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Here are the build errors, in order of appearance:<br>------ Build started: Project: libpcre, Configuration: Release Win32 ------<br> pcre_chartables.c<br>c1 : fatal error C1083: Cannot open source file: 'pcre_chartables.c': No such file or directory<br>(other files here compile ok)<br><br>------ Build started: Project: FreeSwitchCoreLib, Configuration: Release Win32 ------<br> Generating switch_version.h<br> (stuff here goes ok)<br> Generating Code...<br>LINK : fatal error LNK1181: cannot open input file 'E:\downloads\freeswitch-1.0.7\libs\win32\pcre\Win32\Release\libpcre.lib'<br><br>------ Build started: Project: mod_spidermonkey, Configuration: Release Win32 ------<br> mod_spidermonkey.c<br>LINK : fatal error LNK1181: cannot open input file 'E:\downloads\freeswitch-1.0.7\Win32\Release\FreeSwitchCore.lib'<br><br>FreeSwitchCore link error repeats many times for other projects, too many to list. It doesn't build.<br><br>------ Build started: Project: mod_managed, Configuration: Release_CLR Win32 ------<br> freeswitch_managed.cpp<br>freeswitch_managed.cpp : fatal error C1192: #using failed on 'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll'<br> 'The system cannot find the path specified.'<br> freeswitch_wrap.2010.cxx<br>freeswitch_wrap.2010.cxx : fatal error C1192: #using failed on 'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll'<br> 'The system cannot find the path specified.'<br> mod_managed.cpp<br>mod_managed.cpp : fatal error C1192: #using failed on 'E:\downloads\freeswitch-1.0.7\Win32\Debug\mod\FreeSWITCH.Managed.dll'<br> 'The system cannot find the path specified.'<br> Generating Code...<br><br><br>The code I downloaded was from latest.freeswitch.org, the tar.bz2 zip file. From the dependencies it looks like FreeSwitchCore depends on libpcre, so all this cascades from the missing pcre_chartables.c file. Can you send it to me?<br><br><br><br><br>> Date: Mon, 31 Jan 2011 14:00:10 -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>> a=fmtp:98 mode=30 is missing in the 200 ok from the phone.<br>> Also did you mention the revision you are on. I had indicated that the<br>> very latest code may have more tolerant ilbc codec code in it.<br>> http://latest.freeswitch.org<br>> <br>> <br>> On Mon, Jan 31, 2011 at 12:12 PM, Marcin Wojtowicz<br>> <marcin321@hotmail.com> wrote:<br>> > SDP looks ok to me, but there is one warning about ptime in iLBC below. I<br>> > don't see how a wrong codec can be selected because I narrowed down my<br>> > external profile inbound/outbound to PCMU only and my internal is iLBC@30i<br>> > only.<br>> ><br>> ><br>> > freeswitch@kuffel> recv 1206 bytes from udp/[74.63.41.218]:5060 at<br>> > 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#"<br>> > <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<br>> > -0600<br>> > Content-Length: 0<br>> ><br>> > ------------------------------------------------------------------------<br>> > 2011-01-31 12:59:32.187500 [NOTICE] switch_channel.c:808 New Channel<br>> > 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#<br>> > <MYPHONE#>->121628 in context public<br>> > 2011-01-31 12:59:32.203125 [NOTICE] switch_ivr.c:1606 Transfer<br>> > 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#<br>> > <MYPHONE#>->1001 in context default<br>> > 2011-01-31 12:59:32.234375 [NOTICE] switch_channel.c:808 New Channel<br>> > sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155:5060<br>> > [7230b9e8-37a7-4fc6-9b52-25740a6f7ca4]<br>> ><br>> ><br>> > 2011-01-31 12:59:32.265625 [WARNING] sofia_glue.c:213 Codec iLBC payload 98<br>> > added to sdp wanting ptime 30 but it's already 20 (PCMU:0:20), disabling<br>> > 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<br>> > -0600<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>> > Content-Type: application/sdp<br>> > Content-Disposition: session<br>> > Content-Length: 234<br>> > X-FS-Support: update_display<br>> > Remote-Party-ID: "MYPHONE#"<br>> > <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<br>> > 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<br>> > 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:<br>> > <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<br>> > sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155:5060 Update Callee ID<br>> > to "Outbound Call" <M9jdt73ig0oOJSbt6Uyy><br>> > 2011-01-31 12:59:42.296875 [NOTICE] sofia.c:4724 Ring-Ready<br>> > 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<br>> > media<br>> > 2011-01-31 12:59:42.343750 [NOTICE] mod_sofia.c:2252 Pre-Answer<br>> > 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:<br>> > <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<br>> > -0600<br>> > Accept: application/sdp<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: 247<br>> > Remote-Party-ID: "121628"<br>> > <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<br>> > 69.125.20.15;branch=z9hG4bK797eFQ6rgQKmQ;rport=5060;received=69.125.20.15<br>> > To:<br>> > <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<br>> > 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:<br>> > <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<br>> > [sofia/internal/sip:M9jdt73ig0oOJSbt6Uyy@10.208.245.155:5060] has been<br>> > 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:<br>> > <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<br>> > [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<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: 247<br>> > Remote-Party-ID: "Outbound Call"<br>> > <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:<br>> > <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#"<br>> > <sip:MYPHONE#@74.63.41.218>;privacy=off;screen=no<br>> > Content-Length: 0<br>> ><br>> > ________________________________<br>> > 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<br>> > bypass_media is true.