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SDP looks ok to me, but there is one warning about ptime in iLBC below. I don't see how a wrong codec can be selected because I narrowed down my external profile inbound/outbound to PCMU only and my internal is iLBC@30i only.<br><br><br>freeswitch@kuffel> recv 1206 bytes from udp/[74.63.41.218]:5060 at 17:59:32.187500:<br> ------------------------------------------------------------------------<br> INVITE sip:gw+voip.ms@69.125.20.15:5080;transport=udp;gw=voip.ms SIP/2.0<br> Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport<br> From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64<br> To: <sip:gw+voip.ms@69.125.20.15:5080;transport=udp;gw=voip.ms><br> Contact: <sip:MYPHONE#@74.63.41.218><br> Call-ID: 1298959c0099d50d177b6bae689fc028@74.63.41.218<br> CSeq: 102 INVITE<br> User-Agent: VoIPMS/SERAST<br> Max-Forwards: 70<br> Remote-Party-ID: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;privacy=off;screen=no<br> Date: Mon, 31 Jan 2011 17:59:17 GMT<br> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br> Supported: replaces<br> Content-Type: application/sdp<br> Content-Length: 515<br> <br> v=0<br> o=root 2831 2831 IN IP4 74.63.41.218<br> s=session<br> c=IN IP4 74.63.41.218<br> t=0 0<br> m=audio 16884 RTP/AVP 0 4 3 8 112 5 10 7 18 111 101<br> a=rtpmap:0 PCMU/8000<br> a=rtpmap:4 G723/8000<br> a=fmtp:4 annexa=no<br> a=rtpmap:3 GSM/8000<br> a=rtpmap:8 PCMA/8000<br> a=rtpmap:112 AAL2-G726-32/8000<br> a=rtpmap:5 DVI4/8000<br> a=rtpmap:10 L16/8000<br> a=rtpmap:7 LPC/8000<br> a=rtpmap:18 G729/8000<br> a=fmtp:18 annexb=no<br> a=rtpmap:111 G726-32/8000<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-16<br> a=silenceSupp:off - - - -<br> a=ptime:20<br> a=sendrecv<br> ------------------------------------------------------------------------<br>send 396 bytes to udp/[74.63.41.218]:5060 at 17:59:32.187500:<br> ------------------------------------------------------------------------<br> SIP/2.0 100 Trying<br> Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060<br> From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64<br> To: <sip:gw+voip.ms@69.125.20.15:5080;transport=udp;gw=voip.ms><br> Call-ID: 1298959c0099d50d177b6bae689fc028@74.63.41.218<br> CSeq: 102 INVITE<br> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600<br> Content-Length: 0<br> <br> ------------------------------------------------------------------------<br>2011-01-31 12:59:32.187500 [NOTICE] switch_channel.c:808 New Channel sofia/external/MYPHONE#@74.63.41.218 [f35f408a-f863-4784-a308-8b4fb3284b70]<br>2011-01-31 12:59:32.187500 [INFO] mod_dialplan_xml.c:331 Processing MYPHONE# <MYPHONE#>->121628 in context public<br>2011-01-31 12:59:32.203125 [NOTICE] switch_ivr.c:1606 Transfer sofia/external/MYPHONE#@74.63.41.218 to XML[1001@default]<br>2011-01-31 12:59:32.203125 [INFO] mod_dialplan_xml.c:331 Processing MYPHONE# <MYPHONE#>->1001 in context default<br>2011-01-31 12:59:32.234375 [NOTICE] switch_channel.c:808 New Channel sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155:5060 [7230b9e8-37a7-4fc6-9b52-25740a6f7ca4]<br><br><br>2011-01-31 12:59:32.265625 [WARNING] sofia_glue.c:213 Codec iLBC payload 98 added to sdp wanting ptime 30 but it's already 20 (PCMU:0:20), disabling ptime.<br><br><br>send 1315 bytes to tcp/[32.140.14.196]:46743 at 17:59:32.265625:<br> ------------------------------------------------------------------------<br> INVITE sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155:5060;transport=TCP SIP/2.0<br> Via: SIP/2.0/TCP 69.125.20.15;rport;branch=z9hG4bK797eFQ6rgQKmQ<br> Route: <sip:M9jdt73ig0oOJSbt6Uyy@32.140.14.196:46743>;transport=TCP<br> Max-Forwards: 68<br> From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa<br> To: <sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155:5060;transport=TCP><br> Call-ID: b281a1fa-a806-122e-f799-c1188a708e17<br> CSeq: 7912322 INVITE<br> Contact: <sip:mod_sofia@69.125.20.15:5060;transport=tcp><br> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE<br> Supported: timer, precondition, path, replaces<br> Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer<br> Content-Type: application/sdp<br> Content-Disposition: session<br> Content-Length: 234<br> X-FS-Support: update_display<br> Remote-Party-ID: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;party=calling;screen=no;privacy=off<br> <br> v=0<br> o=FreeSWITCH 1296474878 1296474879 IN IP4 69.125.20.15<br> s=FreeSWITCH<br> c=IN IP4 69.125.20.15<br> t=0 0<br> m=audio 21894 RTP/AVP 0 98 101 13<br> a=rtpmap:98 iLBC/8000<br> a=fmtp:98 mode=30<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-16<br> ------------------------------------------------------------------------<br>recv 