<html xmlns:v="urn:schemas-microsoft-com:vml" xmlns:o="urn:schemas-microsoft-com:office:office" xmlns:w="urn:schemas-microsoft-com:office:word" xmlns:m="http://schemas.microsoft.com/office/2004/12/omml" xmlns="http://www.w3.org/TR/REC-html40"><head><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=us-ascii"><meta name=Generator content="Microsoft Word 15 (filtered medium)"><style><!--
/* Font Definitions */
@font-face
{font-family:Wingdings;
panose-1:5 0 0 0 0 0 0 0 0 0;}
@font-face
{font-family:"Cambria Math";
panose-1:2 4 5 3 5 4 6 3 2 4;}
@font-face
{font-family:Calibri;
panose-1:2 15 5 2 2 2 4 3 2 4;}
/* Style Definitions */
p.MsoNormal, li.MsoNormal, div.MsoNormal
{margin:0in;
margin-bottom:.0001pt;
font-size:12.0pt;
font-family:"Times New Roman","serif";}
a:link, span.MsoHyperlink
{mso-style-priority:99;
color:blue;
text-decoration:underline;}
a:visited, span.MsoHyperlinkFollowed
{mso-style-priority:99;
color:purple;
text-decoration:underline;}
span.EmailStyle17
{mso-style-type:personal-reply;
font-family:"Calibri","sans-serif";
color:#1F497D;}
.MsoChpDefault
{mso-style-type:export-only;
font-size:10.0pt;}
@page WordSection1
{size:8.5in 11.0in;
margin:1.0in 1.0in 1.0in 1.0in;}
div.WordSection1
{page:WordSection1;}
--></style><!--[if gte mso 9]><xml>
<o:shapedefaults v:ext="edit" spidmax="1026" />
</xml><![endif]--><!--[if gte mso 9]><xml>
<o:shapelayout v:ext="edit">
<o:idmap v:ext="edit" data="1" />
</o:shapelayout></xml><![endif]--></head><body lang=EN-US link=blue vlink=purple><div class=WordSection1><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>TAC couldn’t find anything wrong with the running-config. They wanted me to redo the debugs as they don’t have 8961s or routers </span><span style='font-size:11.0pt;font-family:Wingdings;color:#1F497D'>J</span><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'>I just did some further testing. If you do a supervised transfer, it works perfectly. If you do an unsupervised transfer, the 3rd phone will have two calls active – One to the originating caller (transferred call) and one to the phone that did the transfer. If the phone that did the transfer ends the call, the transferred call remains up and working. Kind of weird behavior, but I think we can live with this and told them to close the case.<o:p></o:p></span></p><p class=MsoNormal><span style='font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1F497D'><o:p> </o:p></span></p><div><div style='border:none;border-top:solid #E1E1E1 1.0pt;padding:3.0pt 0in 0in 0in'><p class=MsoNormal><b><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'>From:</span></b><span style='font-size:11.0pt;font-family:"Calibri","sans-serif"'> Kenneth Hayes [mailto:kennethwhayes@gmail.com] <br><b>Sent:</b> Friday, November 08, 2013 6:39 PM<br><b>To:</b> Jason Aarons (AM)<br><b>Cc:</b> Jason Aarons (AM); cisco-voip (cisco-voip@puck.nether.net)<br><b>Subject:</b> Re: [cisco-voip] SIP CME-as-SRST transfer failing<o:p></o:p></span></p></div></div><p class=MsoNormal><o:p> </o:p></p><p class=MsoNormal>Jason,<br><br>Here's an example I got from this doc... I think the phones MAC needs<br>to be registered.