vicidial.org

by **gmcust3** » Fri Feb 12, 2010 8:39 pm

My dialer works perfectly , but just after 6 am IST, whenever I dial a number manually from xlite and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as soon as I press 6.

[root@vici ~]# asterisk -r
Asterisk 1.2.30.2, Copyright (C) 1999 - 2007 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'show license' for details.
=========================================================================
Connected to Asterisk 1.2.30.2 currently running on vici (pid = 2567)
-- Remote UNIX connection
Verbosity is at least 21
-- Executing AGI("SIP/cc105-b7905208", "agi://127.0.0.1:4577/call_log") in n ew stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc105-b7905208", "SIP/VOIP74/17274507674||tTor") in n ew stack
-- Called VOIP74/17274507674
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- SIP/VOIP74-083843c0 is ringing
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/VOIP74-083843c0 is making progress passing it to SIP/cc105-b7905208
== Manager 'sendcron' logged off from 127.0.0.1
-- Registered SIP 'cc101' at 192.168.0.4 port 32668 expires 3600
-- Saved useragent "X-Lite release 1011s stamp 41150" for peer cc101
-- SIP/VOIP74-083843c0 is ringing
-- SIP/VOIP74-083843c0 is making progress passing it to SIP/cc105-b7905208
Feb 12 20:21:14 NOTICE[21532]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 74.222.1.92
-- SIP/VOIP74-083843c0 answered SIP/cc105-b7905208
== Spawn extension (default, 9117274507674, 2) exited non-zero on 'SIP/cc105-b7905208'
-- Executing DeadAGI("SIP/cc105-b7905208", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----15-----4") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --15-----4 completed, returning 0
== Refreshing DNS lookups.
-- Executing AGI("SIP/cc101-b7905208", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc101-b7905208", "SIP/VOIP74/17274507674||tTor") in new stack
-- Called VOIP74/17274507674
-- SIP/VOIP74-083843c0 is ringing
-- SIP/VOIP74-083843c0 is making progress passing it to SIP/cc101-b7905208
-- SIP/VOIP74-083843c0 is ringing
-- SIP/VOIP74-083843c0 is making progress passing it to SIP/cc101-b7905208
Feb 12 20:21:47 NOTICE[21705]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 74.222.1.92
-- SIP/VOIP74-083843c0 answered SIP/cc101-b7905208
== Spawn extension (default, 9117274507674, 2) exited non-zero on 'SIP/cc101-b7905208'
-- Executing DeadAGI("SIP/cc101-b7905208", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----5") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --16-----5 completed, returning 0
vici*CLI>

