[Freeswitch-users] Cannot complete outbound call to voipyourlife

[Mario]

When I dial out to voipyourlife I get a fast busy signal no matter what
number I dial. The messages and trace don't help me. Web searches on
freeswitch and viopyourlife/voipyourphone yielded no results. Thanks
very much!

sofia profile external siptrace on
2010-09-20 11:47:21.170623 [NOTICE] switch_channel.c:779 New Channel
sofia/internal/200 at 10.246.1.37 [2d5efef1-c8ee-4818-8563-937f62d04543]

2010-09-20 11:47:21.318116 [INFO] mod_dialplan_xml.c:331 Processing 200
<200>->814181234567 in context default

2010-09-20 11:47:21.328251 [NOTICE] switch_channel.c:779 New Channel
sofia/external/14181234567 at sip.voipyourlife.com
[7e8f01ae-112b-495a-b091-870f5304f51e]


2010-09-20 11:47:24.100398 [NOTICE] sofia.c:5051 Hangup
sofia/external/14181234567 at sip.voipyourlife.com [CS_CONSUME_MEDIA]
[NORMAL_TEMPORARY_FAILURE]
2010-09-20 11:47:24.100398 [INFO] mod_dptools.c:2393 Originate Failed.
Cause: NORMAL_TEMPORARY_FAILURE

2010-09-20 11:47:24.100398 [NOTICE] mod_dptools.c:2456 Hangup
sofia/internal/200 at 10.246.1.37 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE]

2010-09-20 11:47:24.100398 [NOTICE] switch_core_session.c:1228 Session 4
(sofia/external/14181234567 at sip.voipyourlife.com) Ended

2010-09-20 11:47:24.100398 [NOTICE] switch_core_session.c:1230 Close
Channel sofia/external/14181234567 at sip.voipyourlife.com [CS_DESTROY]

2010-09-20 11:47:24.212365 [NOTICE] switch_core_session.c:1228 Session 3
(sofia/internal/200 at 10.246.1.37) Ended

2010-09-20 11:47:24.212365 [NOTICE] switch_core_session.c:1230 Close
Channel sofia/internal/200 at 10.246.1.37 [CS_DESTROY]

recv 650 bytes from udp/[216.143.130.36]:5060 at 18:48:24.027473:


   ------------------------------------------------------------------------
   NOTIFY sip:gw+a1 at 99.39.148.147:5080;transport=udp;gw=a1 SIP/2.0
   Max-Forwards: 70
   Record-Route: <sip:216.143.130.36;lr=on>
   Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bK444b.67fa5084.0
   Via: SIP/2.0/UDP
216.143.130.47:60853;received=216.143.130.47;branch=z9hG4bK.20dfac02;rport=60853;alias
   From: "Unknown" <sip:Unknown at 216.143.130.36>
   To: <sip:a1 at 216.143.130.36>
   Contact: <sip:Unknown at 216.143.130.36>
   Call-ID: 718111ed3c1ed2fb3Rc2Qb6Qr4Nv8Ve8 at 216.143.130.47
   CSeq: 102 NOTIFY
   User-Agent: Asterisk PBX
   Event: message-summary
   Content-Type: application/simple-message-summary
   Content-Length: 36

   Messages-Waiting: no
   Voicemail: 0/0
   ------------------------------------------------------------------------
send 738 bytes to udp/[216.143.130.36]:5060 at 18:48:24.028966:
   ------------------------------------------------------------------------
   SIP/2.0 200 OK
   Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bK444b.67fa5084.0
   Via: SIP/2.0/UDP
216.143.130.47:60853;received=216.143.130.47;branch=z9hG4bK.20dfac02;rport=60853;alias
   Record-Route: <sip:216.143.130.36;lr=on>
   From: "Unknown" <sip:Unknown at 216.143.130.36>
   To: <sip:a1 at 216.143.130.36>;tag=vrU23ac8SFaFK
   Call-ID: 718111ed3c1ed2fb3Rc2Qb6Qr4Nv8Ve8 at 216.143.130.47
   CSeq: 102 NOTIFY
   Contact: <sip:gw+a1 at 99.39.148.147:5080;transport=udp;gw=a1>
   User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-19dc1ab 2010-09-16
20-08-30 +0400
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO,
REGISTER, REFER, NOTIFY
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, hold, refer
   Content-Length: 0

   ------------------------------------------------------------------------