vicidial.org

Posted: **Wed Apr 29, 2015 11:05 pm**

Hi

We have a strange issue on a fresh install of Vicibox 6.0.3 , SVN updated to latest as of yesterday. Manual dial is working fine. In ratio dial method calls are not transferred to agent, Agent can see calls in Queue . Call disconnects after 10 sec.

Thanks in advance...

agioutound logs
---------------------------------------

2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|Perl Environment Dump:
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|NORMAL-----LB
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|callerID changed: unknown
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|AGI Environment Dump:
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- accountcode =
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- arg_1 = NORMAL-----LB
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- callerid = unknown
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- calleridname = unknown
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- callingani2 = 0
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- callingpres = 67
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- callingtns = 0
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- callington = 0
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- channel = DAHDI/i1/09999888888-6
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- context = default
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- dnid = unknown
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- enhanced = 0.0
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- extension = 8368
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- language = en
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- priority = 3
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- rdnis = unknown
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- request = agi-VDAD_ALL_outbound.agi
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- threadid = 139910957102848
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- type = DAHDI
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- uniqueid = 1430361005.25
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi| -- version = 1.8.32.1-vici
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|AGI Variables: |1430361005.25|DAHDI/i1/09999888888-6|8368|DAHDI|unknown|unknown|3|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|+++++ VDAD START : |0|2015-04-30 08:00:16|1.8.32.1-vici|3|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_live_agents where callerid='unknown';|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where callerid='unknown' and status IN('LIVE','XFER');|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|-- VDAD : |0E0|update of vac table: unknown
|UPDATE vicidial_auto_calls set uniqueid='1430361005.25', channel='DAHDI/i1/09999888888-6',status='LIVE',stage='LIVE-0' where callerid='unknown' order by call_time desc limit 1;|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|-- NO VDAC FOUND!!!!!: unknown
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|-- NO VDM FOUND!!!!!!!!!!: unknown
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|1|VDAC-reinsert|INSERT INTO vicidial_auto_calls (server_ip,campaign_id,status,lead_id,uniqueid,callerid,channel,phone_code,phone_number,call_time,call_type,stage,queue_priority) values('192.168.13.20','','LIVE','0','1430361005.25','unknown','DAHDI/i1/09999888888-6','','','2015-04-30 08:00:16','OUT','LIVE-0.25','')|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|1|VDAC-reinsert|INSERT INTO vicidial_auto_calls (server_ip,campaign_id,status,lead_id,uniqueid,callerid,channel,phone_code,phone_number,call_time,call_type,stage,queue_priority) values('192.168.13.20','','LIVE','0','1430361005.25','unknown','DAHDI/i1/09999888888-6','','','2015-04-30 08:00:16','OUT','LIVE-0.25','')|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|
2015-04-30 08:00:16|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||DAHDI/i1/09999888888-6|unknown|1430361005.25|
2015-04-30 08:00:26|agi-VDAD_ALL_outbound.agi||UPDATE vicidial_log FORCE INDEX(lead_id) set status='DROP',end_epoch='1430361026',length_in_sec='9',term_reason='QUEUETIMEOUT' where lead_id = '0' and uniqueid LIKE "1430361005%";|
2015-04-30 08:00:26|agi-VDAD_ALL_outbound.agi|-- VDAD vicidial_log update: |0E0|1430361005.25
2015-04-30 08:00:26|agi-VDAD_ALL_outbound.agi||UPDATE vicidial_list set status='DROP' where lead_id = '0';|
2015-04-30 08:00:26|agi-VDAD_ALL_outbound.agi|-- VDAD vicidial_list update: |0E0|0
2015-04-30 08:00:26|agi-VDAD_ALL_outbound.agi|-- VDAD vac record deleted: |1||
2015-04-30 08:00:26|agi-VDAD_ALL_outbound.agi|-- VDCL call_hungup timout: |0|VH0430080026||DAHDI/i1/09999888888-6|insert to vicidial_manager
---------------------------------------------------------------------------------------------------------------------------