-- Executing Playback("SIP/VOIP74-083843c0", "vm-goodbye") in new stack
-- Playing 'vm-goodbye' (language 'en')
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/9119852016766@default-c4f5,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/9119852016766@default-c4f5,2", "SIP/VOIP74/19852016766||tTor") in new stack
-- Called VOIP74/19852016766
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/9119853842720@default-3562,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/9119853842720@default-3562,2", "SIP/VOIP74/19853842720||tTor") in new stack
-- Called VOIP74/19853842720
-- Executing Hangup("SIP/VOIP74-083843c0", "") in new stack
== Spawn extension (default, 8307, 3) exited non-zero on 'SIP/VOIP74-083843c0'
-- Executing DeadAGI("SIP/VOIP74-083843c0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- SIP/VOIP74-083b2ca8 is ringing
-- SIP/VOIP74-083b85b0 is ringing
-- SIP/VOIP74-083b85b0 is making progress passing it to Local/9119853842720@default-3562,2
-- SIP/VOIP74-08394b90 answered Local/9119792728183@default-b668,2
> Channel Local/9119792728183@default-b668,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback("Local/9119792728183@default-b668,1", "sip-silence") in new stack
-- Playing 'sip-silence' (language 'en')
== Spawn extension (default, 9119792728183, 2) exited non-zero on 'Local/9119792728183@default-b668,2'
-- Executing DeadAGI("Local/9119792728183@default-b668,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----19-----0") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --19-----0 completed, returning 0
-- Executing AGI("SIP/VOIP74-08394b90", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AMD("SIP/VOIP74-08394b90", "2000|2000|1000|5000|120|50|4|256") in new stack
-- AMD: SIP/VOIP74-08394b90 0000000000 (null) (Fmt: 64)
-- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence [1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [256]
-- AMD: Word detected. iWordsCount:1
-- AMD: Changed state to STATE_IN_SILENCE
-- AMD: HUMAN: silenceDuration:1000 afterGreetingSilence:1000
-- Executing AGI("SIP/VOIP74-08394b90", "VD_amd.agi|8369") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_amd.agi
-- AGI Script VD_amd.agi completed, returning 0
-- Executing AGI("SIP/VOIP74-08394b90", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- SIP/VOIP74-083c6c30 is making progress passing it to Local/9119798284454@default-7e59,2
-- SIP/VOIP74-083c6c30 is ringing
-- SIP/VOIP74-083c6c30 is making progress passing it to Local/9119798284454@default-7e59,2
-- SIP/VOIP74-083a22f8 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
-- Executing Hangup("Local/9119735393604@default-024f,2", "") in new stack
== Spawn extension (default, 9119735393604, 3) exited non-zero on 'Local/9119735393604@default-024f,2'
-- Executing DeadAGI("Local/9119735393604@default-024f,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----1-----CONGESTION----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/VOIP74-083b85b0 is ringing
-- SIP/VOIP74-083b85b0 is making progress passing it to Local/9119853842720@default-3562,2
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
-- Executing Answer("SIP/VOIP74-08394b90", "") in new stack
-- Executing Playback("SIP/VOIP74-08394b90", "vm-goodbye") in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Executing Hangup("SIP/VOIP74-08394b90", "") in new stack
== Spawn extension (default, 8307, 3) exited non-zero on 'SIP/VOIP74-08394b90'
-- Executing DeadAGI("SIP/VOIP74-08394b90", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/VOIP74-083c16f0 answered Local/9119795713525@default-6879,2
> Channel Local/9119795713525@default-6879,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback("Local/9119795713525@default-6879,1", "sip-silence") in new stack
-- Playing 'sip-silence' (language 'en')
== Spawn extension (default, 9119795713525, 2) exited non-zero on 'Local/9119795713525@default-6879,2'
-- Executing DeadAGI("Local/9119795713525@default-6879,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----34-----0") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --34-----0 completed, returning 0
-- Executing AGI("SIP/VOIP74-083c16f0", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AMD("SIP/VOIP74-083c16f0", "2000|2000|1000|5000|120|50|4|256") in new stack
-- AMD: SIP/VOIP74-083c16f0 0000000000 (null) (Fmt: 64)
-- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence [1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [256]
-- AMD: Word detected. iWordsCount:1
-- AMD: Changed state to STATE_IN_SILENCE
-- AMD: Word detected. iWordsCount:2
-- AMD: Changed state to STATE_IN_SILENCE
-- AMD: Word detected. iWordsCount:3
-- AMD: Changed state to STATE_IN_SILENCE
-- AMD: Word detected. iWordsCount:4
-- AMD: ANSWERING MACHINE: iWordsCount:4
-- Executing AGI("SIP/VOIP74-083c16f0", "VD_amd.agi|8369") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_amd.agi
== Spawn extension (default, 8369, 4) exited non-zero on 'SIP/VOIP74-083c16f0'
-- Executing DeadAGI("SIP/VOIP74-083c16f0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/VOIP74-083c6c30 answered Local/9119798284454@default-7e59,2
> Channel Local/9119798284454@default-7e59,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback("Local/9119798284454@default-7e59,1", "sip-silence") in new stack
-- Playing 'sip-silence' (language 'en')
== Spawn extension (default, 9119798284454, 2) exited non-zero on 'Local/9119798284454@default-7e59,2'