318 bytes from tcp/[32.140.14.196]:46743 at 17:59:37.093750:<br> ------------------------------------------------------------------------<br> SIP/2.0 100 Trying<br> Via: SIP/2.0/TCP 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15<br> To: <sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155><br> From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa<br> Call-ID: b281a1fa-a806-122e-f799-c1188a708e17<br> CSeq: 7912322 INVITE<br> Content-Length: 0<br> <br> ------------------------------------------------------------------------<br>recv 476 bytes from tcp/[32.140.14.196]:46743 at 17:59:42.296875:<br> ------------------------------------------------------------------------<br> SIP/2.0 180 Ringing<br> Via: SIP/2.0/TCP 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15<br> Contact: <sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155:5060;transport=TCP><br> From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa<br> To: <sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155>;tag=p4rl1jbfvmnbvfs1d5rktoj2<br> Supported: 100rel<br> Call-ID: b281a1fa-a806-122e-f799-c1188a708e17<br> CSeq: 7912322 INVITE<br> Allow: INVITE,ACK,CANCEL,OPTIONS,BYE<br> Content-Length: 0<br> <br> ------------------------------------------------------------------------<br>2011-01-31 12:59:42.296875 [INFO] sofia.c:729 sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155:5060 Update Callee ID to "Outbound Call" <M9jdt73ig0oOJSbt6Uyy><br>2011-01-31 12:59:42.296875 [NOTICE] sofia.c:4724 Ring-Ready sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155:5060!<br>2011-01-31 12:59:42.312500 [INFO] switch_ivr_originate.c:1101 Sending early media<br>2011-01-31 12:59:42.343750 [NOTICE] mod_sofia.c:2252 Pre-Answer sofia/external/MYPHONE#@74.63.41.218!<br>send 1079 bytes to udp/[74.63.41.218]:5060 at 17:59:42.343750:<br> ------------------------------------------------------------------------<br> SIP/2.0 183 Session Progress<br> Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060<br> From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64<br> To: <sip:gw+voip.ms@69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S<br> Call-ID: 1298959c0099d50d177b6bae689fc028@74.63.41.218<br> CSeq: 102 INVITE<br> Contact: <sip:gw+voip.ms@69.125.20.15:5080;transport=udp><br> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600<br> Accept: application/sdp<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY<br> Supported: timer, precondition, path, replaces<br> Allow-Events: talk, hold, refer<br> Content-Type: application/sdp<br> Content-Disposition: session<br> Content-Length: 247<br> Remote-Party-ID: "121628" <sip:121628@192.168.1.100>;party=calling;privacy=off;screen=no<br> <br> v=0<br> o=FreeSWITCH 1296476876 1296476877 IN IP4 69.125.20.15<br> s=FreeSWITCH<br> c=IN IP4 69.125.20.15<br> t=0 0<br> m=audio 19906 RTP/AVP 0 101<br> a=rtpmap:0 PCMU/8000<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-16<br> a=silenceSupp:off - - - -<br> a=ptime:20<br> ------------------------------------------------------------------------<br>recv 772 bytes from tcp/[32.140.14.196]:46743 at 17:59:43.812500:<br> ------------------------------------------------------------------------<br> SIP/2.0 200 OK<br> Via: SIP/2.0/TCP 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15<br> To: <sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155>;tag=p4rl1jbfvmnbvfs1d5rktoj2<br> Contact: <sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155:5060;transport=TCP><br> From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa<br> Supported: timer<br> Call-ID: b281a1fa-a806-122e-f799-c1188a708e17<br> CSeq: 7912322 INVITE<br> Allow: INVITE,ACK,CANCEL,OPTIONS,BYE<br> Content-Type: application/sdp<br> Content-Length: 269<br> <br> v=0<br> o=M9jdt73ig0oOJSbt6Uyy 63464734759229750 63464734759229750 IN IP4 10.208.245.155<br> s=-<br> c=IN IP4 10.208.245.155<br> t=0 0<br> m=audio 49152 RTP/AVP 98 101<br> a=sendrecv<br> a=rtpmap:98 iLBC/8000<br> a=ptime:30<br> a=maxptime:180<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-15<br> ------------------------------------------------------------------------<br>send 464 bytes to tcp/[32.140.14.196]:46743 at 17:59:43.828125:<br> ------------------------------------------------------------------------<br> ACK sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155:5060;transport=TCP SIP/2.0<br> Via: SIP/2.0/TCP 69.125.20.15;rport;branch=z9hG4bK8j17gjQvD096j<br> Max-Forwards: 70<br> From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa<br> To: <sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155:5060;transport=TCP>;tag=p4rl1jbfvmnbvfs1d5rktoj2<br> Call-ID: