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Connected, but No calls to agent

General and Support topics relating to ViciDialNow and GoAutoDial ISO installers

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Connected, but No calls to agent

Postby gmcust3 » Fri Feb 19, 2010 4:49 pm

Code: Select all
[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 = 3502)
Verbosity is at least 21
  == Spawn extension (default, 7118434574589, 2) exited non-zero on 'Local/7118434574589@default-007e,2'
    -- Executing DeadAGI("Local/7118434574589@default-007e,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--PRI-----NODEBUG-----0-----CANCEL---------- completed, returning 0
  == Spawn extension (default, 7115742024489, 2) exited non-zero on 'Local/7115742024489@default-080a,2'
    -- Executing DeadAGI("Local/7115742024489@default-080a,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--PRI-----NODEBUG-----0-----CANCEL---------- completed, returning 0
  == Spawn extension (default, 7114153536171, 2) exited non-zero on 'Local/7114153536171@default-50a6,2'
    -- Executing DeadAGI("Local/7114153536171@default-50a6,2", "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--PRI-----NODEBUG-----0-----CANCEL---------- completed, returning 0
  == Manager 'sendcron' logged off from 127.0.0.1
  == Spawn extension (default, 8369, 1) exited non-zero on 'SIP/Fid74-0990b968'
    -- Executing DeadAGI("SIP/Fid74-0990b968", "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--PRI-----NODEBUG-----0--------------- completed, returning 0
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
Feb 19 16:44:13 NOTICE[4575]: channel.c:2514 __ast_request_and_dial: Unable to request channel SIP/cc104
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing AGI("Local/7113016436516@default-f0c4,2", "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/7113016436516@default-f0c4,2", "SIP/Fid74/13016436516||tTor") in new stack
    -- Called Fid74/13016436516
  == Spawn extension (default, 7113043455620, 2) exited non-zero on 'Local/7113043455620@default-fbed,2'
    -- Executing DeadAGI("Local/7113043455620@default-fbed,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
  == Manager 'sendcron' logged off from 127.0.0.1
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL---------- completed, returning 0
  == Spawn extension (default, 7117076498231, 2) exited non-zero on 'Local/7117076498231@default-aff1,2'
    -- Executing DeadAGI("Local/7117076498231@default-aff1,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
    -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL---------- completed, returning 0
  == Spawn extension (default, 7115019825414, 2) exited non-zero on 'Local/7115019825414@default-5783,2'
    -- Executing DeadAGI("Local/7115019825414@default-5783,2", "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--PRI-----NODEBUG-----0-----CANCEL---------- completed, returning 0
    -- SIP/Fid74-0990b968 is ringing
  == Manager 'sendcron' logged off from 127.0.0.1
  == Spawn extension (default, 8369, 1) exited non-zero on 'SIP/Fid74-0993d888'
    -- Executing DeadAGI("SIP/Fid74-0993d888", "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--PRI-----NODEBUG-----0--------------- completed, returning 0
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing AGI("Local/7118582022706@default-47bf,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/7118582022706@default-47bf,2", "SIP/Fid74/18582022706||tTor") in new stack
    -- Called Fid74/18582022706
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing AGI("Local/7114232883988@default-68ff,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/7114232883988@default-68ff,2", "SIP/Fid74/14232883988||tTor") in new stack
    -- Called Fid74/14232883988
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing AGI("Local/7117349443330@default-66cf,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/7117349443330@default-66cf,2", "SIP/Fid74/17349443330||tTor") in new stack
    -- Called Fid74/17349443330
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing AGI("Local/7114098869473@default-3ac8,2", "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/7114098869473@default-3ac8,2", "SIP/Fid74/14098869473||tTor") in new stack
    -- Called Fid74/14098869473
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing AGI("Local/7119518175078@default-70e1,2", "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/7119518175078@default-70e1,2", "SIP/Fid74/19518175078||tTor") in new stack
    -- Called Fid74/19518175078
  == 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
    -- Executing AGI("Local/7112032347656@default-cfa4,2", "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/7112032347656@default-cfa4,2", "SIP/Fid74/12032347656||tTor") in new stack
    -- Called Fid74/12032347656
    -- Executing AGI("Local/7114107685555@default-ac32,2", "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/7114107685555@default-ac32,2", "SIP/Fid74/14107685555||tTor") in new stack
    -- Called Fid74/14107685555
  == Parsing '/etc/asterisk/manager.conf': Found
  == Manager 'sendcron' logged on from 127.0.0.1
    -- Executing AGI("Local/7119083033527@default-4bc5,2", "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/7119083033527@default-4bc5,2", "SIP/Fid74/19083033527||tTor") in new stack
    -- Called Fid74/19083033527
    -- SIP/Fid74-09912690 is ringing
    -- SIP/Fid74-09904db0 is ringing
    -- SIP/Fid74-0993d888 is ringing
    -- SIP/Fid74-09944678 is ringing
    -- SIP/Fid74-09936d30 is ringing
    -- SIP/Fid74-09930288 is ringing
    -- SIP/Fid74-098fe098 is ringing
    -- SIP/Fid74-09919268 is ringing
vici*CLI>



