/Error/ccsip_api_request_offer: Unable to add passthru hdrs to<br>
container<br>*Jan 15 17:56:18.894: //13938/922252E78D73/SIP/Error/ccsip_api_response_answer: Unable to add<br> passthru headers to container<br>SIP: Attribute mid, level 1 instance 1 not found.<br>
SIP: (13939) Group (a= group line) attribute, level 65535 instance 1 not found.<br>SIP: Attribute mid, level 1 instance 1 not found.<br>*Jan 15 17:56:18.906: //13939/922252E78D73/SIP/Error/sipSPIProcessAckMedia: Could not modify QoS params for midcall INVITE<br>
Cisco3825#<br>Cisco3825#<br>Cisco3825#<br><br></div><div class="gmail_quote">On Tue, Jan 15, 2013 at 9:42 AM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div style="word-wrap:break-word">Given you have an ITSP it's most likely the initial hold that's failing, which is only manifesting when you try to resume it. If you haven't noticed already this is also very likely causing transfers to fail.<div>
<br></div><div>Take a look at the SIP signaling for a call. I believe the most common cause to this is the ITSP not handling our transition from active->inactive->sendonly->active from hold to MOH to resume. The "Duplex Streaming Enabled" parameter is there just for this type of problem.</div>
<div><span><font color="#888888"><br><div>
<span style="text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;border-collapse:separate">-Ryan</span>
</div></font></span><div>
<br><div><div>On Jan 14, 2013, at 6:40 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:</div><br><div><strong>Hello Kenneth</strong></div><div><strong></strong> </div>
<div><strong>I have restarted both CUCM servers so this should have restarted the services when the utils system restart happened</strong></div><div><strong></strong> </div>
<div><br><strong>on my router I see I am using g711 from the debug </strong></div><div><strong></strong> </div><div><strong>I ran a debug voip ccapi inout </strong></div><div><strong></strong> </div><div><strong>I connected a call calling from an external number to a DiD inside of my system. Once the call was connected I put the call on hold and the following debug came out..the music on hold played for the external caller</strong></div>
<div> </div><div>*Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:<br> Stop Tone On Digit=FALSE, Tone=Null,<br> Tone Direction=Sum Network, Params=0x0, Call Id=12741<br>*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742, Xmit Function=0x64204BAC<br>*Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:</div><div>*Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702<br>
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br> Modem=0x0, Codec Bytes=20, Signal Type=2)<br>
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)<br>
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br>
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)<br>*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event=170, Call Id=12742<br>*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:<br> Feature Type=50, Interface=0xC05A65AC, Call Id=12742<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br> Modem=0x0, Codec Bytes=20, Signal Type=2)<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br>
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br>
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br>
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event=171, Call Id=12741<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:<br>
Interface=0xC05A65AC, Call Id=12742<br>*Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:<br> Stop Tone On Digit=FALSE, Tone=Null,<br> Tone Direction=Sum Network, Params=0x0, Call Id=12741<br>*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event=96, Call Id=12742<br>*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.839: //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741, Xmit Function=0x64204BAC<br>*Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:</div><div>*Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702<br>
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br> Modem=0x0, Codec Bytes=20, Signal Type=2)<br>
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br>
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event=170, Call Id=12741<br>*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br>
Modem=0x0, Codec Bytes=20, Signal Type=2)<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br>
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)<br>*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br>
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event=171, Call Id=12742<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:<br>
Interface=0xC05A65AC, Call Id=12742<br>Cisco3825#<br>Cisco3825#<br>Cisco3825#</div><div> </div><div> </div><div><strong>I then after that took off the hold and the following debug came out. The call on the PSDN side still played the hold music while there was no voice on the deskphone side.</strong></div>
<div> </div><div>*Jan 14 23:47:40.779: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:<br> Stop Tone On Digit=FALSE, Tone=Null,<br> Tone Direction=Sum Network, Params=0x0, Call Id=12741<br>*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742, Xmit Function=0x64204BAC<br>*Jan 14 23:47:40.783: //12741/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:</div><div>*Jan 14 23:47:40.783: cc_api_get_xcode_stream : 4702<br>