Asterisk CLI output

---------

[Apr 30 08:00:05] VERBOSE[36003] pbx.c: [Apr 30 08:00:05] -- Executing [929999888888@default:1] AGI("Local/929999888888@default-00000008;2", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 30 08:00:05] VERBOSE[36003] res_agi.c: [Apr 30 08:00:05] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=ffnd_AUTO))
[Apr 30 08:00:05] VERBOSE[36003] res_agi.c: [Apr 30 08:00:05] -- <Local/929999888888@default-00000008;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 30 08:00:05] VERBOSE[36003] pbx.c: [Apr 30 08:00:05] -- Executing [929999888888@default:2] Dial("Local/929999888888@default-00000008;2", "DAHDI/R0/09999888888,,To") in new stack
[Apr 30 08:00:05] DEBUG[36003] sig_pri.c: sig_pri_request 26
[Apr 30 08:00:05] DEBUG[36003] sig_pri.c: CALLER NAME: V4300800050000016484 NUM: 0000000000
[Apr 30 08:00:05] VERBOSE[36003] sig_pri.c: [Apr 30 08:00:05] -- Requested transfer capability: 0x00 - SPEECH
[Apr 30 08:00:05] VERBOSE[36003] app_dial.c: [Apr 30 08:00:05] -- Called DAHDI/R0/09999888888
[Apr 30 08:00:06] VERBOSE[36003] app_dial.c: [Apr 30 08:00:06] -- DAHDI/i1/09999888888-6 is proceeding passing it to Local/929999888888@default-00000008;2
[Apr 30 08:00:06] VERBOSE[36008] manager.c: [Apr 30 08:00:06] == Manager 'sendcron' logged on from 127.0.0.1
[Apr 30 08:00:06] VERBOSE[36008] manager.c: [Apr 30 08:00:06] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 30 08:00:06] VERBOSE[36003] app_dial.c: [Apr 30 08:00:06] -- DAHDI/i1/09999888888-6 is ringing
[Apr 30 08:00:06] VERBOSE[36003] app_dial.c: [Apr 30 08:00:06] -- DAHDI/i1/09999888888-6 is making progress passing it to Local/929999888888@default-00000008;2
[Apr 30 08:00:16] VERBOSE[36003] app_dial.c: [Apr 30 08:00:16] -- DAHDI/i1/09999888888-6 answered Local/929999888888@default-00000008;2
[Apr 30 08:00:16] VERBOSE[36002] pbx.c: [Apr 30 08:00:16] > Channel Local/929999888888@default-00000008;1 was answered.
[Apr 30 08:00:16] VERBOSE[36021] pbx.c: [Apr 30 08:00:16] -- Executing [8368@default:1] Playback("Local/929999888888@default-00000008;1", "sip-silence") in new stack
[Apr 30 08:00:16] VERBOSE[36021] file.c: [Apr 30 08:00:16] -- <Local/929999888888@default-00000008;1> Playing 'sip-silence.gsm' (language 'en')
[Apr 30 08:00:16] VERBOSE[36003] pbx.c: [Apr 30 08:00:16] -- Executing [h@default:1] AGI("Local/929999888888@default-00000008;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----11-----0") in new stack
[Apr 30 08:00:16] VERBOSE[36021] pbx.c: [Apr 30 08:00:16] -- Executing [8368@default:2] AGI("DAHDI/i1/09999888888-6", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 30 08:00:16] VERBOSE[36021] res_agi.c: [Apr 30 08:00:16] -- <DAHDI/i1/09999888888-6>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 30 08:00:16] VERBOSE[36021] pbx.c: [Apr 30 08:00:16] -- Executing [8368@default:3] AGI("DAHDI/i1/09999888888-6", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack
[Apr 30 08:00:16] VERBOSE[36021] res_agi.c: [Apr 30 08:00:16] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi
[Apr 30 08:00:17] VERBOSE[36002] manager.c: [Apr 30 08:00:17] == Manager 'sendcron' logged off from 127.0.0.1
[Apr 30 08:00:17] VERBOSE[36003] res_agi.c: [Apr 30 08:00:17] -- <Local/929999888888@default-00000008;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... --11-----0 completed, returning 0
[Apr 30 08:00:18] VERBOSE[36003] pbx.c: [Apr 30 08:00:18] == Spawn extension (default, 929999888888, 2) exited non-zero on 'Local/929999888888@default-00000008;2'
[Apr 30 08:00:26] VERBOSE[36021] res_agi.c: [Apr 30 08:00:26] -- <DAHDI/i1/09999888888-6>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 4
[Apr 30 08:00:26] VERBOSE[36021] pbx.c: [Apr 30 08:00:26] == Spawn extension (default, 8368, 3) exited non-zero on 'DAHDI/i1/09999888888-6'
[Apr 30 08:00:26] VERBOSE[36021] pbx.c: [Apr 30 08:00:26] -- Executing [h@default:1] AGI("DAHDI/i1/09999888888-6", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Apr 30 08:00:26] VERBOSE[36021] res_agi.c: [Apr 30 08:00:26] -- <DAHDI/i1/09999888888-6>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Apr 30 08:00:26] VERBOSE[36021] chan_dahdi.c: [Apr 30 08:00:26] -- Hungup 'DAHDI/i1/09999888888-6'
[Apr 30 08:00:26] VERBOSE[36038] manager.c: [Apr 30 08:00:26] == Manager 'sendcron' logged on from 127.0.0.1