-- Executing DeadAGI("Local/9119798284454@default-7e59,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----35-----0") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --35-----0 completed, returning 0
-- Executing AGI("SIP/VOIP74-083c6c30", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AMD("SIP/VOIP74-083c6c30", "2000|2000|1000|5000|120|50|4|256") in new stack
-- AMD: SIP/VOIP74-083c6c30 0000000000 (null) (Fmt: 64)
-- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence [1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [256]
-- AMD: Word detected. iWordsCount:1
-- AMD: Changed state to STATE_IN_SILENCE
-- AMD: HUMAN: silenceDuration:1000 afterGreetingSilence:1000
-- Executing AGI("SIP/VOIP74-083c6c30", "VD_amd.agi|8369") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_amd.agi
-- AGI Script VD_amd.agi completed, returning 0
-- Executing AGI("SIP/VOIP74-083c6c30", "agi-VDAD_ALL_outbound.agi|NORMAL-----LB") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
-- SIP/VOIP74-083b85b0 is ringing
-- SIP/VOIP74-083b85b0 is making progress passing it to Local/9119853842720@default-3562,2
-- AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0
-- Executing Answer("SIP/VOIP74-083c6c30", "") in new stack
-- Executing Playback("SIP/VOIP74-083c6c30", "vm-goodbye") in new stack
-- Playing 'vm-goodbye' (language 'en')
-- Executing Hangup("SIP/VOIP74-083c6c30", "") in new stack
== Spawn extension (default, 8307, 3) exited non-zero on 'SIP/VOIP74-083c6c30'
-- Executing DeadAGI("SIP/VOIP74-083c6c30", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- SIP/VOIP74-083b85b0 answered Local/9119853842720@default-3562,2
> Channel Local/9119853842720@default-3562,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback("Local/9119853842720@default-3562,1", "sip-silence") in new stack
== Spawn extension (default, 9119853842720, 2) exited non-zero on 'Local/9119853842720@default-3562,2'
-- Executing DeadAGI("Local/9119853842720@default-3562,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----36-----0") in new stack
-- Playing 'sip-silence' (language 'en')
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --36-----0 completed, returning 0
-- Executing AGI("SIP/VOIP74-083b85b0", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing AMD("SIP/VOIP74-083b85b0", "2000|2000|1000|5000|120|50|4|256") in new stack
-- AMD: SIP/VOIP74-083b85b0 0000000000 (null) (Fmt: 64)
-- AMD: initialSilence [2000] greeting [2000] afterGreetingSilence [1000] totalAnalysisTime [5000] minimumWordLength [120] betweenWordsSilence [50] maximumNumberOfWords [4] silenceThreshold [256]
-- SIP/VOIP74-083b2ca8 is making progress passing it to Local/9119852016766@default-c4f5,2
-- AMD: Word detected. iWordsCount:1
-- AMD: Changed state to STATE_IN_SILENCE
-- AMD: Word detected. iWordsCount:2
-- AMD: Changed state to STATE_IN_SILENCE
-- AMD: Word detected. iWordsCount:3
-- AMD: Changed state to STATE_IN_SILENCE
-- AMD: Word detected. iWordsCount:4
-- AMD: ANSWERING MACHINE: iWordsCount:4
-- Executing AGI("SIP/VOIP74-083b85b0", "VD_amd.agi|8369") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/VD_amd.agi
== Spawn extension (default, 8369, 4) exited non-zero on 'SIP/VOIP74-083b85b0'
-- Executing DeadAGI("SIP/VOIP74-083b85b0", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Spawn extension (default, 9119852016766, 2) exited non-zero on 'Local/9119852016766@default-c4f5,2'
-- Executing DeadAGI("Local/9119852016766@default-c4f5,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
-- Executing AGI("SIP/cc101-b7901338", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc101-b7901338", "SIP/VOIP74/12062036895||tTor") in new stack
-- Called VOIP74/12062036895
== Refreshing DNS lookups.
-- SIP/VOIP74-083843c0 is ringing
-- SIP/VOIP74-083843c0 is making progress passing it to SIP/cc101-b7901338
-- SIP/VOIP74-083843c0 is ringing
-- SIP/VOIP74-083843c0 is making progress passing it to SIP/cc101-b7901338
Feb 12 20:26:29 NOTICE[23766]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 74.222.1.92
-- SIP/VOIP74-083843c0 answered SIP/cc101-b7901338
== Spawn extension (default, 9112062036895, 2) exited non-zero on 'SIP/cc101-b7901338'
-- Executing DeadAGI("SIP/cc101-b7901338", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----13-----4") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --13-----4 completed, returning 0
vici*CLI>
vici*CLI>
-- Executing AGI("SIP/cc105-0918e450", "agi://127.0.0.1:4577/call_log") in n ew stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc105-0918e450", "SIP/VOIP74/17274507674||tTor") in n ew stack
-- Called VOIP74/17274507674
-- SIP/VOIP74-0919f390 is ringing
-- SIP/VOIP74-0919f390 is making progress passing it to SIP/cc105-0918e450
-- Registered SIP 'cc102' at 192.168.0.1 port 2426 expires 3600
-- Saved useragent "X-Lite release 1011s stamp 41150" for peer cc102
-- SIP/VOIP74-0919f390 is ringing
-- SIP/VOIP74-0919f390 is making progress passing it to SIP/cc105-0918e450
Feb 12 20:31:39 NOTICE[2807]: rtp.c:331 process_rfc3389: Comfort noise support i ncomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 74.222.1.92
-- SIP/VOIP74-0919f390 answered SIP/cc105-0918e450
== Spawn extension (default, 9117274507674, 2) exited non-zero on 'SIP/cc105-0 918e450'
-- Executing DeadAGI("SIP/cc105-0918e450", "agi://127.0.0.1:4577/call_log--H Vcauses--PRI-----NODEBUG-----16-----ANSWER-----15-----4") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----1 6-----ANSWER-----15-----4 completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1

