<html>
<head>
<style><!--
.hmmessage P
{
margin:0px;
padding:0px
}
body.hmmessage
{
font-size: 10pt;
font-family:Tahoma
}
--></style>
</head>
<body class='hmmessage'>
OK, so I gave up on bypass media, but now I have another problem. This time I set up freeswitch to communicate with voip.ms using PCMU codec (configured in my external profile), and use iLBC on my phone (codec configured in my internal profile, where the phone registers). When I call my mobile it rings, but when I pick up all I hear is a high pitched squeal. Am I missing something here?<br><br>> Date: Sun, 30 Jan 2011 16:34:09 -0600<br>> From: anthony.minessale@gmail.com<br>> To: freeswitch-users@lists.freeswitch.org<br>> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when 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 <marcin321@hotmail.com> wrote:<br>> > I just want to add that I enabled STUN on my cell so now the SDP message 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: marcin321@hotmail.com<br>> > To: freeswitch-users@lists.freeswitch.org<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>> > my nokia cellphone on AT&T wireless. This setup is intended to conserve the<br>> > battery usage.<br>> > I've managed to make everything work well when I'm calling in over any phone<br>> > to my cell phone, and freeswitch is enabled to work in bypass_media = 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>> > INVITE sip:MYPHONE@192.168.1.100;transport=TCP SIP/2.0<br>> > Via: SIP/2.0/TCP<br>> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport<br>> > From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj<br>> > To: <sip:MYPHONE@192.168.1.100><br>> > Contact: <sip:M9jdt73ig0oOJSbt6Uyy@10.153.174.6:5060;transport=TCP><br>> > Supported: 100rel,timer<br>> > CSeq: 5244503 INVITE<br>> > Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn<br>> > Allow: 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>> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:<br>> > ------------------------------------------------------------------------<br>> > SIP/2.0 100 Trying<br>> > Via: SIP/2.0/TCP<br>> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180<br>> > From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj<br>> > To: <sip:MYPHONE@192.168.1.100><br>> > Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn<br>> > CSeq: 5244503 INVITE<br>> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05<br>> > -0600<br>> > Content-Length: 0<br>> ><br>> > ------------------------------------------------------------------------<br>> > send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:<br>> > ------------------------------------------------------------------------<br>> > SIP/2.0 407 Proxy Authentication Required<br>> > Via: SIP/2.0/TCP<br>> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport=51328;received=32.136.78.180<br>> > From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj2011-01-28<br>> > 16:15:58.406250 [WARNING] sofia_reg.c:1241 SIP auth challenge (INVITE) on<br>> > sofia profile 'internal' for [MYPHONE@192.168.1.100] from ip 32.136.78.180<br>> ><br>> > To: <sip:MYPHONE@192.168.1.100>;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 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, 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>> > recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:<br>> > ------------------------------------------------------------------------<br>> > ACK sip:MYPHONE@192.168.1.100;transport=TCP SIP/2.0<br>> > Via: SIP/2.0/TCP<br>> > 10.153.174.6:5060;branch=z9hG4bK212idg50p1hc6vjdu7steua;rport<br>> > From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj<br>> > To: <sip:MYPHONE@192.168.1.100>;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>> > recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:<br>> > ------------------------------------------------------------------------<br>> > INVITE sip:MYPHONE@192.168.1.100;transport=TCP SIP/2.0<br>> > Via: SIP/2.0/TCP<br>> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport<br>> > From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj<br>> > To: <sip:MYPHONE@192.168.1.100><br>> > Contact: <sip:M9jdt73ig0oOJSbt6Uyy@10.153.174.6:5060;transport=TCP><br>> > Supported: 100rel,timer<br>> > CSeq: 5244504 INVITE<br>> > Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn<br>> > Allow: 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>> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE@192.168.1.100;transport=TCP",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>> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:<br>> > ------------------------------------------------------------------------<br>> > SIP/2.0 100 Trying<br>> > Via: SIP/2.0/TCP<br>> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180<br>> > From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj<br>> > To: <sip:MYPHONE@192.168.1.100><br>> > Call-ID: bNQCM4UsoIde5f7V6_8z1_Cqd4Q5xn<br>> > CSeq: 5244504 INVITE<br>> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 18-04-05<br>> > -0600<br>> > Content-Length: 0<br>> ><br>> > ------------------------------------------------------------------------<br>> > 2011-01-28 16:15:59.406250 [NOTICE] switch_channel.c:808 New Channel<br>> > sofia/internal/1001@192.168.1.100 [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>> > INVITE sip:1MYPHONE@newyork.voip.ms 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>> > <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S<br>> > To: <sip:1MYPHONE@newyork.voip.ms><br>> > Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a<br>> > CSeq: 7788615 INVITE<br>> > Contact: <sip:gw+voip.ms@69.125.20.15:5080;transport=udp;gw=voip.ms><br>> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 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>> > <sip:1001@69.125.20.15>;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>> > recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:<br>> > ------------------------------------------------------------------------<br>> > SIP/2.0 407 Proxy Authentication Required<br>> > Via: SIP/2.0/UDP<br>> > 69.125.20.15:5080;branch=z9hG4bKNp4ZFeKSD43tS;received=69.125.20.15;rport=5080<br>> > From: "Extension 1001"<br>> > <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S<br>> > To: <sip:1MYPHONE@newyork.voip.ms>;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>> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:<br>> > ------------------------------------------------------------------------<br>> > ACK sip:1MYPHONE@newyork.voip.ms 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>> > <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S<br>> > To: <sip:1MYPHONE@newyork.voip.ms>;tag=as7e7ea843<br>> > Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a<br>> > CSeq: 7788615 ACK<br>> > Content-Length: 0<br>> ><br>> > ------------------------------------------------------------------------<br>> > send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:<br>> > ------------------------------------------------------------------------<br>> > INVITE sip:1MYPHONE@newyork.voip.ms 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>> > <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S<br>> > To: <sip:1MYPHONE@newyork.voip.ms><br>> > Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a<br>> > CSeq: 7788616 INVITE<br>> > Contact: <sip:gw+voip.ms@69.125.20.15:5080;transport=udp;gw=voip.ms><br>> > Expires: 300<br>> > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-7a04104 2011-01-13 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", realm="newyork.voip.ms",<br>> > nonce="2d534dd6", algorithm=MD5, uri="sip:1MYPHONE@newyork.voip.ms",<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>> > <sip:1001@69.125.20.15>;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>> > recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:<br>> > ------------------------------------------------------------------------<br>> > SIP/2.0 100 Trying<br>> > Via: SIP/2.0/UDP<br>> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080<br>> > From: "Extension 1001"<br>> > <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S<br>> > To: <sip:1MYPHONE@newyork.voip.ms><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: <sip:1MYPHONE@74.63.41.218><br>> > Content-Length: 0<br>> ><br>> > ------------------------------------------------------------------------<br>> > recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:<br>> > ------------------------------------------------------------------------<br>> > SIP/2.0 503 Service Unavailable<br>> > Via: SIP/2.0/UDP<br>> > 69.125.20.15:5080;branch=z9hG4bKpZXrH93vaDtDN;received=69.125.20.15;rport=5080<br>> > From: "Extension 1001"<br>> > <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S<br>> > To: <sip:1MYPHONE@newyork.voip.ms>;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: <sip:1MYPHONE@74.63.41.218><br>> > Content-Length: 0<br>> ><br>> > ------------------------------------------------------------------------<br>> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:<br>> > ------------------------------------------------------------------------<br>> > ACK sip:1MYPHONE@newyork.voip.ms 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>> > <sip:121628@newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S<br>> > To: <sip:1MYPHONE@newyork.voip.ms>;tag=as632cb7d9<br>> > Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a<br>> > CSeq: 7788616 ACK<br>> > Content-Length: 0<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>> > sofia/internal/1001@192.168.1.100 has executed the last dialplan<br>> > instruction, hanging up.<br>> > 2011-01-28 16:15:59.562500 [NOTICE] switch_core_state_machine.c:191 Hangup<br>> > sofia/internal/1001@192.168.1.100 [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 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>> > SIP/2.0 503 Service Unavailable<br>> > Via: SIP/2.0/TCP<br>> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport=51328;received=32.136.78.180<br>> > From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj<br>> > To: <sip:MYPHONE@192.168.1.100>;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 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, 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>> > (sofia/internal/1001@192.168.1.100) Ended<br>> > Content-Length: 02011-01-28 16:15:59.593750 [NOTICE]<br>> > switch_core_session.c:1308 Close Channel sofia/internal/1001@192.168.1.100<br>> > [CS_DESTROY]<br>> ><br>> > Remote-Party-ID: "MYPHONE"<br>> > <sip:MYPHONE@192.168.1.100>;party=calling;privacy=off;screen=no<br>> ><br>> > ------------------------------------------------------------------------<br>> > recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:<br>> > ------------------------------------------------------------------------<br>> > ACK sip:MYPHONE@192.168.1.100;transport=TCP SIP/2.0<br>> > Via: SIP/2.0/TCP<br>> > 10.153.174.6:5060;branch=z9hG4bKlelbaqjr084phj1q4r0cg9b;rport<br>> > From: <sip:1001@192.168.1.100>;tag=eg6idg0knphc729fu7sj<br>> > To: <sip:MYPHONE@192.168.1.100>;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>> > qop=auth,realm="192.168.1.100",nonce="8dc0f0de-04b7-44ba-9ef9-a3caa135ec28",algorithm=MD5,username="1001",cnonce="0a1e034e285d559db5d6ded3fb0ce4ee",nc=00000001,uri="sip:MYPHONE@192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"<br>> > Content-Length: 0<br>> ><br>> > ------------------------------------------------------------------------<br>> ><br>> > Thank you in advance.<br>> ><br>> > _______________________________________________ FreeSWITCH-users mailing<br>> > list FreeSWITCH-users@lists.freeswitch.org<br>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> > http://www.freeswitch.org<br>> > _______________________________________________<br>> > FreeSWITCH-users mailing list<br>> > FreeSWITCH-users@lists.freeswitch.org<br>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> > http://www.freeswitch.org<br>> ><br>> ><br>> <br>> <br>> <br>> -- <br>> Anthony Minessale II<br>> <br>> FreeSWITCH http://www.freeswitch.org/<br>> ClueCon http://www.cluecon.com/<br>> Twitter: http://twitter.com/FreeSWITCH_wire<br>> <br>> AIM: anthm<br>> MSN:anthony_minessale@hotmail.com<br>> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<br>> IRC: irc.freenode.net #freeswitch<br>> <br>> FreeSWITCH Developer Conference<br>> sip:888@conference.freeswitch.org<br>> googletalk:conf+888@conference.freeswitch.org<br>> pstn:+19193869900<br>> <br>> _______________________________________________<br>> FreeSWITCH-users mailing list<br>> FreeSWITCH-users@lists.freeswitch.org<br>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> http://www.freeswitch.org<br> </body>
</html>