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

Extension.conf
-------------------------------------------------------------
exten => _92[7-9]X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _92[7-9]X.,2,Dial(${ogtrunk}/0${EXTEN:2},,To)
exten => _92[7-9]X.,3,Hangup()

exten => _92[1-6]X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _92[1-6]X.,2,Dial(${ogtrunk}/${EXTEN:2},,To)
exten => _92[1-6]X.,3,Hangup()

exten => _920[1-9]X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _920[1-9]X.,2,Dial(${ogtrunk}/${EXTEN:2},,To)
exten => _920[1-9]X.,3,Hangup()
--------------------------------------------------------------------------------------------

Lvish

----
HP ML10, Quad Core Xeon , 4 GB ram, 2 Port PRI Allo make.,
VICIBOX 6.0.3,
VERSION: 2.12-482a
BUILD: 150422-1953
Asterisk version -- 1.8.32.1-vici
SVN --2312
DB schema --1407

Posted: **Sat May 02, 2015 11:59 pm**

Hi

Anybody faced this issue.. i tried recompling asterisk,installed 1.8.32.0-rc1, changed dialplan , o flag is in dial plan... still no luck.

thanks in advance.

lvish

Posted: **Thu May 07, 2015 8:49 pm**

There seems to be many 'unknown' values, but besides the point. Have you checked what the calls are being dispositioned as?

Posted: **Fri May 08, 2015 12:19 am**

Hi

Thanks for response, call gets disposed as NA or Drop. Looks like a roadblocker bug ..

regds

lvish

Posted: **Tue Jul 07, 2015 2:10 pm**

Where you able to solve the problem. I have the same exact here. The weird thing is thaht our systrem was working fine. We had to reinstall due to a server failure. After reinstallation manual calls work fine but autodial does not pass calls to agents.

Posted: **Wed Jul 08, 2015 9:52 pm**

crangel wrote:Where you able to solve the problem. I have the same exact here. The weird thing is thaht our systrem was working fine. We had to reinstall due to a server failure. After reinstallation manual calls work fine but autodial does not pass calls to agents.

1) Welcome to the Party! 8-)

2) As you are obviously new here, I have some suggestions to help us all help you:

When you post, please post your entire configuration including (but not limited to) your installation method and vicidial version with build.

This IS a requirement for posting along with reading the stickies (at the top of each forum) and the manager's manual (available on EFLO.net, both free and paid versions)

You should also post: Asterisk version, telephony hardware (model number is helpful here), cluster information if you have one, and whether any other software is installed in the box. If your installation method is "from scratch" you must post your operating system and should also post the .iso version from which you installed your original operating system. If your installation is "Hosted" list the site name of the host.

If this is a "Cloud" or "Virtual" server, please note the technology involved along with the version of that techology (ie: VMware Server Version 2.0.2). If it is not, merely stating the Motherboard model # and CPU would be helpful.