What could be the issues ?

by **gardo** » Sat Feb 13, 2010 2:26 am

What did your VoIP provider say?

by **gmcust3** » Sat Feb 13, 2010 4:46 am

I tried the SAME VOIP from another center and Its Ok there.

I tried the Same dialer Xlite over Static IP, problem is there.

I tried the same number from other Dialer , it works perfectly.

CLI is of no help ?

by **gmcust3** » Sat Feb 13, 2010 6:39 pm

Any other output apart from CLI will help ?

by **gmcust3** » Sun Feb 14, 2010 1:54 am

Normal Hang Up :
-----------------------------

vici*CLI>
-- Executing AGI("SIP/cc101-09f44300", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc101-09f44300", "SIP/VOIP74/17274507674||tTor") in new stack
-- Called VOIP74/17274507674
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/VOIP74-09ecb770 is ringing
-- SIP/VOIP74-09ecb770 is making progress passing it to SIP/cc101-09f44300
-- SIP/VOIP74-09ecb770 is ringing
-- SIP/VOIP74-09ecb770 is making progress passing it to SIP/cc101-09f44300
Feb 14 01:50:21 NOTICE[24692]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 74.222.1.92
-- SIP/VOIP74-09ecb770 answered SIP/cc101-09f44300
== Spawn extension (default, 9117274507674, 2) exited non-zero on 'SIP/cc101-09f44300'
-- Executing DeadAGI("SIP/cc101-09f44300", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----21-----11") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -21-----11 completed, returning 0
vici*CLI>

Hang Up when pressed any key from the soft Phone:
-------------------------------------------------------------------------------

vici*CLI>
-- Executing AGI("SIP/cc101-09f44300", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("SIP/cc101-09f44300", "SIP/VOIP74/17274507674||tTor") in new stack
-- Called VOIP74/17274507674
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/VOIP74-09ecb770 is ringing
-- SIP/VOIP74-09ecb770 is making progress passing it to SIP/cc101-09f44300
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/VOIP74-09ecb770 is ringing
-- SIP/VOIP74-09ecb770 is making progress passing it to SIP/cc101-09f44300
Feb 14 01:51:16 NOTICE[24845]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 74.222.1.92
-- SIP/VOIP74-09ecb770 answered SIP/cc101-09f44300
== Spawn extension (default, 9117274507674, 2) exited non-zero on 'SIP/cc101-09f44300'
-- Executing DeadAGI("SIP/cc101-09f44300", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----22-----10") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -22-----10 completed, returning 0
vici*CLI>

Dial Plan :

register =>user:pass123@74.222.1.92:5060

[VOIP74_7]
disallow=all
allow=g729
allow=g711
allow=ulaw
type=friend
username=user
secret=password
host=74.222.1.92
dtmfmode=rfc2833

SIP74_7 = SIP/VOIP74_7

exten => _7X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _7X.,2,Dial(${SIP74_7}/${EXTEN:2},,tTor)
exten => _7X.,3,Hangup

Please guide me .

Entry from Master.csv

""cc101" <cc101>","cc101","9117274507674","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","DeadAGI","agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----","2010-02-14 01:47:02","2010-02-14 01:47:14","2010-02-14 01:47:19","17","5","ANSWERED","DOCUMENTATION","","1266130022.0",""
""cc101" <cc101>","cc101","9119545090201","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","DeadAGI","agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----","2010-02-14 01:47:35","2010-02-14 01:47:38","2010-02-14 01:47:42","7","4","ANSWERED","DOCUMENTATION","","1266130055.2",""
""cc101" <cc101>","cc101","9119545090201","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","DeadAGI","agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----","2010-02-14 01:48:06","2010-02-14 01:48:09","2010-02-14 01:48:14","8","5","ANSWERED","DOCUMENTATION","","1266130086.4",""
""cc101" <cc101>","cc101","9117274507674","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","Dial","SIP/VOIP74/17274507674||tTor","2010-02-14 01:48:24","2010-02-14 01:48:35","2010-02-14 01:48:38","14","3","ANSWERED","DOCUMENTATION","","1266130104.6",""
""cc101" <cc101>","cc101","9117274507674","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","DeadAGI","agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----","2010-02-14 01:48:43","2010-02-14 01:48:55","2010-02-14 01:48:57","14","2","ANSWERED","DOCUMENTATION","","1266130123.8",""
""cc101" <cc101>","cc101","9117274507674","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","Dial","SIP/VOIP74/17274507674||tTor","2010-02-14 01:50:12","2010-02-14 01:50:22","2010-02-14 01:50:33","21","11","ANSWERED","DOCUMENTATION","","1266130212.10",""
""cc101" <cc101>","cc101","9117274507674","default","SIP/cc101-09f44300","SIP/VOIP74-09ecb770","DeadAGI","agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----","2010-02-14 01:51:05","2010-02-14 01:51:17","2010-02-14 01:51:27","22","10","ANSWERED","DOCUMENTATION","","1266130265.12",""
""cc101" <cc101>","cc101","7117274507674","default","SIP/cc101-09f44300","SIP/VOIP74_7-09ecb770","DeadAGI","agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----","2010-02-14 01:57:20","2010-02-14 01:57:32","2010-02-14 01:57:36","16","4","ANSWERED","DOCUMENTATION","","1266130640.14",""
""cc101" <cc101>","cc101","7117274507674","default","SIP/cc101-09f44300","SIP/VOIP74_7-09ecb770","Dial","SIP/VOIP74_7/17274507674||tTor","2010-02-14 02:00:57","","2010-02-14 02:00:59","2","0","NO ANSWER","DOCUMENTATION","","1266130857.16",""