<br><br>See the example:<br>voice register pool 2<br>id mac 0012.7F3B.9025 <-phone<br>number 1 2800<br>codec g711ulaw<br>!<br>voice register pool 3<br>id mac 0012.7F57.628F<br>number 1 2801<br>codec g711ulaw<br>!<br><br><br>On Fri, Nov 8, 2013 at 6:27 PM, Jason Aarons (AM)<br><<a href="mailto:jason.aarons@dimensiondata.com">jason.aarons@dimensiondata.com</a>> wrote: <br>> Here is the debug ccsip messages, notice the error down at 17:08:12.228 ?<br>> Researching it.<br>><br>><br>><br>> *Nov 8 17:08:12.224: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>><br>> Received:<br>><br>> INVITE <a href="sip:7454@10.30.2.253:5060">sip:7454@10.30.2.253:5060</a> SIP/2.0<br>><br>> Via: SIP/2.0/UDP 10.30.2.253:5060;branch=z9hG4bKBAFA43B<br>><br>> Remote-Party-ID: "7484"<br>> <<a href="sip:7484@10.30.2.253">sip:7484@10.30.2.253</a>>;party=calling;screen=yes;privacy=off<br>><br>> From: "7484" <<a href="sip:7484@10.27.16.11">sip:7484@10.27.16.11</a>>;tag=7C7715E4-81A<br>><br>> To: <<a href="sip:7454@10.30.2.253">sip:7454@10.30.2.253</a>><br>><br>> Date: Fri, 08 Nov 2013 23:08:12 GMT<br>><br>> Call-ID: <a href="mailto:79856A55-480111E3-AFB5CA0E-60C774C9@10.30.2.253">79856A55-480111E3-AFB5CA0E-60C774C9@10.30.2.253</a> <br>><br>> Supported: 100rel,timer,resource-priority,replaces<br>><br>> Min-SE: 1800<br>><br>> Cisco-Guid: 2038626741-1208029667-2947533326-1623684297<br>><br>> User-Agent: Cisco-SIPGateway/IOS-12.x<br>><br>> Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE,<br>> NOTIFY, INFO, REGISTER<br>><br>> CSeq: 101 INVITE<br>><br>> Timestamp: 1383952092<br>><br>> Contact: <<a href="sip:7484@10.30.2.253:5060">sip:7484@10.30.2.253:5060</a>><br>><br>> Call-Info:<br>> <<a href="sip:10.30.2.253:5060">sip:10.30.2.253:5060</a>>;method="NOTIFY;Event=telephone-event;Duration=2000"<br>><br>> Expires: 180<br>><br>> Allow-Events: telephone-event<br>><br>> Max-Forwards: 69<br>><br>> Content-Type: application/sdp<br>><br>> Content-Disposition: session;handling=required<br>><br>> Content-Length: 409<br>><br>><br>><br>> v=0<br>><br>> o=CiscoSystemsSIP-GW-UserAgent 3729 2253 IN IP4 10.30.2.253<br>><br>> s=SIP Call<br>><br>> c=IN IP4 10.30.2.253<br>><br>> t=0 0<br>><br>> m=audio 17618 RTP/AVP 0 18 100 101 121 19<br>><br>> c=IN IP4 10.30.2.253<br>><br>> a=rtpmap:0 PCMU/8000<br>><br>> a=rtpmap:18 G729/8000<br>><br>> a=fmtp:18 annexb=no<br>><br>> a=rtpmap:100 X-NSE/8000<br>><br>> a=fmtp:100 192-194<br>><br>> a=rtpmap:101 telephone-event/8000<br>><br>> a=fmtp:101 0-15<br>><br>> a=rtpmap:121 frf-dialed-digit/8000<br>><br>> a=fmtp:121 0-15<br>><br>> a=rtpmap:19 CN/8000<br>><br>><br>><br>> *Nov 8 17:08:12.228: %CALL_CONTROL-6-CALL_LOOP: The incoming call has a<br>> global identifier already present in the list of currently handled calls. It<br>> is being refused.