vici*CLI>
== Manager 'sendcron' logged off from 127.0.0.1
-- SIP/Fid74-0993bb48 answered Local/7112127773456@default-749b,2
> Channel Local/7112127773456@default-749b,1 was answered.
== Manager 'sendcron' logged off from 127.0.0.1
-- Executing Playback("Local/7112127773456@default-749b,1", "sip-silence") in new stack
-- Playing 'sip-silence' (language 'en')
== Spawn extension (default, 7112127773456, 2) exited non-zero on 'Local/7112127773456@default-749b,2'
-- Executing DeadAGI("Local/7112127773456@default-749b,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----16-----0") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --16-----0 completed, returning 0
== Manager 'sendcron' logged off from 127.0.0.1
== Spawn extension (default, 7112127773456, 2) exited non-zero on 'Local/7112127773456@default-341b,2'
-- Executing DeadAGI("Local/7112127773456@default-341b,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") 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
== Spawn extension (default, 7112127773456, 2) exited non-zero on 'Local/7112127773456@default-33fe,2'
-- Executing DeadAGI("Local/7112127773456@default-33fe,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") 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
== Spawn extension (default, 7112127773456, 2) exited non-zero on 'Local/7112127773456@default-6244,2'
-- Executing DeadAGI("Local/7112127773456@default-6244,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
vici*CLI>



Calls are connecting as I can see in VOIP VSR panel as calls are getting connected but agents not getting calls.

Above CLI will help in anyway ?

Admin screen shows

Friday February 19, 2010 17:09:20 PM

and

On Server , date shows :

Fri Feb 19 17:07:31 EST 2010

Lead Stats :

Code: Select all
STATUSES WITHIN THIS LIST:
STATUS   STATUS NAME   CALLED   NOT CALLED
AA   Answering Machine Auto   10   0
DC   Disconnected Number   1   0
DROP   Agent Not Available   1   0
ERI   Agent Error   1   0
N   No Answer   5   0
NA   No Answer AutoDial   805   0
NEW   New Lead   204   14255
NI   Not Interested   1   0
SUBTOTALS   1028   14255
TOTAL   15283

GoAutoDial CE
VERSION: 2.4-309a
BUILD: 110430-1642
No other software installed on the box.
I've read the manager manual.
gmcust3
 
Posts: 1148
Joined: Sat Oct 24, 2009 1:15 pm
Top

Postby williamconley » Fri Feb 26, 2010 1:47 am

what is a "VOIP VSR"?

do you have any other software in this server besides what came with vicidialnow?

i'd like to see the results of a single call attempt. i never saw "answer" and what happens after answer, just the END of a call that was theoretically answered ... and lots of ringing ...

has this server ever worked or is it "new"?
Vicidial Installation and Repair, plus Hosting and Colocation
Newest Product: Vicidial Agent Only Beep - Beta
http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
williamconley
 