Also, I see that my event log file size is 0.

by **gmcust3** » Sun Feb 14, 2010 6:35 pm

When I sip debug ::::::

Normal Hang Up :
****************************************

vici*CLI>
<-- SIP read from 74.222.1.92:5060:
SIP/2.0 180 Ringing
CSeq: 103 INVITE
Via: SIP/2.0/UDP 59.XXX.45.XXX:5060;branch=z9hG4bK222028a9;rport
From: "cc101" <sip:cc101@59.XXX.45.XXX>;tag=as0774001a
Call-ID: 5ee8bb432d30975d451e7b6b5ed50ce9@59.XXX.45.XXX
To: <sip:17274507674@74.222.1.92>;tag=140230101526679285626473
Contact: <sip:74.222.1.92:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 244

v=0
o=VoipSwitch 7472 7472 IN IP4 74.222.1.92
s=VoipSIP
i=Audio Session
c=IN IP4 74.222.1.92
t=0 0
m=audio 6472 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (9 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 74.222.1.92:6472
Found description format G729
Found description format telephone-event
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
-- SIP/VOIP74_7-09f86b70 is ringing
-- SIP/VOIP74_7-09f86b70 is making progress passing it to SIP/cc101-09f44300
Feb 14 18:30:40 NOTICE[1338]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 74.222.1.92
vici*CLI>
<-- SIP read from 74.222.1.92:5060:
SIP/2.0 200 OK
CSeq: 103 INVITE
Via: SIP/2.0/UDP 59.XXX.45.XXX:5060;branch=z9hG4bK222028a9;rport
From: "cc101" <sip:cc101@59.XXX.45.XXX>;tag=as0774001a
Call-ID: 5ee8bb432d30975d451e7b6b5ed50ce9@59.XXX.45.XXX
To: <sip:17274507674@74.222.1.92>;tag=140230101526679285626473
Contact: <sip:74.222.1.92:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 244

v=0
o=VoipSwitch 7472 7472 IN IP4 74.222.1.92
s=VoipSIP
i=Audio Session
c=IN IP4 74.222.1.92
t=0 0
m=audio 6472 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (9 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 74.222.1.92:6472
Found description format G729
Found description format telephone-event
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:74.222.1.92:5060;transport=udp>
set_destination: Parsing <sip:74.222.1.92:5060;transport=udp> for address/port to send to
set_destination: set destination to 74.222.1.92, port 5060
Transmitting (NAT) to 74.222.1.92:5060:
ACK sip:74.222.1.92:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 59.XXX.45.XXX:5060;branch=z9hG4bK5d8ae005;rport
From: "cc101" <sip:cc101@59.XXX.45.XXX>;tag=as0774001a
To: <sip:17274507674@74.222.1.92>;tag=140230101526679285626473
Contact: <sip:cc101@59.XXX.45.XXX>
Call-ID: 5ee8bb432d30975d451e7b6b5ed50ce9@59.XXX.45.XXX
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "cc101" <sip:cc101@59.XXX.45.XXX>;privacy=off;screen=no
Content-Length: 0

---
-- SIP/VOIP74_7-09f86b70 answered SIP/cc101-09f44300
We're at 59.XXX.45.XXX port 13470
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 59.94.2.209:36482:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 59.94.2.209:36482;branch=z9hG4bK-d8754z-eb2be41f8d755477-1---d8754z-;received=59.94.2.209;rport=36482
From: "cc101"<sip:cc101@59.XXX.45.XXX>;tag=e01e265a
To: "7117274507674"<sip:7117274507674@59.XXX.45.XXX>;tag=as5d7fa7a6
Call-ID: ZmJhZjAzZTA5ZDU2YzQ4Y2EyODk0MDkwMzFhNjZjZTE.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7117274507674@59.XXX.45.XXX>
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 2568 2568 IN IP4 59.XXX.45.XXX
s=session
c=IN IP4 59.XXX.45.XXX
t=0 0
m=audio 13470 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
vici*CLI>
<-- SIP read from 59.94.2.209:36482:
ACK sip:7117274507674@59.XXX.45.XXX SIP/2.0
Via: SIP/2.0/UDP 59.94.2.209:36482;branch=z9hG4bK-d8754z-475a8b6a87686538-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc101@59.94.2.209:36482>
To: "7117274507674"<sip:7117274507674@59.XXX.45.XXX>;tag=as5d7fa7a6
From: "cc101"<sip:cc101@59.XXX.45.XXX>;tag=e01e265a
Call-ID: ZmJhZjAzZTA5ZDU2YzQ4Y2EyODk0MDkwMzFhNjZjZTE.
CSeq: 2 ACK
Proxy-Authorization: Digest username="cc101",realm="asterisk",nonce="394ac427",uri="sip:7117274507674@59.XXX.45.XXX",response="ecad08cd4c00549c52250b7a98b0a3f8",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

