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

Mon Jan 31 21:12:30 MSK 2011

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
_______________________________________________
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