[Freeswitch-users] Outbound only calls don't connect when bypass_media is true.
Anthony Minessale
anthony.minessale at gmail.com
Mon Jan 31 23:00:10 MSK 2011
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 at 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 at 30i
> only.
>
>
> freeswitch at kuffel> recv 1206 bytes from udp/[74.63.41.218]:5060 at
> 17:59:32.187500:
> ------------------------------------------------------------------------
> INVITE sip:gw+voip.ms at 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 at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> Contact: <sip:MYPHONE#@74.63.41.218>
> Call-ID: 1298959c0099d50d177b6bae689fc028 at 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 at 69.125.20.15:5080;transport=udp;gw=voip.ms>
> Call-ID: 1298959c0099d50d177b6bae689fc028 at 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 at 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 at 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 at 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 at 32.140.14.196:46743>;transport=TCP
> Max-Forwards: 68
> From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> To: <sip:M9jdt73ig0oOJSbt6Uyy at 10.208.245.155:5060;transport=TCP>
> Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> CSeq: 7912322 INVITE
> Contact: <sip:mod_sofia at 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 at 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 at 10.208.245.155:5060;transport=TCP>
> From: "MYPHONE#" <sip:MYPHONE#@192.168.1.100>;tag=cF7Ure4ZUFjXa
> To:
> <sip:M9jdt73ig0oOJSbt6Uyy at 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 at 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 at 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 at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
> Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> CSeq: 102 INVITE
> Contact: <sip:gw+voip.ms at 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 at 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 at 10.208.245.155>;tag=p4rl1jbfvmnbvfs1d5rktoj2
> Contact: <sip:M9jdt73ig0oOJSbt6Uyy at 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 at 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 at 10.208.245.155:5060;transport=TCP>;tag=p4rl1jbfvmnbvfs1d5rktoj2
> Call-ID: b281a1fa-a806-122e-f799-c1188a708e17
> CSeq: 7912322 ACK
> Contact: <sip:mod_sofia at 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 at 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 at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
> Call-ID: 1298959c0099d50d177b6bae689fc028 at 74.63.41.218
> CSeq: 102 INVITE
> Contact: <sip:gw+voip.ms at 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 at 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 at 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 at 69.125.20.15:5080;transport=udp;gw=voip.ms>;tag=eK0X80BS0091S
> Contact: <sip:MYPHONE#@74.63.41.218>
> Call-ID: 1298959c0099d50d177b6bae689fc028 at 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 at teotech.com
> To: freeswitch-users at 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 at 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 at 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 at gmail.com
>> To: freeswitch-users at 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 at 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 at 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 at gmail.com
>> >> To: freeswitch-users at 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 at 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 at hotmail.com
>> >> > To: freeswitch-users at 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 at 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 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > To: <sip:MYPHONE at 192.168.1.100>
>> >> > Contact:
>> >> > <sip:M9jdt73ig0oOJSbt6Uyy at 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 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > To: <sip:MYPHONE at 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 at 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 at 192.168.1.100] from ip
>> >> > 32.136.78.180
>> >> >
>> >> > To: <sip:MYPHONE at 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 at 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 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > To: <sip:MYPHONE at 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 at 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 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > To: <sip:MYPHONE at 192.168.1.100>
>> >> > Contact:
>> >> > <sip:M9jdt73ig0oOJSbt6Uyy at 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 at 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 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > To: <sip:MYPHONE at 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 at 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 at 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 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > To: <sip:1MYPHONE at newyork.voip.ms>
>> >> > Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> >> > CSeq: 7788615 INVITE
>> >> > Contact:
>> >> > <sip:gw+voip.ms at 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 at 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 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > To: <sip:1MYPHONE at 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 at 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 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > To: <sip:1MYPHONE at 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 at 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 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > To: <sip:1MYPHONE at newyork.voip.ms>
>> >> > Call-ID: a4fa75b2-a5c6-122e-9b9c-c5a13034e45a
>> >> > CSeq: 7788616 INVITE
>> >> > Contact:
>> >> > <sip:gw+voip.ms at 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 at 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 at 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 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > To: <sip:1MYPHONE at 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 at 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 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > To: <sip:1MYPHONE at 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 at 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 at 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 at newyork.voip.ms;transport=udp>;tag=Ny7H8Nt8eSy1S
>> >> > To: <sip:1MYPHONE at 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 at 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 at 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 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > To: <sip:MYPHONE at 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 at 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 at 192.168.1.100
>> >> > [CS_DESTROY]
>> >> >
>> >> > Remote-Party-ID: "MYPHONE"
>> >> > <sip:MYPHONE at 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 at 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 at 192.168.1.100>;tag=eg6idg0knphc729fu7sj
>> >> > To: <sip:MYPHONE at 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 at 192.168.1.100;transport=TCP",response="6c16edff1f978e58fadf6fb464ab8913"
>> >> > Content-Length: 0
>> >> >
>> >> >
>> >> >
>> >> > ------------------------------------------------------------------------
>> >> >
>> >> > Thank you in advance.
>> >> >
>> >> > _______________________________________________ FreeSWITCH-users
>> >> > mailing
>> >> > list FreeSWITCH-users at 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 at 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 at hotmail.com
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org
>> >> googletalk:conf+888 at conference.freeswitch.org
>> >> pstn:+19193869900
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at 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 at 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 at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:+19193869900
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at 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 at 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 at 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 at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
googletalk:conf+888 at conference.freeswitch.org
pstn:+19193869900
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