--- (11 headers 0 lines) ---
vici*CLI>
<-- SIP read from 59.94.2.209:36482:
BYE sip:7117274507674@59.XXX.45.XXX SIP/2.0
Via: SIP/2.0/UDP 59.94.2.209:36482;branch=z9hG4bK-d8754z-092ce364e31cd771-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc101@59.94.2.209:36482>
To: "7117274507674"<sip:7117274507674@59.XXX.45.XXX>;tag=as5d7fa7a6
From: "cc101"<sip:cc101@59.XXX.45.XXX>;tag=e01e265a
Call-ID: ZmJhZjAzZTA5ZDU2YzQ4Y2EyODk0MDkwMzFhNjZjZTE.
CSeq: 3 BYE
Proxy-Authorization: Digest username="cc101",realm="asterisk",nonce="394ac427",uri="sip:7117274507674@59.XXX.45.XXX",response="1d91968f593763cf37cc17ee8f133f60",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Reason: SIP;description="User Hung Up"
Content-Length: 0

--- (12 headers 0 lines) ---
Sending to 59.94.2.209 : 36482 (NAT)
Transmitting (NAT) to 59.94.2.209:36482:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 59.94.2.209:36482;branch=z9hG4bK-d8754z-092ce364e31cd771-1---d8754z-;received=59.94.2.209;rport=36482
From: "cc101"<sip:cc101@59.XXX.45.XXX>;tag=e01e265a
To: "7117274507674"<sip:7117274507674@59.XXX.45.XXX>;tag=as5d7fa7a6
Call-ID: ZmJhZjAzZTA5ZDU2YzQ4Y2EyODk0MDkwMzFhNjZjZTE.
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7117274507674@59.XXX.45.XXX>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing

---
Scheduling destruction of call '5ee8bb432d30975d451e7b6b5ed50ce9@59.XXX.45.XXX' in 32000 ms
set_destination: Parsing <sip:74.222.1.92:5060;transport=udp> for address/port to send to
set_destination: set destination to 74.222.1.92, port 5060
Reliably Transmitting (NAT) to 74.222.1.92:5060:
BYE sip:74.222.1.92:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 59.XXX.45.XXX:5060;branch=z9hG4bK268240ba;rport
From: "cc101" <sip:cc101@59.XXX.45.XXX>;tag=as0774001a
To: <sip:17274507674@74.222.1.92>;tag=140230101526679285626473
Call-ID: 5ee8bb432d30975d451e7b6b5ed50ce9@59.XXX.45.XXX
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "cc101" <sip:cc101@59.XXX.45.XXX>;privacy=off;screen=no
Proxy-Authorization: Digest username="deepak", realm="VoipSwitch", algorithm=MD5, uri="sip:74.222.1.92:5060", nonce="126619022614150304502302607006", response="0fe16f2311d651db084705c96593aabb", opaque=""
Content-Length: 0

---
== Spawn extension (default, 7117274507674, 2) exited non-zero on 'SIP/cc101-09f44300'
-- Executing DeadAGI("SIP/cc101-09f44300", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----15-----4") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --15-----4 completed, returning 0
Destroying call 'ZmJhZjAzZTA5ZDU2YzQ4Y2EyODk0MDkwMzFhNjZjZTE.'
vici*CLI>
<-- SIP read from 74.222.1.92:5060:
SIP/2.0 200 OK
CSeq: 104 BYE
Via: SIP/2.0/UDP 59.XXX.45.XXX:5060;branch=z9hG4bK268240ba;rport
From: "cc101" <sip:cc101@59.XXX.45.XXX>;tag=as0774001a
Call-ID: 5ee8bb432d30975d451e7b6b5ed50ce9@59.XXX.45.XXX
To: <sip:17274507674@74.222.1.92>;tag=140230101526679285626473
Contact: <sip:74.222.1.92:5060;transport=udp>
Content-Length: 0