Similar to This:

Vicibox X.X from .iso | Vicidial X.X.X-XXX Build XXXXXX-XXXX | Asterisk X.X.X | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | Intel DG35EC | Core2Quad Q6600

3) Did you bring the old DB into the new server? If so ... did the IP change? (and did you run the ip updater script?) did you modify the asterisk version in admin->servers? And did you UPGRADE the DB, since the Vicidial Version undoubtedly changed between your two installs?

4) CLI Output from a good and a bad call (test calls, not in production, not mixed in with 3000 lines of unrelated code ...) would be useful as often the "same problem" is merely the "same symptom" with a completely different cause.

5) Happy Hunting! 8-)

Posted: **Thu Jul 23, 2015 11:28 pm**

I appear to be in the same boat. Manual works, Autodial doesn't.

Temporarily disabled firewall no luck.

vicibox 6.0.3, upgraded to most recent SVN as of a week ago.

Vicibox 6.0.3 from .iso | Vicidial 2.12-496a Build 150710-1120 | SVN 2347 | Asterisk 1.8.32.3-vici | Single Server | No Digium/Sangoma Hardware | No Extra Software After Installation | 2x Xeon E5-2630 | 32GB ECC DDR4

Asterisk CLI:

Code: Select all: [Jul 24 14:02:53] -- Executing [9610269253124@default:1] AGI("Local/9610269253124@default-00000044;2", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 14:02:53] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=PDBUSNSW)) [Jul 24 14:02:53] -- <Local/9610269253124@default-00000044;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 14:02:53] -- Executing [9610269253124@default:2] Dial("Local/9610269253124@default-00000044;2", "SIP/mytel1/0269253124,,tTor") in new stack [Jul 24 14:02:53] == Using SIP RTP CoS mark 5 [Jul 24 14:02:53] -- Called SIP/mytel1/0269253124 [Jul 24 14:02:53] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 14:02:53] -- Executing [9610298327710@default:1] AGI("Local/9610298327710@default-00000045;2", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 14:02:53] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=PDBUSNSW)) [Jul 24 14:02:53] -- <Local/9610298327710@default-00000045;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 14:02:53] -- Executing [9610298327710@default:2] Dial("Local/9610298327710@default-00000045;2", "SIP/mytel1/0298327710,,tTor") in new stack [Jul 24 14:02:53] == Using SIP RTP CoS mark 5 [Jul 24 14:02:53] -- Called SIP/mytel1/0298327710 [Jul 24 14:02:54] -- SIP/mytel1-0000002b is ringing [Jul 24 14:02:54] -- SIP/mytel1-0000002a is making progress passing it to Local/9610269253124@default-00000044;2 [Jul 24 14:02:58] -- SIP/mytel1-0000002b is making progress passing it to Local/9610298327710@default-00000045;2 [Jul 24 14:02:58] -- SIP/mytel1-0000002b answered Local/9610298327710@default-00000045;2 [Jul 24 14:02:58] > Channel Local/9610298327710@default-00000045;1 was answered. [Jul 24 14:02:58] -- Executing [8368@default:1] Playback("Local/9610298327710@default-00000045;1", "sip-silence") in new stack [Jul 24 14:02:58] -- <Local/9610298327710@default-00000045;1> Playing 'sip-silence.gsm' (language 'en') [Jul 24 14:02:58] -- Executing [8368@default:2] AGI("Local/9610298327710@default-00000045;1", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 14:02:58] -- <Local/9610298327710@default-00000045;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 14:02:58] -- Executing [8368@default:3] AGI("Local/9610298327710@default-00000045;1", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jul 24 14:02:58] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jul 24 14:02:58] -- Executing [h@default:1] AGI("Local/9610298327710@default-00000045;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----5-----0") in new stack [Jul 24 14:02:59] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 14:02:59] -- <Local/9610298327710@default-00000045;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----5-----0 completed, returning 0 [Jul 24 14:02:59] == Spawn extension (default, 9610298327710, 2) exited non-zero on 'Local/9610298327710@default-00000045;2' [Jul 24 14:02:59] -- <SIP/mytel1-0000002b>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0 [Jul 24 14:02:59] -- Executing [8368@default:4] AGI("SIP/mytel1-0000002b", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jul 24 14:02:59] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jul 24 14:03:01] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 14:03:01] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 14:03:01] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 14:03:05] -- <SIP/mytel1-0000002b>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 4 [Jul 24 14:03:05] == Spawn extension (default, 8368, 4) exited non-zero on 'SIP/mytel1-0000002b' [Jul 24 14:03:05] -- Executing [h@default:1] AGI("SIP/mytel1-0000002b", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack [Jul 24 14:03:05] -- <SIP/mytel1-0000002b>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0 [Jul 24 14:03:06] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 14:03:06] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 14:03:12] -- SIP/mytel1-0000002a answered Local/9610269253124@default-00000044;2 [Jul 24 14:03:12] > Channel Local/9610269253124@default-00000044;1 was answered. [Jul 24 14:03:12] -- Executing [8368@default:1] Playback("Local/9610269253124@default-00000044;1", "sip-silence") in new stack [Jul 24 14:03:12] -- <Local/9610269253124@default-00000044;1> Playing 'sip-silence.gsm' (language 'en') [Jul 24 14:03:12] -- Executing [h@default:1] AGI("Local/9610269253124@default-00000044;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----19-----0") in new stack [Jul 24 14:03:12] -- Executing [8368@default:2] AGI("SIP/mytel1-0000002a", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 14:03:12] -- <SIP/mytel1-0000002a>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 14:03:12] -- Executing [8368@default:3] AGI("SIP/mytel1-0000002a", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jul 24 14:03:12] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jul 24 14:03:13] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 14:03:13] -- <Local/9610269253124@default-00000044;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----19-----0 completed, returning 0 [Jul 24 14:03:13] == Spawn extension (default, 9610269253124, 2) exited non-zero on 'Local/9610269253124@default-00000044;2' [Jul 24 14:03:22] -- <SIP/mytel1-0000002a>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 4 [Jul 24 14:03:22] == Spawn extension (default, 8368, 3) exited non-zero on 'SIP/mytel1-0000002a' [Jul 24 14:03:22] -- Executing [h@default:1] AGI("SIP/mytel1-0000002a", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack [Jul 24 14:03:22] -- <SIP/mytel1-0000002a>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0