b281a1fa-a806-122e-f799-c1188a708e17<br> CSeq: 7912322 ACK<br> Contact: <sip:mod_sofia@69.125.20.15:5060;transport=tcp><br> Content-Length: 0<br> <br> ------------------------------------------------------------------------<br>2011-01-31 12:59:43.828125 [NOTICE] sofia.c:5230 Channel [sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155:5060] has been answered<br>send 1061 bytes to udp/[74.63.41.218]:5060 at 17:59:43.843750:<br> ------------------------------------------------------------------------<br> SIP/2.0 200 OK<br> Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK30d7505b;rport=5060<br> From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64<br> To: <sip:gw+voip.ms@69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S<br> Call-ID: 1298959c0099d50d177b6bae689fc028@74.63.41.218<br> CSeq: 102 INVITE<br> Contact: <sip:gw+voip.ms@69.125.20.15:5080;transport=udp><br>2011-01-31 12:59:43.843750 [NOTICE] switch_ivr_originate.c:3328 Channel [sofia/external/MYPHONE#@74.63.41.218] has been answered<br> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05 -0600<br> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY<br> Supported: timer, precondition, path, replaces<br> Allow-Events: talk, hold, refer<br> Content-Type: application/sdp<br> Content-Disposition: session<br> Content-Length: 247<br> Remote-Party-ID: "Outbound Call" <sip:M9jdt73ig0oOJSbt6Uyy@192.168.1.100>;party=calling;privacy=off;screen=no<br> <br> v=0<br> o=FreeSWITCH 1296476876 1296476877 IN IP4 69.125.20.15<br> s=FreeSWITCH<br> c=IN IP4 69.125.20.15<br> t=0 0<br> m=audio 19906 RTP/AVP 0 101<br> a=rtpmap:0 PCMU/8000<br> a=rtpmap:101 telephone-event/8000<br> a=fmtp:101 0-16<br> a=silenceSupp:off - - - -<br> a=ptime:20<br> ------------------------------------------------------------------------<br>recv 533 bytes from udp/[74.63.41.218]:5060 at 17:59:43.859375:<br> ------------------------------------------------------------------------<br> ACK sip:gw+voip.ms@69.125.20.15:5080;transport=udp SIP/2.0<br> Via: SIP/2.0/UDP 74.63.41.218:5060;branch=z9hG4bK0cbbc6ef;rport<br> From: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;tag=as66f1bf64<br> To: <sip:gw+voip.ms@69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S<br> Contact: <sip:MYPHONE#@74.63.41.218><br> Call-ID: 1298959c0099d50d177b6bae689fc028@74.63.41.218<br> CSeq: 102 ACK<br> User-Agent: VoIPMS/SERAST<br> Max-Forwards: 70<br> Remote-Party-ID: "MYPHONE#" <sip:MYPHONE#@74.63.41.218>;privacy=off;screen=no<br> Content-Length: 0<br><br><hr id="stopSpelling">From: robert.hadley@teotech.com<br>To: freeswitch-users@lists.freeswitch.org<br>Date: Mon, 31 Jan 2011 09:33:48 -0800<br>Subject: Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true.<br><br>
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</style><div class="ecxWordSection1"><p class="ecxMsoNormal"><span style="font-size: 11pt; font-family: 'Calibri','sans-serif'; color: rgb(31, 73, 125);"> </span></p><p class="ecxMsoNormal"><span style="font-size: 11pt; font-family: 'Calibri','sans-serif'; color: rgb(31, 73, 125);">Check the codecs in the SDP or try manual hardcoding the codecs presented for both legs, we had a squeal problem going to a softphone that turned out to be the BV32 codec was being selected instead of SPEEX16.</span></p><p class="ecxMsoNormal"><span style="font-size: 11pt; font-family: 'Calibri','sans-serif'; color: rgb(31, 73, 125);"> </span></p><p class="ecxMsoNormal"><span style="font-size: 11pt; font-family: 'Calibri','sans-serif'; color: rgb(31, 73, 125);">Robert</span></p><p class="ecxMsoNormal"><span style="font-size: 11pt; font-family: 'Calibri','sans-serif'; color: rgb(31, 73, 125);"> </span></p><div><div style="border-right: medium none; border-width: 1pt medium medium; border-style: solid none none; border-color: rgb(181, 196, 223) -moz-use-text-color -moz-use-text-color; padding: 3pt 0in 0in;"><p class="ecxMsoNormal"><b><span style="font-size: 10pt; font-family: 'Tahoma','sans-serif';">From:</span></b><span style="font-size: 10pt; font-family: 'Tahoma','sans-serif';"> Marcin Wojtowicz [mailto:marcin321@hotmail.com] <br><b>Sent:</b> Monday, January 31, 2011 9:17 AM<br><b>To:</b> freeswitch<br><b>Subject:</b> Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true.</span></p></div></div><p class="ecxMsoNormal"> </p><p class="ecxMsoNormal"><span style="font-size: 10pt; font-family: 'Tahoma','sans-serif';">Yes, I had it set up to iLBC@30i. It's not my cell phone (configured to ilbc, ptime=30 and mode=30), because when I call my freeswitch voicemail number, the sound is fine. I suspect it is something on the voip.ms <-> freeswitch leg because I created a sample ringback (8khz, mono, 16bit) wave file and directed my dialplan to it, but when I call from my home number to my cell, instead of hearing the ringer, I get choppy squeal.