--- (8 headers 0 lines) ---
Destroying call '5ee8bb432d30975d451e7b6b5ed50ce9@59.XXX.45.XXX'
vici*CLI>

Hang when i press any Key in Xlite:
*********************************************

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

---
Destroying call 'N2U0YzNkYTE3M2M3MTc2YjM5ZDk2MmI5M2M4ZGExN2E.'
vici*CLI>
<-- SIP read from 74.222.1.92:5060:
SIP/2.0 180 Ringing
CSeq: 103 INVITE
Via: SIP/2.0/UDP 59.XXX.45.XXX:5060;branch=z9hG4bK23c435f4;rport
From: "cc101" <sip:cc101@59.XXX.45.XXX>;tag=as55640593
Call-ID: 624d703b163c67d12e3de1d229b0ce86@59.XXX.45.XXX
To: <sip:17274507674@74.222.1.92>;tag=140231101542680040006477
Contact: <sip:74.222.1.92:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 244

v=0
o=VoipSwitch 7476 7476 IN IP4 74.222.1.92
s=VoipSIP
i=Audio Session
c=IN IP4 74.222.1.92
t=0 0
m=audio 6476 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (9 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 74.222.1.92:6476
Found description format G729
Found description format telephone-event
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
-- SIP/VOIP74_7-09f86b70 is ringing
-- SIP/VOIP74_7-09f86b70 is making progress passing it to SIP/cc101-09f44300
Feb 14 18:31:55 NOTICE[1545]: rtp.c:331 process_rfc3389: Comfort noise support incomplete in Asterisk (RFC 3389). Please turn off on client if possible. Client IP: 74.222.1.92
vici*CLI>
<-- SIP read from 74.222.1.92:5060:
SIP/2.0 200 OK
CSeq: 103 INVITE
Via: SIP/2.0/UDP 59.XXX.45.XXX:5060;branch=z9hG4bK23c435f4;rport
From: "cc101" <sip:cc101@59.XXX.45.XXX>;tag=as55640593
Call-ID: 624d703b163c67d12e3de1d229b0ce86@59.XXX.45.XXX
To: <sip:17274507674@74.222.1.92>;tag=140231101542680040006477
Contact: <sip:74.222.1.92:5060;transport=udp>
Content-Type: application/sdp
Content-Length: 244

v=0
o=VoipSwitch 7476 7476 IN IP4 74.222.1.92
s=VoipSIP
i=Audio Session
c=IN IP4 74.222.1.92
t=0 0
m=audio 6476 RTP/AVP 18 101
a=rtpmap:18 G729/8000/1
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (9 headers 12 lines) ---
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 74.222.1.92:6476
Found description format G729
Found description format telephone-event
Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x100 (g729)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:74.222.1.92:5060;transport=udp>
set_destination: Parsing <sip:74.222.1.92:5060;transport=udp> for address/port to send to
set_destination: set destination to 74.222.1.92, port 5060
Transmitting (NAT) to 74.222.1.92:5060:
ACK sip:74.222.1.92:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 59.XXX.45.XXX:5060;branch=z9hG4bK4ea97732;rport
From: "cc101" <sip:cc101@59.XXX.45.XXX>;tag=as55640593
To: <sip:17274507674@74.222.1.92>;tag=140231101542680040006477
Contact: <sip:cc101@59.XXX.45.XXX>
Call-ID: 624d703b163c67d12e3de1d229b0ce86@59.XXX.45.XXX
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "cc101" <sip:cc101@59.XXX.45.XXX>;privacy=off;screen=no
Content-Length: 0

---
-- SIP/VOIP74_7-09f86b70 answered SIP/cc101-09f44300
We're at 59.XXX.45.XXX port 13238
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 59.94.2.209:36482:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 59.94.2.209:36482;branch=z9hG4bK-d8754z-0e09ec7cb967eb6d-1---d8754z-;received=59.94.2.209;rport=36482
From: "cc101"<sip:cc101@59.XXX.45.XXX>;tag=8811bd08
To: "7117274507674"<sip:7117274507674@59.XXX.45.XXX>;tag=as280d5d4b
Call-ID: ZDkzZjY1NDQ0YzgzYjBhNmM2OTdkMzIxYmM0YjcwMDA.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:7117274507674@59.XXX.45.XXX>
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 2568 2568 IN IP4 59.XXX.45.XXX
s=session
c=IN IP4 59.XXX.45.XXX
t=0 0
m=audio 13238 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
vici*CLI>
<-- SIP read from 59.94.2.209:36482:
ACK sip:7117274507674@59.XXX.45.XXX SIP/2.0
Via: SIP/2.0/UDP 59.94.2.209:36482;branch=z9hG4bK-d8754z-487fae12ea2f9c6c-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc101@59.94.2.209:36482>
To: "7117274507674"<sip:7117274507674@59.XXX.45.XXX>;tag=as280d5d4b
From: "cc101"<sip:cc101@59.XXX.45.XXX>;tag=8811bd08
Call-ID: ZDkzZjY1NDQ0YzgzYjBhNmM2OTdkMzIxYmM0YjcwMDA.
CSeq: 2 ACK
Proxy-Authorization: Digest username="cc101",realm="asterisk",nonce="1e8379a5",uri="sip:7117274507674@59.XXX.45.XXX",response="f6b6862539a964ab30b3db09f3e2e9b1",algorithm=MD5
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