Posted: **Thu Jul 23, 2015 11:37 pm**

AGI("Local/9610298327710@default-0000004

Local. Doesn't have sound.

1) You said you upgraded ... was it working before the upgrade?

2) Manual works ... do you have two-way sound when you dial manually?

3) When you say "firewall disabled" you left off a very important piece of information: Does this server have a private IP or a public IP? Or both? And if you say it has a public IP, please be sure you mean that it does not have a router between it and that public IP (the public IP in that case is actually on the router, not the server!).

4) Did you change the server's IP at any time? What is the externip value?

Posted: **Thu Jul 23, 2015 11:40 pm**

1) sorry, scratch install from 6.0.3, then upgraded straight away as per install instructions.

2) Manual definitely has sound. Both ways.

3) Private only, But wanted to rule it out anyway.

4) never changed the IP beyond installation

Bonus output:

Code: Select all: Connected to Asterisk 1.8.32.3-vici currently running on LargeDialer (pid = 11681) Verbosity is at least 21 LargeDialer*CLI> dialplan show 8368@default [ Context 'default' created by 'pbx_config' ] '8368' => 1. Playback(sip-silence) [pbx_config] 2. AGI(agi://127.0.0.1:4577/call_log) [pbx_config] 3. AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB) [pbx_config] 4. AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB) [pbx_config] 5. Hangup() [pbx_config] -= 1 extension (5 priorities) in 1 context. =- LargeDialer*CLI> Disconnected from Asterisk server LargeDialer:~ # screen -list There are screens on: 131823.ASTVDremote (Detached) 20528.ASTVDadapt (Detached) 67379.ASTemail (Detached) 12306.ASTVDadFILL (Detached) 12303.ASTfastlog (Detached) 12294.ASTVDauto (Detached) 12291.ASTlisten (Detached) 12288.ASTsend (Detached) 12285.ASTupdate (Detached) 11675.asterisk (Detached) 11670.astshell20150715103836 (Detached) 11 Sockets in /var/run/screens/S-root. LargeDialer:~ # dahdi_cfg -v DAHDI Tools Version - 2.10.1 DAHDI Version: 2.10.1 Echo Canceller(s): Configuration ====================== 0 channels to configure.

