Re: [Freeswitch-users] Outbound only calls don't connect when bypass_media is true.

31 Jan 2011


      I tried building with visual C++ express 2010, but there were numerous fatal errors and it didn't finish.
...
Date: Mon, 31 Jan 2011 14:00:10 -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.
a=fmtp:98 mode=30 is missing in the 200 ok from the phone.
Also did you mention the revision you are on. I had indicated that the
very latest code may have more tolerant ilbc codec code in it.
http://latest.freeswitch.org
On Mon, Jan 31, 2011 at 12:12 PM, Marcin Wojtowicz
<marcin321@hotmail.com> wrote:
...
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
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--
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
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--
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ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale@hotmail.com
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Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
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Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm
MSN:anthony_minessale@hotmail.com
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