Posts: 20253
Joined: Wed Oct 31, 2007 4:17 pm
Location: Davenport, FL (By Disney!)
Top

Re: Connected, but No calls to agent

Postby terry.sahil » Mon Apr 13, 2015 4:24 pm

I was also facing the same issue when I put my phone in "ON-hook" mode then calls rings on phone and as soon as I answer the call ,it got disconnected . This happen only when I use hard IP phone Grandstream GPX 1400 .Calls try to ring again and again but unable to answer that call . Please help ....
terry.sahil
 
Posts: 5
Joined: Thu Jun 13, 2013 6:50 am
Top

Re: Connected, but No calls to agent

Postby williamconley » Mon Apr 13, 2015 4:35 pm

Use sip debug to determine the cause of the disconnect. Often turns out to be a codec problem.
Vicidial Installation and Repair, plus Hosting and Colocation
Newest Product: Vicidial Agent Only Beep - Beta
http://www.PoundTeam.com # 352-269-0000 # +44(203) 769-2294
williamconley
 
Posts: 20253
Joined: Wed Oct 31, 2007 4:17 pm
Location: Davenport, FL (By Disney!)
Top

Re: Connected, but No calls to agent

Postby terry.sahil » Tue May 05, 2015 6:21 pm

Hi,

Thanks for your reply . Still facing the same issue . I checked the logs but I did not found any visible issue . Please have a look on below given logs :

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

<------------->
[May 6 03:47:48] --- (12 headers 0 lines) ---
[May 6 03:47:48] list_route: hop: <sip:7777@182.72.102.41:5060>
[May 6 03:47:48] -- SIP/7777-000017c4 is ringing
[May 6 03:47:50] -- Started music on hold, class 'default', on SIP/united-AUS1-000017c3
[May 6 03:47:50]
<--- SIP read from UDP:182.72.102.41:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.103.5.6:5060;branch=z9hG4bK7795bdf4
From: "MyTech" <sip:asterisk@63.103.5.6>;tag=as51948124
To: <sip:7777@182.72.102.41:5060>;tag=165290820
Call-ID: 7a5811034ee44e071fcf457c1a3c8a5d@63.103.5.6:5060
CSeq: 102 INVITE
Contact: <sip:7777@182.72.102.41:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1400 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 284

v=0
o=7777 8000 8000 IN IP4 182.72.102.41
s=SIP Call
c=IN IP4 182.72.102.41
t=0 0
m=audio 5004 RTP/AVP 0 8 18 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
[May 6 03:47:50] --- (12 headers 14 lines) ---
[May 6 03:47:50] Found RTP audio format 0
[May 6 03:47:50] Found RTP audio format 8
[May 6 03:47:50] Found RTP audio format 18
[May 6 03:47:50] Found RTP audio format 101
[May 6 03:47:50] Found audio description format PCMU for ID 0
[May 6 03:47:50] Found audio description format PCMA for ID 8
[May 6 03:47:50] Found audio description format G729 for ID 18
[May 6 03:47:50] Found audio description format telephone-event for ID 101
[May 6 03:47:50] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
[May 6 03:47:50] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[May 6 03:47:50] Peer audio RTP is at port 182.72.102.41:5004
[May 6 03:47:50] list_route: hop: <sip:7777@182.72.102.41:5060>
[May 6 03:47:50] set_destination: Parsing <sip:7777@182.72.102.41:5060> for address/port to send to
[May 6 03:47:50] set_destination: set destination to 182.72.102.41:5060
[May 6 03:47:50] Transmitting (no NAT) to 182.72.102.41:5060:
ACK sip:7777@182.72.102.41:5060 SIP/2.0
Via: SIP/2.0/UDP 63.103.5.6:5060;branch=z9hG4bK7d05236c
Max-Forwards: 70
From: "MyTech" <sip:asterisk@63.103.5.6>;tag=as51948124
To: <sip:7777@182.72.102.41:5060>;tag=165290820
Contact: <sip:asterisk@63.103.5.6:5060>
Call-ID: 7a5811034ee44e071fcf457c1a3c8a5d@63.103.5.6:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.23.0-1_centos5
Content-Length: 0