Posted: **Fri Jul 24, 2015 12:46 am**

3) Private only, But wanted to rule it out anyway.

In which case you did not disable the firewall, since this server will always have a firewall. You could consider putting the server in the DMZ, but technically that won't even be proof. LOL. You could attempt forwarding port 5060(UDP) and port range 10000-25000 (UDP) and cross your fingers.

You did not specify if it was working before the upgrade. I'm not prepared to assume that it was (and if it wasn't, you likely have a bad install ...)

What about that externip value? Is it set to the outside IP for the router? (IE: the public IP?)

Why are you scratch installing instead of using Vicibox? Martyrdom ...?

Posted: **Fri Jul 24, 2015 12:52 am**

OH sorry.

We terminate to another internal PBX provided by our VOIP Provider. We've operated in this manner for several years.
All communication stays internal. Sorry if that was confusing earlier.

And yes, it was vicibox. 6.0.3. When I said scratch I meant it was a fresh install.

Some Failed Autodial:

Code: Select all: [Jul 24 15:45:07] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 15:45:16] > Refreshing DNS lookups. [Jul 24 15:45:39] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 15:45:39] -- Executing [9610296237755@default:1] AGI("Local/9610296237755@default-00000004;2", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 15:45:39] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=PDBUSNSW)) [Jul 24 15:45:39] -- <Local/9610296237755@default-00000004;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 15:45:39] -- Executing [9610296237755@default:2] Dial("Local/9610296237755@default-00000004;2", "SIP/mytel1/0296237755,,tTor") in new stack [Jul 24 15:45:39] == Using SIP RTP CoS mark 5 [Jul 24 15:45:39] -- Called SIP/mytel1/0296237755 [Jul 24 15:45:39] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 15:45:39] -- Executing [9610263310211@default:1] AGI("Local/9610263310211@default-00000005;2", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 15:45:39] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=PDBUSNSW)) [Jul 24 15:45:39] -- <Local/9610263310211@default-00000005;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 15:45:39] -- Executing [9610263310211@default:2] Dial("Local/9610263310211@default-00000005;2", "SIP/mytel1/0263310211,,tTor") in new stack [Jul 24 15:45:39] == Using SIP RTP CoS mark 5 [Jul 24 15:45:39] -- Called SIP/mytel1/0263310211 [Jul 24 15:45:40] -- SIP/mytel1-00000001 is making progress passing it to Local/9610296237755@default-00000004;2 [Jul 24 15:45:41] -- SIP/mytel1-00000002 is making progress passing it to Local/9610263310211@default-00000005;2 [Jul 24 15:45:47] -- SIP/mytel1-00000002 answered Local/9610263310211@default-00000005;2 [Jul 24 15:45:47] > Channel Local/9610263310211@default-00000005;1 was answered. [Jul 24 15:45:47] -- Executing [8368@default:1] Playback("Local/9610263310211@default-00000005;1", "sip-silence") in new stack [Jul 24 15:45:47] -- <Local/9610263310211@default-00000005;1> Playing 'sip-silence.gsm' (language 'en') [Jul 24 15:45:47] -- Executing [h@default:1] AGI("Local/9610263310211@default-00000005;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----8-----0") in new stack [Jul 24 15:45:47] -- Executing [8368@default:2] AGI("SIP/mytel1-00000002", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 15:45:47] -- <SIP/mytel1-00000002>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 15:45:47] -- Executing [8368@default:3] AGI("SIP/mytel1-00000002", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jul 24 15:45:47] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jul 24 15:45:47] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 15:45:48] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 15:45:48] -- <Local/9610263310211@default-00000005;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----8-----0 completed, returning 0 [Jul 24 15:45:48] == Spawn extension (default, 9610263310211, 2) exited non-zero on 'Local/9610263310211@default-00000005;2' [Jul 24 15:45:51] -- SIP/mytel1-00000001 answered Local/9610296237755@default-00000004;2 [Jul 24 15:45:51] > Channel Local/9610296237755@default-00000004;1 was answered. [Jul 24 15:45:51] -- Executing [8368@default:1] Playback("Local/9610296237755@default-00000004;1", "sip-silence") in new stack [Jul 24 15:45:51] -- <Local/9610296237755@default-00000004;1> Playing 'sip-silence.gsm' (language 'en') [Jul 24 15:45:51] -- Executing [h@default:1] AGI("Local/9610296237755@default-00000004;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----12-----0") in new stack [Jul 24 15:45:51] -- Executing [8368@default:2] AGI("SIP/mytel1-00000001", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 15:45:51] -- <SIP/mytel1-00000001>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 15:45:51] -- Executing [8368@default:3] AGI("SIP/mytel1-00000001", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jul 24 15:45:51] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jul 24 15:45:51] -- <SIP/mytel1-00000001>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0 [Jul 24 15:45:51] -- Executing [8368@default:4] AGI("SIP/mytel1-00000001", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jul 24 15:45:51] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jul 24 15:45:51] -- <SIP/mytel1-00000001>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0 [Jul 24 15:45:51] -- Executing [8368@default:5] Hangup("SIP/mytel1-00000001", "") in new stack [Jul 24 15:45:51] == Spawn extension (default, 8368, 5) exited non-zero on 'SIP/mytel1-00000001' [Jul 24 15:45:51] -- Executing [h@default:1] AGI("SIP/mytel1-00000001", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack [Jul 24 15:45:51] -- <SIP/mytel1-00000001>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0 [Jul 24 15:45:52] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 15:45:52] -- <Local/9610296237755@default-00000004;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----12-----0 completed, returning 0 [Jul 24 15:45:52] == Spawn extension (default, 9610296237755, 2) exited non-zero on 'Local/9610296237755@default-00000004;2' [Jul 24 15:45:57] -- <SIP/mytel1-00000002>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 4 [Jul 24 15:45:57] == Spawn extension (default, 8368, 3) exited non-zero on 'SIP/mytel1-00000002' [Jul 24 15:45:57] -- Executing [h@default:1] AGI("SIP/mytel1-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack [Jul 24 15:45:57] -- <SIP/mytel1-00000002>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0 [Jul 24 15:45:57] == Manager 'sendcron' logged on from 127.0.0.1