<br>> ><br>> ><br>> ><br>> > Check the codecs in the SDP or try manual hardcoding the codecs presented<br>> > for both legs, we had a squeal problem going to a softphone that turned out<br>> > to be the BV32 codec was being selected instead of SPEEX16.<br>> ><br>> ><br>> ><br>> > Robert<br>> ><br>> ><br>> ><br>> > From: Marcin Wojtowicz [mailto:marcin321@hotmail.com]<br>> > Sent: Monday, January 31, 2011 9:17 AM<br>> > To: freeswitch<br>> > Subject: Re: [Freeswitch-users] Outbound only calls don't connect when<br>> > bypass_media is true.<br>> ><br>> ><br>> ><br>> > Yes, I had it set up to iLBC@30i. It's not my cell phone (configured to<br>> > ilbc, ptime=30 and mode=30), because when I call my freeswitch voicemail<br>> > number, the sound is fine. I suspect it is something on the voip.ms <-><br>> > freeswitch leg because I created a sample ringback (8khz, mono, 16bit) wave<br>> > file and directed my dialplan to it, but when I call from my home number to<br>> > 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: anthony.minessale@gmail.com<br>> >> To: freeswitch-users@lists.freeswitch.org<br>> >> Subject: Re: [Freeswitch-users] Outbound only calls don't connect when<br>> >> 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>> >> <marcin321@hotmail.com> wrote:<br>> >> > OK, so I gave up on bypass media, but now I have another problem. This<br>> >> > time<br>> >> > I set up freeswitch to communicate with voip.ms using PCMU codec<br>> >> > (configured<br>> >> > in my external profile), and use iLBC on my phone (codec configured in<br>> >> > 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<br>> >> > missing<br>> >> > 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<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<br>> >> >> <marcin321@hotmail.com><br>> >> >> wrote:<br>> >> >> > I just want to add that I enabled STUN on my cell so now the SDP<br>> >> >> > message<br>> >> >> > in<br>> >> >> > the INVITE to voip.ms contains the public IP of my phone, but it<br>> >> >> > 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<br>> >> >> > TCP<br>> >> >> > -><br>> >> >> > my nokia cellphone on AT&T wireless. This setup is intended to<br>> >> >> > conserve<br>> >> >> > the<br>> >> >> > battery usage.<br>> >> >> > I've managed to make everything work well when I'm calling in over<br>> >> >> > 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<br>> >> >> > 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<br>> >> >> > 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>> >> >> > ------------------------------------------------------------------------<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:<br>> >> >> > <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:<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>> >> >> > ------------------------------------------------------------------------<br>> >> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:<br>> >> >> ><br>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > SIP/2.0 100 Trying<br>> >> >> > Via: SIP/2.0/TCP<br>> >> >> ><br>> >> >> ><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<br>> >> >> > 18-04-05<br>> >> >> > -0600<br>> >> >> > Content-Length: 0<br>> >> >> ><br>> >> >> ><br>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > send 875 bytes to tcp/[32.136.78.180]:51328 at 21:15:58.406250:<br>> >> >> ><br>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > SIP/2.0 407 Proxy Authentication Required<br>> >> >> > Via: SIP/2.0/TCP<br>> >> >> ><br>> >> >> ><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<br>> >> >> > (INVITE)<br>> >> >> > on<br>> >> >> > sofia profile 'internal' for [MYPHONE@192.168.1.100] from ip<br>> >> >> > 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<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,<br>> >> >> > qop="auth"<br>> >> >> > Content-Length: 0<br>> >> >> ><br>> >> >> ><br>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > recv 368 bytes from tcp/[32.136.78.180]:51328 at 21:15:58.765625:<br>> >> >> ><br>> >> >> ><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>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > recv 1222 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.406250:<br>> >> >> ><br>> >> >> ><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:<br>> >> >> > <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:<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>> >> >> ><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>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > send 