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