A successful Manual Dial:

Code: Select all: [Jul 24 15:47:55] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 15:47:55] -- Executing [8600051@default:1] MeetMe("Local/8600051@default-00000006;2", "8600051,F") in new stack [Jul 24 15:47:55] > Channel Local/8600051@default-00000006;1 was answered. [Jul 24 15:47:55] -- Executing [9610478805437@default:1] AGI("Local/8600051@default-00000006;1", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 15:47:55] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=PDBUSNSW)) [Jul 24 15:47:55] -- <Local/8600051@default-00000006;1>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 15:47:55] -- Executing [9610478805437@default:2] Dial("Local/8600051@default-00000006;1", "SIP/mytel1/0478805437,,tTor") in new stack [Jul 24 15:47:55] == Using SIP RTP CoS mark 5 [Jul 24 15:47:55] -- Called SIP/mytel1/0478805437 [Jul 24 15:47:56] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 15:48:00] -- SIP/mytel1-00000003 is making progress passing it to Local/8600051@default-00000006;1 [Jul 24 15:48:01] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 15:48:01] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 15:48:01] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 15:48:03] -- SIP/mytel1-00000003 answered Local/8600051@default-00000006;1 [Jul 24 15:48:06] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 15:48:06] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 15:48:14] -- Executing [h@default:1] AGI("Local/8600051@default-00000006;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----19-----11") in new stack [Jul 24 15:48:14] -- <Local/8600051@default-00000006;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----19-----11 completed, returning 0 [Jul 24 15:48:14] == Spawn extension (default, 9610478805437, 2) exited non-zero on 'Local/8600051@default-00000006;1' [Jul 24 15:48:14] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-00000006;2' [Jul 24 15:48:14] -- Executing [h@default:1] AGI("Local/8600051@default-00000006;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack [Jul 24 15:48:14] -- <Local/8600051@default-00000006;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0 [Jul 24 15:48:16] == Manager 'sendcron' logged on from 127.0.0.1

Posted: **Fri Jul 24, 2015 1:08 am**

You did not specify if it was working before the upgrade. I'm not prepared to assume that it was (and if it wasn't, you likely have a bad install ...)