387 bytes to tcp/[32.136.78.180]:51328 at 21:15:59.406250:<br>> >> >> ><br>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > SIP/2.0 100 Trying<br>> >> >> > Via: SIP/2.0/TCP<br>> >> >> ><br>> >> >> ><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<br>> >> >> > 18-04-05<br>> >> >> > -0600<br>> >> >> > Content-Length: 0<br>> >> >> ><br>> >> >> ><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<br>> >> >> > [e5841001-04bd-4e16-9519-64ff2c7a8c2f]<br>> >> >> > 2011-01-28 16:15:59.421875 [INFO] mod_dialplan_xml.c:331 Processing<br>> >> >> > 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>> >> >> > ------------------------------------------------------------------------<br>> >> >> > INVITE sip:1MYPHONE@newyork.voip.ms SIP/2.0<br>> >> >> > Via: SIP/2.0/UDP<br>> >> >> > 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:<br>> >> >> > <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<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>> >> >> > <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>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > recv 570 bytes from udp/[74.63.41.218]:5060 at 21:15:59.468750:<br>> >> >> ><br>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > SIP/2.0 407 Proxy Authentication Required<br>> >> >> > Via: SIP/2.0/UDP<br>> >> >> ><br>> >> >> ><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>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:<br>> >> >> ><br>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > ACK sip:1MYPHONE@newyork.voip.ms SIP/2.0<br>> >> >> > Via: SIP/2.0/UDP<br>> >> >> > 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>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > send 1344 bytes to udp/[74.63.41.218]:5060 at 21:15:59.468750:<br>> >> >> ><br>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > INVITE sip:1MYPHONE@newyork.voip.ms SIP/2.0<br>> >> >> > Via: SIP/2.0/UDP<br>> >> >> > 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:<br>> >> >> > <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<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="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>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > recv 488 bytes from udp/[74.63.41.218]:5060 at 21:15:59.484375:<br>> >> >> ><br>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > SIP/2.0 100 Trying<br>> >> >> > Via: SIP/2.0/UDP<br>> >> >> ><br>> >> >> ><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>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > recv 516 bytes from udp/[74.63.41.218]:5060 at 21:15:59.562500:<br>> >> >> ><br>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > SIP/2.0 503 Service Unavailable<br>> >> >> > Via: SIP/2.0/UDP<br>> >> >> ><br>> >> >> ><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>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > send 359 bytes to udp/[74.63.41.218]:5060 at 21:15:59.562500:<br>> >> >> ><br>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > ACK sip:1MYPHONE@newyork.voip.ms SIP/2.0<br>> >> >> > Via: SIP/2.0/UDP<br>> >> >> > 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>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > 2011-01-28 16:15:59.562500 [INFO] mod_dptools.c:2612 Originate<br>> >> >> > 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<br>> >> >> > 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<br>> >> >> > 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>> >> >> > ------------------------------------------------------------------------<br>> >> >> > SIP/2.0 503 Service Unavailable<br>> >> >> > Via: SIP/2.0/TCP<br>> >> >> ><br>> >> >> ><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<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<br>> >> >> > 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<br>> >> >> > 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>> >> >> ><br>> >> >> > ------------------------------------------------------------------------<br>> >> >> > recv 650 bytes from tcp/[32.136.78.180]:51328 at 21:15:59.953125:<br>> >> >> ><br>> >> >> ><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>> >> >> ><br>> >> >> ><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>> >> >> > ------------------------------------------------------------------------<br>> >> >> ><br>> >> >> > Thank you in advance.<br>> >> >> ><br>> >> >> > _______________________________________________ FreeSWITCH-users<br>> >> >> > mailing<br>> >> >> > list FreeSWITCH-users@lists.freeswitch.org<br>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> >> >> ><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>> >> >> ><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>> >> >><br>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> >> >> http://www.freeswitch.org<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>> >> ><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>> ><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>
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