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br> Modem=0x0, Codec Bytes=20, Signal Type=2)<br>
*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)<br>
*Jan 14 23:47:40.783: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br>
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1046)<br>*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event=170, Call Id=12742<br>*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.783: //12742/9906C1828C05/CCAPI/cc_api_call_feature:<br> Feature Type=50, Interface=0xC05A65AC, Call Id=12742<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br> Modem=0x0, Codec Bytes=20, Signal Type=2)<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br>
Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br>
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>*Jan 14 23:47:40.811: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br>
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event=171, Call Id=12741<br>*Jan 14 23:47:40.811: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.815: //12742/9906C1828C05/CCAPI/cc_api_call_facility:<br>
Interface=0xC05A65AC, Call Id=12742<br>*Jan 14 23:47:40.819: //12741/9906C1828C05/CCAPI/ccGenerateToneInfo:<br> Stop Tone On Digit=FALSE, Tone=Null,<br> Tone Direction=Sum Network, Params=0x0, Call Id=12741<br>*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event=96, Call Id=12742<br>*Jan 14 23:47:40.819: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.839: //12742/9906C1828C05/CCAPI/cc_api_remote_codec_dnld_done:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741, Xmit Function=0x64204BAC<br>*Jan 14 23:47:40.839: //12742/xxxxxxxxxxxx/CCAPI/cc_api_get_xcode_stream:</div><div>*Jan 14 23:47:40.839: cc_api_get_xcode_stream : 4702<br>
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br> Modem=0x0, Codec Bytes=20, Signal Type=2)<br>
*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br>
Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>
*Jan 14 23:47:40.843: //12742/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br>
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=1516)<br>*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event=170, Call Id=12741<br>*Jan 14 23:47:40.843: //12741/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.859: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Destination Interface=0xC05A65AC, Destination Call Id=12741, Source Call Id=12742,<br> Caps(Codec=0x1, Fax Rate=0x2, Vad=0x2,<br>
Modem=0x0, Codec Bytes=20, Signal Type=2)<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_caps_ind:<br> Caps(Playout Mode=1, Playout Initial=60(ms), Playout Min=40(ms),<br> Playout Max=1000(ms), Fax Nom=300(ms))<br>
*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br> Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br>
Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)<br>*Jan 14 23:47:40.863: //12741/9906C1828C05/CCAPI/cc_api_caps_ack:<br> Destination Interface=0xC05A65AC, Destination Call Id=12742, Source Call Id=12741,<br>
Caps(Codec=g711ulaw(0x1), Fax Rate=FAX_RATE_NONE(0x1), Vad=ON(0x2),<br> Modem=OFF(0x0), Codec Bytes=160, Signal Type=2, Seq Num Start=3996)<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br>
Event=171, Call Id=12742<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_event_indication:<br> Event Is Sent To Conferenced SPI(s) Directly<br>*Jan 14 23:47:40.863: //12742/9906C1828C05/CCAPI/cc_api_call_facility:<br>
Interface=0xC05A65AC, Call Id=12742<br>Cisco3825#<br>Cisco3825#<br>Cisco3825#</div><div> </div><div class="gmail_quote">On Mon, Jan 14, 2013 at 6:20 PM, Kenneth Hayes <span dir="ltr"><<a href="mailto:kennethwhayes@gmail.com" target="_blank">kennethwhayes@gmail.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div dir="auto"><div>Have you also restarted the Cisco IP Media Services?<br>
<br>Sent from my iPhone</div><div><div><br>On Jan 14, 2013, at 6:12 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:<br>
<br></div><blockquote type="cite"><div>My ITSP will only support 711ulaw for me currently I believe. They hard coded it with me when I was initially setting it up.</div><div> </div><div>Do you think this could be a codec issue? How would I go about identifying if it is?</div>
<div> </div><div>Dane<br><br></div><div class="gmail_quote">On Mon, Jan 14, 2013 at 6:09 PM, Kenneth Hayes <span dir="ltr"><<a href="mailto:kennethwhayes@gmail.com" target="_blank">kennethwhayes@gmail.com</a>></span> wrote:<br>
<blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote"><div dir="auto"><div>Have you tried different audio codecs?<br>
<br>Sent from my iPhone</div><div><div><br>On Jan 14, 2013, at 6:06 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:<br>
<br></div><blockquote type="cite"><div>Ryan (sorry I forgot to reply to all)</div><div><br>Thanks for the Reply</div><div> </div><div>Oddly enough we are.</div><div> </div><div>This probably has something to do with MOH in general?</div>
<div><br>Internally when I user puts another user on hold everything works. No MOH plays and they can hold and unhold the call just fine.</div>