<br>><br>><br>><br>> *Nov 8 17:08:12.244: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg:<br>><br>> Sent:<br>><br>> SIP/2.0 500 Internal Server Error<br>><br>> Via: SIP/2.0/UDP 10.30.2.253:5060;branch=z9hG4bKBAFA43B<br>><br>> From: "7484" <<a href="sip:7484@10.27.16.11">sip:7484@10.27.16.11</a>>;tag=7C7715E4-81A<br>><br>> To: <<a href="sip:7454@10.30.2.253">sip:7454@10.30.2.253</a>>;tag=7C7715EC-B31<br>><br>> Date: Fri, 08 Nov 2013 23:08:12 GMT<br>><br>> Call-ID: <a href="mailto:79856A55-480111E3-AFB5CA0E-60C774C9@10.30.2.253">79856A55-480111E3-AFB5CA0E-60C774C9@10.30.2.253</a> <br>><br>> Timestamp: 1383952092<br>><br>> CSeq: 101 INVITE<br>><br>> Allow-Events: telephone-event<br>><br>> Server: Cisco-SIPGateway/IOS-12.x<br>><br>> Content-Length: 0<br>><br>><br>><br>> From: cisco-voip [<a href="mailto:cisco-voip-bounces@puck.nether.net">mailto:cisco-voip-bounces@puck.nether.net</a>] On Behalf Of<br>> Jason Aarons (AM)<br>> Sent: Friday, November 08, 2013 6:16 PM<br>> To: Kenneth Hayes<br>><br>><br>> Cc: cisco-voip (<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>)<br>> Subject: Re: [cisco-voip] SIP CME-as-SRST transfer failing<br>><br>><br>><br>> This is 100% SIP. The 8961 doesn’t do SCCP.<br>><br>><br>><br>> From: Kenneth Hayes [<a href="mailto:kennethwhayes@gmail.com">mailto:kennethwhayes@gmail.com</a>]<br>> Sent: Friday, November 08, 2013 6:13 PM<br>> To: Jason Aarons (AM)<br>> Cc: cisco-voip (<a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a>)<br>> Subject: Re: [cisco-voip] SIP CME-as-SRST transfer failing<br>><br>><br>><br>><br>><br>> Where is your telephony-service?<br>><br>> Sent from my iPhone<br>><br>><br>> On Nov 8, 2013, at 6:00 PM, "Jason Aarons (AM)"<br>> <<a href="mailto:jason.aarons@dimensiondata.com">jason.aarons@dimensiondata.com</a>> wrote: <br>><br>><br>><br>> Ok what am I missing. SIP 8961s and transfer is failing whilst in SRST<br>> mode.<br>><br>><br>><br>><br>><br>> voice service voip<br>><br>> ip address trusted list<br>><br>> ipv4 192.168.16.10<br>><br>> ipv4 192.168.16.11<br>><br>> allow-connections h323 to h323<br>><br>> allow-connections h323 to sip<br>><br>> allow-connections sip to h323<br>><br>> allow-connections sip to sip<br>><br>> redirect ip2ip<br>><br>> fax protocol none<br>><br>> modem passthrough nse codec g711ulaw<br>><br>> sip<br>><br>> bind control source-interface GigabitEthernet0/1.30<br>><br>> bind media source-interface GigabitEthernet0/1.30<br>><br>> registrar server<br>><br>> !<br>><br>> voice class codec 99<br>><br>> codec preference 1 g711ulaw<br>><br>> codec preference 2 g729r8<br>><br>> !<br>><br>> voice class codec 1<br>><br>> codec preference 1 g711ulaw<br>><br>> codec preference 2 g729r8<br>><br>> !<br>><br>> voice class h323 1<br>><br>> h225 timeout tcp establish 3<br>><br>> !<br>><br>> !<br>><br>> voice register global<br>><br>> max-dn 480<br>><br>> max-pool 1<br>><br>> !<br>><br>> voice register pool 1<br>><br>> id network 192.168.2.0 mask 255.255.255.0<br>><br>> preference 1<br>><br>> incoming called-number<br>><br>> proxy 192.168.2.253 preference 1<br>><br>> dtmf-relay rtp-nte cisco-rtp sip-notify<br>><br>> voice-class codec 1<br>><br>> _______________________________________________<br>> cisco-voip mailing list<br>> <a href="mailto:cisco-voip@puck.nether.net">cisco-voip@puck.nether.net</a> <br>> <a href="https://puck.nether.net/mailman/listinfo/cisco-voip" target="_blank">https://puck.nether.net/mailman/listinfo/cisco-voip</a> <br>><br>><br>><br>> itevomcid<o:p></o:p></p></div></body></html>