--- (11 headers 0 lines) ---
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
== Manager 'sendcron' logged off from 127.0.0.1
vici*CLI>
<-- SIP read from 74.222.1.92:5060:
BYE sip:cc101@59.XXX.45.XXX SIP/2.0
CSeq: 1 BYE
Via: SIP/2.0/UDP 74.222.1.92:5060;branch=z9hG4bk1402311015571468019359
From: <sip:17274507674@74.222.1.92>;tag=140231101542680040006477
Call-ID: 624d703b163c67d12e3de1d229b0ce86@59.XXX.45.XXX
To: "cc101" <sip:cc101@59.XXX.45.XXX>;tag=as55640593
Content-Length: 0

--- (7 headers 0 lines) ---
Sending to 74.222.1.92 : 5060 (NAT)
Transmitting (NAT) to 74.222.1.92:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 74.222.1.92:5060;branch=z9hG4bk1402311015571468019359;received=74.222.1.92
From: <sip:17274507674@74.222.1.92>;tag=140231101542680040006477
To: "cc101" <sip:cc101@59.XXX.45.XXX>;tag=as55640593
Call-ID: 624d703b163c67d12e3de1d229b0ce86@59.XXX.45.XXX
CSeq: 1 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:cc101@59.XXX.45.XXX>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing

---
== Spawn extension (default, 7117274507674, 2) exited non-zero on 'SIP/cc101-09f44300'
-- Executing DeadAGI("SIP/cc101-09f44300", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----15-----4") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --15-----4 completed, returning 0
Scheduling destruction of call 'ZDkzZjY1NDQ0YzgzYjBhNmM2OTdkMzIxYmM0YjcwMDA.' in 32000 ms
set_destination: Parsing <sip:cc101@59.94.2.209:36482> for address/port to send to
set_destination: set destination to 59.94.2.209, port 36482
Reliably Transmitting (NAT) to 59.94.2.209:36482:
BYE sip:cc101@59.94.2.209:36482 SIP/2.0
Via: SIP/2.0/UDP 59.XXX.45.XXX:5060;branch=z9hG4bK19721f77;rport
From: "7117274507674"<sip:7117274507674@59.XXX.45.XXX>;tag=as280d5d4b
To: "cc101"<sip:cc101@59.XXX.45.XXX>;tag=8811bd08
Call-ID: ZDkzZjY1NDQ0YzgzYjBhNmM2OTdkMzIxYmM0YjcwMDA.
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
Destroying call '624d703b163c67d12e3de1d229b0ce86@59.XXX.45.XXX'
vici*CLI>
<-- SIP read from 59.94.2.209:36482:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 59.XXX.45.XXX:5060;branch=z9hG4bK19721f77;rport=5060
Contact: <sip:cc101@59.94.2.209:36482>
To: "cc101"<sip:cc101@59.XXX.45.XXX>;tag=8811bd08
From: "7117274507674"<sip:7117274507674@59.XXX.45.XXX>;tag=as280d5d4b
Call-ID: ZDkzZjY1NDQ0YzgzYjBhNmM2OTdkMzIxYmM0YjcwMDA.
CSeq: 102 BYE
User-Agent: X-Lite release 1104o stamp 56125
Content-Length: 0

--- (9 headers 0 lines) ---
Destroying call 'ZDkzZjY1NDQ0YzgzYjBhNmM2OTdkMzIxYmM0YjcwMDA.'
vici*CLI>

If no Soluton, do I need to Reinstall all VNOw 1.3 again ?

Or just asterisk ?

by **gmcust3** » Sun Feb 14, 2010 6:49 pm

Interestingly .Note this :

Phone Number : 7274507674 Room ID: 6055

When I dial this number through Xlite and asterisk , on pressing any key , line get disconnect.

When I dial this number through Skype, its perfect.

Phone Number : 2127773456

When I dial this number through Xlite and asterisk , on pressing any key , line DOESNT get disconnect.

When I dial this number through Skype, its perfect.

vicidial.org

Line DC

Line DC

Who is online