---
[May 6 03:47:50] -- SIP/7777-000017c4 answered Local/063*103*005*006*7777@default-00001c1e;2
[May 6 03:47:50] > Channel Local/063*103*005*006*7777@default-00001c1e;1 was answered.
[May 6 03:47:50] -- Executing [8331*5777*Y5060347480000031039*7777*7777@default:1] Playback("Local/063*103*005*006*7777@default-00001c1e;1", "sip-silence") in new stack
[May 6 03:47:50] -- <Local/063*103*005*006*7777@default-00001c1e;1> Playing 'sip-silence.gsm' (language 'en')
[May 6 03:47:50] -- Executing [8331*5777*Y5060347480000031039*7777*7777@default:2] AGI("Local/063*103*005*006*7777@default-00001c1e;1", "agi-VDAD_RINGALL.agi,8331*5777*Y5060347480000031039*7777*7777") in new stack
[May 6 03:47:50] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_RINGALL.agi
[May 6 03:47:50] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 6 03:47:50] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 6 03:47:50] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 6 03:47:50] -- <Local/063*103*005*006*7777@default-00001c1e;1>AGI Script agi-VDAD_RINGALL.agi completed, returning 0
[May 6 03:47:50] -- Executing [8331*5777*Y5060347480000031039*7777*7777@default:3] AGI("Local/063*103*005*006*7777@default-00001c1e;1", "agi-VDAD_RINGALL.agi,8331*5777*Y5060347480000031039*7777*7777") in new stack
[May 6 03:47:50] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_RINGALL.agi
[May 6 03:47:50] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 6 03:47:50] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 6 03:47:50] Reliably Transmitting (NAT) to 208.79.14.65:5060:
OPTIONS sip:testterrycom SIP/2.0
Via: SIP/2.0/UDP 63.103.5.6:5060;branch=z9hG4bK09acff41;rport
Max-Forwards: 70
From: "asterisk" <sip:448004101075@63.103.5.6>;tag=as043d5a19
To: <sip:testterrycom>
Contact: <sip:448004101075@63.103.5.6:5060>
Call-ID: 7d0f0a81211592c437bee50f7d0efafb@63.103.5.6:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.23.0-1_centos5
Date: Tue, 05 May 2015 22:17:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[May 6 03:47:50] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 6 03:47:50] -- <Local/063*103*005*006*7777@default-00001c1e;1>AGI Script agi-VDAD_RINGALL.agi completed, returning 0
[May 6 03:47:50] -- Executing [8331*5777*Y5060347480000031039*7777*7777@default:4] AGI("Local/063*103*005*006*7777@default-00001c1e;1", "agi-VDAD_RINGALL.agi,8331*5777*Y5060347480000031039*7777*7777") in new stack
[May 6 03:47:50] -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-VDAD_RINGALL.agi
[May 6 03:47:50]
<--- SIP read from UDP:208.79.14.65:5060 --->
SIP/2.0 501 Unsupported Method
Via: SIP/2.0/UDP 63.103.5.6:5060;branch=z9hG4bK09acff41;rport=5060
To: <sip:testterrycom>;tag=8aa6723a
From: "asterisk" <sip:448004101075@63.103.5.6>;tag=as043d5a19
Call-ID: 7d0f0a81211592c437bee50f7d0efafb@63.103.5.6:5060
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
[May 6 03:47:50] --- (7 headers 0 lines) ---
[May 6 03:47:50] Really destroying SIP dialog '7d0f0a81211592c437bee50f7d0efafb@63.103.5.6:5060' Method: OPTIONS
[May 6 03:47:50] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 6 03:47:50] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 6 03:47:50] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[May 6 03:47:50] -- <Local/063*103*005*006*7777@default-00001c1e;1>AGI Script agi-VDAD_RINGALL.agi completed, returning 0
[May 6 03:47:50] -- Executing [8331*5777*Y5060347480000031039*7777*7777@default:5] Hangup("Local/063*103*005*006*7777@default-00001c1e;1", "") in new stack