<div> </div><div>I tested calling from an external number. Once I put the external caller on hold the MOH played but I was unable to resume the call. When I hit resume on the deskphone the MOH still played to the external caller and there was no sound on the deskphone.<br>
<br></div><div class="gmail_quote">On Mon, Jan 14, 2013 at 5:25 PM, Ryan Ratliff <span dir="ltr"><<a href="mailto:rratliff@cisco.com" target="_blank">rratliff@cisco.com</a>></span> wrote:<br><blockquote style="margin:0px 0px 0px 0.8ex;padding-left:1ex;border-left-color:rgb(204,204,204);border-left-width:1px;border-left-style:solid" class="gmail_quote">
<div style="word-wrap:break-word">Do you get similar behavior if you just hold and resume the call outside SNR features? <div><span><font color="#888888"><br><div>
<span style="text-transform:none;text-indent:0px;letter-spacing:normal;word-spacing:0px;white-space:normal;border-collapse:separate">-Ryan</span>
</div></font></span><div>
<br><div><div>On Jan 14, 2013, at 4:18 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:</div><br><div>Using keyboard-interactive authentication. <br>
Password: <br><br>Cisco3825# <br>Cisco3825#sh ver <br>
Cisco IOS Software, 3800 Software (C3825-ADVENTERPRISEK9_IVS_LI-M), Version 15.1<br>(4)M5, RELEASE SOFTWARE (fc1) <br>Technical Support: <a href="http://www.cisco.com/techsupport" target="_blank">http://www.cisco.com/techsupport</a> <br>
Copyright (c) 1986-2012 by Cisco Systems, Inc. <br>Compiled Tue 04-Sep-12 17:25 by prod_rel_team <br><br>ROM: System Bootstrap, Version 12.4(13r)T10, RELEASE SOFTWARE (fc1) <br>
<br>Cisco3825 uptime is 1 week, 1 day, 1 hour, 38 minutes <br>System returned to ROM by power-on <br>System image file is "flash:c3825-adventerprisek9_ivs_li-mz.151-4.M5.bin" <br>
Last reload type: Normal Reload <br><br><br>This product contains cryptographic features and is subject to United <br>States and local country laws governing import, export, transfer and <br>
use. Delivery of Cisco cryptographic products does not imply <br>third-party authority to import, export, distribute or use encryption. <br>Importers, exporters, distributors and users are responsible for <br>
compliance with U.S. and local country laws. By using this product you <br>agree to comply with applicable laws and regulations. If you are unable <br>to comply with U.S. and local laws, return this product immediately. <br>
<br>A summary of U.S. laws governing Cisco cryptographic products may be found at: <br><a href="http://www.cisco.com/wwl/export/crypto/tool/stqrg.html" target="_blank">http://www.cisco.com/wwl/export/crypto/tool/stqrg.html</a> <br>
<br>If you require further assistance please contact us by sending email to <br><a href="mailto:export@cisco.com" target="_blank">export@cisco.com</a>. <br>
<br>Cisco 3825 (revision 1.2) with 1011712K/36864K bytes of memory. <br>Processor board ID FTX1237A1T0 <br>2 Gigabit Ethernet interfaces <br>
2 Channelized T1/PRI ports <br>1 Virtual Private Network (VPN) Module <br>DRAM configuration is 64 bits wide with parity enabled. <br>
479K bytes of NVRAM. <br>500472K bytes of ATA System CompactFlash (Read/Write) <br><br><br>License Info: <br>
<br>License UDI: <br><br>------------------------------------------------- <br>Device# PID SN <br>
<br>Sent from my mobile device<br><br>On Jan 14, 2013, at 4:11 PM, Kenneth Hayes <<a href="mailto:kennethwhayes@gmail.com" target="_blank">kennethwhayes@gmail.com</a>> wrote:<br><br><blockquote type="cite">What version of code are you running on the CUBE?<br>
<br>Sent from my iPhone<br><br>On Jan 14, 2013, at 3:43 PM, Dane Newman <<a href="mailto:dane.newman@gmail.com" target="_blank">dane.newman@gmail.com</a>> wrote:<br><br><blockquote type="cite">Hello<br><br>I have an issue when users are connected to a call and hit the mobility soft key button on 9971 phones when a call is active, the phone system rings on the mobile number configured in the system. When they pick up the the mobile number it just plays what sounds like hold music on both ends of the call (I believe this music is coming from cucm but I haven't heard it before) instead of providing 2 way voice.<br>
<br>In another senario with what I believe is the same issue. If a user picks up on there cell phone first (using single number reach) opposed to the deskphone the call is connected with 2 way voice and no issues exist. If the user then hangs up his cell phone with the intent to take the call on his deskphone the calling party starts hearing the hold music. Once the user picks up the call on his deskphone he hears nothing but the calling party is still hearing the hold music. It is not working as intended where 2 way voice happens once the user hangs up his mobile phone and picks up on his deskphone 2 way voice should happen.<br>
<br>My topology is as follows..<br><br><br>PSDN --> SIP TRUNK FROM ITSP --> 3825 CUBE --->CUCM -->DESKPHOHE<br><br>Calls are sent back out the SIP trunk to the ITSP when using mobile connect/snr.<br><br>Does anyone have any ideas how I can make 2 way voice happen instead of the hold music when the calls are picked up?<br>
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