<br><br>?<br><br>> Date: Mon, 31 Jan 2011 10:40:10 -0600<br>> From: <a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a><br>> To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true.<br>> <br>> Many things have problems doing iLBC right.<br>> I recommend you define it in your configs as iLBC@30i or it will try<br>> using the 20ms version which is not compatible with many other<br>> platforms. Also make sure you are on the latest version of FS since<br>> we have tweaked iLBC behavior to compensate for problems like this.<br>> <br>> <br>> <br>> On Sun, Jan 30, 2011 at 10:40 PM, Marcin Wojtowicz<br>> <<a href="mailto:marcin321@hotmail.com">marcin321@hotmail.com</a>> wrote:<br>> > OK, so I gave up on bypass media, but now I have another problem. This time<br>> > I set up freeswitch to communicate with voip.ms using PCMU codec (configured<br>> > in my external profile), and use iLBC on my phone (codec configured in my<br>> > internal profile, where the phone registers). When I call my mobile it<br>> > rings, but when I pick up all I hear is a high pitched squeal. Am I missing<br>> > something here?<br>> ><br>> >> Date: Sun, 30 Jan 2011 16:34:09 -0600<br>> >> From: <a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a><br>> >> To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>> >> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when<br>> >> bypass_media is true.<br>> >><br>> >> Just do not use bypass media.<br>> >> That is all you can do in that situation.<br>> >><br>> >><br>> >> On Sun, Jan 30, 2011 at 3:44 PM, Marcin Wojtowicz <<a href="mailto:marcin321@hotmail.com">marcin321@hotmail.com</a>><br>> >> wrote:<br>> >> > I just want to add that I enabled STUN on my cell so now the SDP message<br>> >> > in<br>> >> > the INVITE to voip.ms contains the public IP of my phone, but it still<br>> >> > doesn't work.<br>> >> ><br>> >> > ________________________________<br>> >> > From: <a href="mailto:marcin321@hotmail.com">marcin321@hotmail.com</a><br>> >> > To: <a href="mailto:freeswitch-users@lists.freeswitch.org">freeswitch-users@lists.freeswitch.org</a><br>> >> > Date: Fri, 28 Jan 2011 19:54:19 -0500<br>> >> > Subject: [Freeswitch-users] Outbound only calls don't connect when<br>> >> > bypass_media is true.<br>> >> ><br>> >> > Hello,<br>> >> ><br>> >> > I'm a new user of freeswitch, so please bear with me. I have the<br>> >> > following setup:<br>> >> > voip.ms <- SIP over UDP -> my desktop running freeswitch <- SIP over TCP<br>> >> > -><br>> >> > my nokia cellphone on AT&T wireless. This setup is intended to conserve<br>> >> > the<br>> >> > battery usage.<br>> >> > I've managed to make everything work well when I'm calling in over any<br>> >> > phone<br>> >> > to my cell phone, and freeswitch is enabled to work in bypass_media =<br>> >> > true,<br>> >> > even though by cell is behind NAT on at&t's network. Things break when I<br>> >> > pick up my cell and try to call my home phone (or any phone for that<br>> >> > matter). This is the relevant snippet from my dialplan:<br>> >> > <extension name="outbound"><br>> >> > <condition field="destination_number"<br>> >> > expression="^1?([2-9]\d{2}[2-9]\d{6})$"><br>> >> > <!--<action application="set" data="bypass_media=true"/>--><br>> >> > <action application="bridge" data="sofia/gateway/voip.ms/1$1"/><br>> >> > </condition><br>> >> > </extension><br>> >> ><br>> >> > Like shown above, my call will go to my home phone. When I uncomment the<br>> >> > bypass_media tag, my call will not connect. Here are the siptraces<br>> >> > I replaced my real home phone number in the with "MYPHONE".<br>> >> ><br>> >> > recv 940 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.406250:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > INVITE <a target="_blank">sip:MYPHONE@192.168.1.100;transport=TCP</a> SIP/2.0<br>> >> > Via: SIP/2.0/TCP<br>> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport<br>> >> > From: <<a target="_blank">sip:1001@192.168.1.100</a>>;tag=eg6idg0knphc729fu7sj<br>> >> > To: <<a target="_blank">sip:MYPHONE@192.168.1.100</a>><br>> >> > Contact: <<a target="_blank">sip:M9jdt73ig0oOJSbt6Uyy@10.153.174.6:5060;transport=TCP</a>><br>> >> > Supported: 100rel,timer<br>> >> > CSeq: 5244503 INVITE<br>> >> > Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn<br>> >> > Allow:<br>> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE<br>> >> > User-Agent: S60 RM-624 v 20.2.042 (en)<br>> >> > Expires: 120<br>> >> > Privacy: None<br>> >> > Session-Expires: 1800<br>> >> > Max-Forwards: 70<br>> >> > Content-Type: application/sdp<br>> >> > Accept-Language: en<br>> >> > Content-Length: 292<br>> >> ><br>> >> > v=0<br>> >> > o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6<br>> >> > s=-<br>> >> > c=IN IP4 10.153.174.6<br>> >> > t=0 0<br>> >> > m=audio 49152 RTP/AVP 18 97 98<br>> >> > a=sendrecv<br>> >> > a=rtpmap:18 G729/8000<br>> >> > a=ptime:20<br>> >> > a=maxptime:40<br>> >> > a=fmtp:18 annexb=no<br>> >> > a=rtpmap:97 iLBC/8000<br>> >> > a=rtpmap:98 telephone-event/8000<br>> >> > a=fmtp:98 0-15<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > SIP/2.0 100 Trying<br>> >> > Via: SIP/2.0/TCP<br>> >> ><br>> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180<br>> >> > From: <<a target="_blank">sip:1001@192.168.1.100</a>>;tag=eg6idg0knphc729fu7sj<br>> >> > To: <<a target="_blank">sip:MYPHONE@192.168.1.100</a>><br>> >> > Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn<br>> >> > CSeq: 5244503 INVITE<br>> >> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13<br>> >> > 18-04-05<br>> >> > -0600<br>> >> > Content-Length: 0<br>> >> ><br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > SIP/2.0 407 Proxy Authentication Required<br>> >> > Via: SIP/2.0/TCP<br>> >> ><br>> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180<br>> >> > From: <<a target="_blank">sip:1001@192.168.1.100</a>>;tag=eg6idg0knphc729fu7sj2011-01-28<br>> >> > 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE)<br>> >> > on<br>> >> > sofia profile 'internal' for [MYPHONE@192.168.1.100] from ip<br>> >> > 32.136.78.180<br>> >> ><br>> >> > To: <<a target="_blank">sip:MYPHONE@192.168.1.100</a>>;tag=FQy5v5emcyt1m<br>> >> > Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn<br>> >> > CSeq: 5244503 INVITE<br>> >> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13<br>> >> > 18-04-05<br>> >> > -0600<br>> >> > Accept: application/sdp<br>> >> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,<br>> >> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE<br>> >> > Supported: timer, precondition, path, replaces<br>> >> > Allow-Events: talk, hold, presence, dialog, line-seize, call-info,<br>> >> > sla,<br>> >> > include-session-description, presence.winfo, message-summary, refer<br>> >> > Proxy-Authenticate: Digest realm="192.168.1.100",<br>> >> > nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28", algorithm=MD5, qop="auth"<br>> >> > Content-Length: 0<br>> >> ><br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > ACK <a target="_blank">sip:MYPHONE@192.168.1.100;transport=TCP</a> SIP/2.0<br>> >> > Via: SIP/2.0/TCP<br>> >> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport<br>> >> > From: <<a target="_blank">sip:1001@192.168.1.100</a>>;tag=eg6idg0knphc729fu7sj<br>> >> > To: <<a target="_blank">sip:MYPHONE@192.168.1.100</a>>;tag=FQy5v5emcyt1m<br>> >> > Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn<br>> >> > CSeq: 5244503 ACK<br>> >> > Supported: sec-agree<br>> >> > Max-Forwards: 70<br>> >> > Content-Length: 0<br>> >> ><br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > INVITE <a target="_blank">sip:MYPHONE@192.168.1.100;transport=TCP</a> SIP/2.0<br>> >> > Via: SIP/2.0/TCP<br>> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport<br>> >> > From: <<a target="_blank">sip:1001@192.168.1.100</a>>;tag=eg6idg0knphc729fu7sj<br>> >> > To: <<a target="_blank">sip:MYPHONE@192.168.1.100</a>><br>> >> > Contact: <<a target="_blank">sip:M9jdt73ig0oOJSbt6Uyy@10.153.174.6:5060;transport=TCP</a>><br>> >> > Supported: 100rel,timer<br>> >> > CSeq: 5244504 INVITE<br>> >> > Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn<br>> >> > Allow:<br>> >> > UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE<br>> >> > User-Agent: S60 RM-624 v 20.2.042 (en)<br>> >> > Expires: 120<br>> >> > Privacy: None<br>> >> > Session-Expires: 1800<br>> >> > Max-Forwards: 70<br>> >> > Proxy-Authorization: Digest<br>> >> ><br>> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="<a target="_blank">sip:MYPHONE@192.168.1.100;transport=TCP</a>",response="6c16edff1f978e58fadf6fb464ab8913"<br>> >> > Content-Type: application/sdp<br>> >> > Accept-Language: en<br>> >> > Content-Length: 292<br>> >> ><br>> >> > v=0<br>> >> > o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6<br>> >> > s=-<br>> >> > c=IN IP4 10.153.174.6<br>> >> > t=0 0<br>> >> > m=audio 49152 RTP/AVP 18 97 98<br>> >> > a=sendrecv<br>> >> > a=rtpmap:18 G729/8000<br>> >> > a=ptime:20<br>> >> > a=maxptime:40<br>> >> > a=fmtp:18 annexb=no<br>> >> > a=rtpmap:97 iLBC/8000<br>> >> > a=rtpmap:98 telephone-event/8000<br>> >> > a=fmtp:98 0-15<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > SIP/2.0 100 Trying<br>> >> > Via: SIP/2.0/TCP<br>> >> ><br>> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180<br>> >> > From: <<a target="_blank">sip:1001@192.168.1.100</a>>;tag=eg6idg0knphc729fu7sj<br>> >> > To: <<a target="_blank">sip:MYPHONE@192.168.1.100</a>><br>> >> > Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn<br>> >> > CSeq: 5244504 INVITE<br>> >> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13<br>> >> > 18-04-05<br>> >> > -0600<br>> >> > Content-Length: 0<br>> >> ><br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel<br>> >> > <a href="mailto:sofia/internal/1001@192.168.1.100">sofia/internal/1001@192.168.1.100</a> [e5841001-04bd-4e16-9519-64ff2c7a8c2f]<br>> >> > 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing 1001<br>> >> > <1001>->MYPHONE in context default<br>> >> > 2011-01-28 16:15:59.437500 [NOTICE] switch_channel.c:808 New Channel<br>> >> > sofia/external/1MYPHONE [60940227-9ae0-4c0c-abdb-f3988852fae0]<br>> >> > send 1141 bytes to udp/[74.63.41.218]:5060 at 21:15:59.453125:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > INVITE <a target="_blank">sip:1MYPHONE@newyork.voip.ms</a> SIP/2.0<br>> >> > Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS<br>> >> > Max-Forwards: 69<br>> >> > From: "Extension 1001"<br>> >> > <<a target="_blank">sip:121628@newyork.voip.ms;transport=udp</a>>;tag=Ny7H8Nt8eSy1S<br>> >> > To: <<a target="_blank">sip:1MYPHONE@newyork.voip.ms</a>><br>> >> > Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a<br>> >> > CSeq: 7788615 INVITE<br>> >> > Contact: <<a target="_blank">sip:gw+voip.ms@69.125.20.15:5080;transport=udp;gw=voip.ms</a>><br>> >> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13<br>> >> > 18-04-05<br>> >> > -0600<br>> >> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,<br>> >> > REGISTER, REFER, NOTIFY<br>> >> > Supported: timer, precondition, path, replaces<br>> >> > Allow-Events: talk, hold, refer<br>> >> > Content-Type: application/sdp<br>> >> > Content-Disposition: session<br>> >> > Content-Length: 280<br>> >> > X-FS-Support: update_display<br>> >> > Remote-Party-ID: "Extension 1001"<br>> >> > <<a target="_blank">sip:1001@69.125.20.15</a>>;party=calling;screen=yes;privacy=off<br>> >> ><br>> >> > v=0<br>> >> > o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6<br>> >> > s=-<br>> >> > c=IN IP4 10.153.174.6<br>> >> > t=0 0<br>> >> > m=audio 49152 RTP/AVP 18 97 98<br>> >> > a=rtpmap:18 G729/8000<br>> >> > a=fmtp:18 annexb=no<br>> >> > a=rtpmap:97 iLBC/8000<br>> >> > a=rtpmap:98 telephone-event/8000<br>> >> > a=fmtp:98 0-15<br>> >> > a=ptime:20<br>> >> > a=maxptime:40<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > SIP/2.0 407 Proxy Authentication Required<br>> >> > Via: SIP/2.0/UDP<br>> >> ><br>> >> > 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080<br>> >> > From: "Extension 1001"<br>> >> > <<a target="_blank">sip:121628@newyork.voip.ms;transport=udp</a>>;tag=Ny7H8Nt8eSy1S<br>> >> > To: <<a target="_blank">sip:1MYPHONE@newyork.voip.ms</a>>;tag=as7e7ea843<br>> >> > Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a<br>> >> > CSeq: 7788615 INVITE<br>> >> > User-Agent: VoIPMS/SERAST<br>> >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>> >> > Supported: replaces<br>> >> > Proxy-Authenticate: Digest algorithm=MD5, realm="newyork.voip.ms",<br>> >> > nonce="2d534dd6"<br>> >> > Content-Length: 0<br>> >> ><br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > ACK <a target="_blank">sip:1MYPHONE@newyork.voip.ms</a> SIP/2.0<br>> >> > Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKNp4ZFeKSD43tS<br>> >> > Max-Forwards: 69<br>> >> > From: "Extension 1001"<br>> >> > <<a target="_blank">sip:121628@newyork.voip.ms;transport=udp</a>>;tag=Ny7H8Nt8eSy1S<br>> >> > To: <<a target="_blank">sip:1MYPHONE@newyork.voip.ms</a>>;tag=as7e7ea843<br>> >> > Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a<br>> >> > CSeq: 7788615 ACK<br>> >> > Content-Length: 0<br>> >> ><br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > INVITE <a target="_blank">sip:1MYPHONE@newyork.voip.ms</a> SIP/2.0<br>> >> > Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN<br>> >> > Max-Forwards: 69<br>> >> > From: "Extension 1001"<br>> >> > <<a target="_blank">sip:121628@newyork.voip.ms;transport=udp</a>>;tag=Ny7H8Nt8eSy1S<br>> >> > To: <<a target="_blank">sip:1MYPHONE@newyork.voip.ms</a>><br>> >> > Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a<br>> >> > CSeq: 7788616 INVITE<br>> >> > Contact: <<a target="_blank">sip:gw+voip.ms@69.125.20.15:5080;transport=udp;gw=voip.ms</a>><br>> >> > Expires: 300<br>> >> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13<br>> >> > 18-04-05<br>> >> > -0600<br>> >> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,<br>> >> > REGISTER, REFER, NOTIFY<br>> >> > Supported: timer, precondition, path, replaces<br>> >> > Allow-Events: talk, hold, refer<br>> >> > Proxy-Authorization: Digest username="121628",<br>> >> > realm="newyork.voip.ms",<br>> >> > nonce="2d534dd6", algorithm=MD5, uri="<a target="_blank">sip:1MYPHONE@newyork.voip.ms</a>",<br>> >> > response="16f3301efae13df926da7550f709d28a"<br>> >> > Content-Type: application/sdp<br>> >> > Content-Disposition: session<br>> >> > Content-Length: 280<br>> >> > X-FS-Support: update_display<br>> >> > Remote-Party-ID: "Extension 1001"<br>> >> > <<a target="_blank">sip:1001@69.125.20.15</a>>;party=calling;screen=yes;privacy=off<br>> >> ><br>> >> > v=0<br>> >> > o=1001 63464487340299625 63464487340299625 IN IP4 10.153.174.6<br>> >> > s=-<br>> >> > c=IN IP4 10.153.174.6<br>> >> > t=0 0<br>> >> > m=audio 49152 RTP/AVP 18 97 98<br>> >> > a=rtpmap:18 G729/8000<br>> >> > a=fmtp:18 annexb=no<br>> >> > a=rtpmap:97 iLBC/8000<br>> >> > a=rtpmap:98 telephone-event/8000<br>> >> > a=fmtp:98 0-15<br>> >> > a=ptime:20<br>> >> > a=maxptime:40<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > SIP/2.0 100 Trying<br>> >> > Via: SIP/2.0/UDP<br>> >> ><br>> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080<br>> >> > From: "Extension 1001"<br>> >> > <<a target="_blank">sip:121628@newyork.voip.ms;transport=udp</a>>;tag=Ny7H8Nt8eSy1S<br>> >> > To: <<a target="_blank">sip:1MYPHONE@newyork.voip.ms</a>><br>> >> > Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a<br>> >> > CSeq: 7788616 INVITE<br>> >> > User-Agent: VoIPMS/SERAST<br>> >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>> >> > Supported: replaces<br>> >> > Contact: <<a target="_blank">sip:1MYPHONE@74.63.41.218</a>><br>> >> > Content-Length: 0<br>> >> ><br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > SIP/2.0 503 Service Unavailable<br>> >> > Via: SIP/2.0/UDP<br>> >> ><br>> >> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080<br>> >> > From: "Extension 1001"<br>> >> > <<a target="_blank">sip:121628@newyork.voip.ms;transport=udp</a>>;tag=Ny7H8Nt8eSy1S<br>> >> > To: <<a target="_blank">sip:1MYPHONE@newyork.voip.ms</a>>;tag=as632cb7d9<br>> >> > Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a<br>> >> > CSeq: 7788616 INVITE<br>> >> > User-Agent: VoIPMS/SERAST<br>> >> > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>> >> > Supported: replaces<br>> >> > Contact: <<a target="_blank">sip:1MYPHONE@74.63.41.218</a>><br>> >> > Content-Length: 0<br>> >> ><br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > ACK <a target="_blank">sip:1MYPHONE@newyork.voip.ms</a> SIP/2.0<br>> >> > Via: SIP/2.0/UDP 69.125.20.15:5080;rport;branch=z9hG4bKpZXrH93vaDtDN<br>> >> > Max-Forwards: 69<br>> >> > From: "Extension 1001"<br>> >> > <<a target="_blank">sip:121628@newyork.voip.ms;transport=udp</a>>;tag=Ny7H8Nt8eSy1S<br>> >> > To: <<a target="_blank">sip:1MYPHONE@newyork.voip.ms</a>>;tag=as632cb7d9<br>> >> > Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a<br>> >> > CSeq: 7788616 ACK<br>> >> > Content-Length: 0<br>> >> ><br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate Failed.<br>> >> > Cause: NO_ANSWER<br>> >> > 2011-01-28 16:15:59.562500 [NOTICE] sofia.c:5286 Hangup<br>> >> > sofia/external/1MYPHONE [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]<br>> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:189<br>> >> > <a href="mailto:sofia/internal/1001@192.168.1.100">sofia/internal/1001@192.168.1.100</a> has executed the last dialplan<br>> >> > instruction, hanging up.<br>> >> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191<br>> >> > Hangup<br>> >> > <a href="mailto:sofia/internal/1001@192.168.1.100">sofia/internal/1001@192.168.1.100</a> [CS_EXECUTE] [NORMAL_CLEARING]<br>> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1306 Session 2<br>> >> > (sofia/external/1MYPHONE) Ended<br>> >> > 2011-01-28 16:15:59.578125 [NOTICE] switch_core_session.c:1308 Close<br>> >> > Channel<br>> >> > sofia/external/1MYPHONE [CS_DESTROY]<br>> >> > send 887 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.593750:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > SIP/2.0 503 Service Unavailable<br>> >> > Via: SIP/2.0/TCP<br>> >> ><br>> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180<br>> >> > From: <<a target="_blank">sip:1001@192.168.1.100</a>>;tag=eg6idg0knphc729fu7sj<br>> >> > To: <<a target="_blank">sip:MYPHONE@192.168.1.100</a>>;tag=g0Qyy0ZQ96gmg<br>> >> > Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn<br>> >> > CSeq: 5244504 INVITE<br>> >> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13<br>> >> > 18-04-05<br>> >> > -0600<br>> >> > Accept: application/sdp<br>> >> > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,<br>> >> > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE<br>> >> > Supported: timer, precondition, path, replaces<br>> >> > Allow-Events: talk, hold, presence, dialog, line-seize, call-info,<br>> >> > sla,<br>> >> > include-session-description, presence.winfo, message-summary, refer<br>> >> > Reason: Q.850;cause=16;text="NORMAL_CLEARING"<br>> >> > 2011-01-28 16:15:59.593750 [NOTICE] switch_core_session.c:1306 Session 1<br>> >> > (<a href="mailto:sofia/internal/1001@192.168.1.100">sofia/internal/1001@192.168.1.100</a>) Ended<br>> >> > Content-Length: 02011-01-28 16:15:59.593750 [NOTICE]<br>> >> > switch_core_session.c:1308 Close Channel<br>> >> > <a href="mailto:sofia/internal/1001@192.168.1.100">sofia/internal/1001@192.168.1.100</a><br>> >> > [CS_DESTROY]<br>> >> ><br>> >> > Remote-Party-ID: "MYPHONE"<br>> >> > <<a target="_blank">sip:MYPHONE@192.168.1.100</a>>;party=calling;privacy=off;screen=no<br>> >> ><br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:<br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> > ACK <a target="_blank">sip:MYPHONE@192.168.1.100;transport=TCP</a> SIP/2.0<br>> >> > Via: SIP/2.0/TCP<br>> >> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport<br>> >> > From: <<a target="_blank">sip:1001@192.168.1.100</a>>;tag=eg6idg0knphc729fu7sj<br>> >> > To: <<a target="_blank">sip:MYPHONE@192.168.1.100</a>>;tag=g0Qyy0ZQ96gmg<br>> >> > Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn<br>> >> > CSeq: 5244504 ACK<br>> >> > Supported: sec-agree<br>> >> > Max-Forwards: 70<br>> >> > Proxy-Authorization: Digest<br>> >> ><br>> >> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="<a target="_blank">sip:MYPHONE@192.168.1.100;transport=TCP</a>",response="6c16edff1f978e58fadf6fb464ab8913"<br>> >> > Content-Length: 0<br>> >> ><br>> >> ><br>> >> > ------------------------------------------------------------------------<br>> >> ><br>> >> > Thank you in advance.<br>> >> ><br>> >> > _______________________________________________ FreeSWITCH-users mailing<br>> >> > list <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>> >> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> >> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>> >> > _______________________________________________<br>> >> > FreeSWITCH-users mailing list<br>> >> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>> >> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> >> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>> >> ><br>> >> ><br>> >><br>> >><br>> >><br>> >> --<br>> >> Anthony Minessale II<br>> >><br>> >> FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>> >> ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>> >> Twitter: <a href="http://twitter.com/FreeSWITCH_wire" target="_blank">http://twitter.com/FreeSWITCH_wire</a><br>> >><br>> >> AIM: anthm<br>> >> <a target="_blank">MSN:anthony_minessale@hotmail.com</a><br>> >> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<br>> >> IRC: irc.freenode.net #freeswitch<br>> >><br>> >> FreeSWITCH Developer Conference<br>> >> <a target="_blank">sip:888@conference.freeswitch.org</a><br>> >> googletalk:conf+888@conference.freeswitch.org<br>> >> pstn:+19193869900<br>> >><br>> >> _______________________________________________<br>> >> FreeSWITCH-users mailing list<br>> >> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>> >> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> >> <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>> ><br>> > _______________________________________________<br>> > FreeSWITCH-users mailing list<br>> > <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>> > <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> > <a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>> ><br>> ><br>> <br>> <br>> <br>> -- <br>> Anthony Minessale II<br>> <br>> FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>> ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>> Twitter: <a href="http://twitter.com/FreeSWITCH_wire" target="_blank">http://twitter.com/FreeSWITCH_wire</a><br>> <br>> AIM: anthm<br>> <a target="_blank">MSN:anthony_minessale@hotmail.com</a><br>> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<br>> IRC: irc.freenode.net #freeswitch<br>> <br>> FreeSWITCH Developer Conference<br>> <a target="_blank">sip:888@conference.freeswitch.org</a><br>> googletalk:conf+888@conference.freeswitch.org<br>> pstn:+19193869900<br>> <br>> _______________________________________________<br>> FreeSWITCH-users mailing list<br>> <a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>> <a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" 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