[May 6 03:47:50] == Spawn extension (default, 8331*5777*Y5060347480000031039*7777*7777, 5) exited non-zero on 'Local/063*103*005*006*7777@default-00001c1e;1'
[May 6 03:47:50] -- Executing [h@default:1] AGI("Local/063*103*005*006*7777@default-00001c1e;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[May 6 03:47:50] -- <Local/063*103*005*006*7777@default-00001c1e;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[May 6 03:47:50] -- Executing [h@default:1] AGI("Local/063*103*005*006*7777@default-00001c1e;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----2-----0") in new stack
[May 6 03:47:50] -- <Local/063*103*005*006*7777@default-00001c1e;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----2-----0 completed, returning 0
[May 6 03:47:50] Scheduling destruction of SIP dialog '7a5811034ee44e071fcf457c1a3c8a5d@63.103.5.6:5060' in 19648 ms (Method: INVITE)
[May 6 03:47:50] set_destination: Parsing <sip:7777@182.72.102.41:5060> for address/port to send to
[May 6 03:47:50] set_destination: set destination to 182.72.102.41:5060
[May 6 03:47:50] Reliably Transmitting (no NAT) to 182.72.102.41:5060:
BYE sip:7777@182.72.102.41:5060 SIP/2.0
Via: SIP/2.0/UDP 63.103.5.6:5060;branch=z9hG4bK5c452018
Max-Forwards: 70
From: "MyTech" <sip:asterisk@63.103.5.6>;tag=as51948124
To: <sip:7777@182.72.102.41:5060>;tag=165290820
Call-ID: 7a5811034ee44e071fcf457c1a3c8a5d@63.103.5.6:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<------------->
[May 6 03:47:51] --- (7 headers 0 lines) ---
[May 6 03:47:51] Really destroying SIP dialog '1c7de0de275e681f6f2927c7606251e1@63.103.5.6:5060' Method: OPTIONS
[May 6 03:47:51] Retransmitting #1 (no NAT) to 182.72.102.41:5060:
BYE sip:7777@182.72.102.41:5060 SIP/2.0
Via: SIP/2.0/UDP 63.103.5.6:5060;branch=z9hG4bK5c452018
Max-Forwards: 70
From: "MyTech" <sip:asterisk@63.103.5.6>;tag=as51948124
To: <sip:7777@182.72.102.41:5060>;tag=165290820
Call-ID: 7a5811034ee44e071fcf457c1a3c8a5d@63.103.5.6:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.23.0-1_centos5
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[May 6 03:47:51]
<--- SIP read from UDP:182.72.102.41:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.103.5.6:5060;branch=z9hG4bK5c452018
From: "MyTech" <sip:asterisk@63.103.5.6>;tag=as51948124
To: <sip:7777@182.72.102.41:5060>;tag=165290820
Call-ID: 7a5811034ee44e071fcf457c1a3c8a5d@63.103.5.6:5060
CSeq: 103 BYE
Contact: <sip:7777@182.72.102.41:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1400 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

<------------->
[May 6 03:47:51] --- (11 headers 0 lines) ---
[May 6 03:47:51] Really destroying SIP dialog '7a5811034ee44e071fcf457c1a3c8a5d@63.103.5.6:5060' Method: INVITE
[May 6 03:47:51]
<--- SIP read from UDP:182.72.102.41:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 63.103.5.6:5060;branch=z9hG4bK5c452018
From: "MyTech" <sip:asterisk@63.103.5.6>;tag=as51948124
To: <sip:7777@182.72.102.41:5060>;tag=165290820
Call-ID: 7a5811034ee44e071fcf457c1a3c8a5d@63.103.5.6:5060
CSeq: 103 BYE
Contact: <sip:7777@182.72.102.41:5060>
Supported: replaces, path, timer
User-Agent: Grandstream GXP1400 1.0.6.11
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0
-------------------------------------------------------------------------------------------------------------------------

Thanks,
Terry
terry.sahil
 
Posts: 5
Joined: Thu Jun 13, 2013 6:50 am
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