When you upgraded, did you tell it to load the sample conf files? If not, reinstall and tell it to do that this time (and reboot). Back up the conf files in /etc/asterisk before you do this, of course.

Posted: **Fri Jul 24, 2015 1:25 am**

I followed both the manager manual and install manual quite closely. I don't recall the sample config step. (would you be able to point me in the direction of the required steps?)

Is there any quick fix, or is it a case of bulldozing and starting over?

Posted: **Fri Jul 24, 2015 1:33 am**

cd /usr/src/astguiclient/trunk
perl install.pl --help

there's an option for sample conf's ... activate it. then during the installation process, it will already be checked and you don't have to change it.

Reinstalling Vicidial has no effect, unless you've customized the scripts or configuration files (which is not recommended) since all it does is copy the scripts to where they belong. If you reinstall, all you're doing is copying those same files back to the same place. But those .conf files often have changes that are necessary during an upgrade. Without the new versions, odd things can happen ...

Posted: **Fri Jul 24, 2015 1:51 am**

No such luck.

After doing install -> rebuild conf -> reboot, still getting the same error:

Code: Select all: [Jul 24 16:48:07] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 16:48:07] -- Executing [9610260412171@default:1] AGI("Local/9610260412171@default-00000001;2", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 16:48:07] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=PDBUSNSW)) [Jul 24 16:48:07] -- <Local/9610260412171@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 16:48:07] -- Executing [9610260412171@default:2] Dial("Local/9610260412171@default-00000001;2", "SIP/mytel1/0260412171,,tTor") in new stack [Jul 24 16:48:07] == Using SIP RTP CoS mark 5 [Jul 24 16:48:07] -- Called SIP/mytel1/0260412171 [Jul 24 16:48:08] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 16:48:08] -- Executing [9610247223951@default:1] AGI("Local/9610247223951@default-00000002;2", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 16:48:08] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=PDBUSNSW)) [Jul 24 16:48:08] -- <Local/9610247223951@default-00000002;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 16:48:08] -- Executing [9610247223951@default:2] Dial("Local/9610247223951@default-00000002;2", "SIP/mytel1/0247223951,,tTor") in new stack [Jul 24 16:48:08] == Using SIP RTP CoS mark 5 [Jul 24 16:48:08] -- Called SIP/mytel1/0247223951 [Jul 24 16:48:09] -- SIP/mytel1-00000001 is making progress passing it to Local/9610260412171@default-00000001;2 [Jul 24 16:48:13] -- SIP/mytel1-00000002 is making progress passing it to Local/9610247223951@default-00000002;2 [Jul 24 16:48:16] -- SIP/mytel1-00000001 answered Local/9610260412171@default-00000001;2 [Jul 24 16:48:16] > Channel Local/9610260412171@default-00000001;1 was answered. [Jul 24 16:48:16] -- Executing [8368@default:1] Playback("Local/9610260412171@default-00000001;1", "sip-silence") in new stack [Jul 24 16:48:16] -- <Local/9610260412171@default-00000001;1> Playing 'sip-silence.gsm' (language 'en') [Jul 24 16:48:16] -- Executing [h@default:1] AGI("Local/9610260412171@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----9-----0") in new stack [Jul 24 16:48:16] -- Executing [8368@default:2] AGI("SIP/mytel1-00000001", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 16:48:16] -- <SIP/mytel1-00000001>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 16:48:16] -- Executing [8368@default:3] AGI("SIP/mytel1-00000001", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jul 24 16:48:16] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jul 24 16:48:17] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 16:48:17] -- <Local/9610260412171@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----9-----0 completed, returning 0 [Jul 24 16:48:17] == Spawn extension (default, 9610260412171, 2) exited non-zero on 'Local/9610260412171@default-00000001;2' [Jul 24 16:48:22] -- SIP/mytel1-00000002 answered Local/9610247223951@default-00000002;2 [Jul 24 16:48:22] > Channel Local/9610247223951@default-00000002;1 was answered. [Jul 24 16:48:22] -- Executing [8368@default:1] Playback("Local/9610247223951@default-00000002;1", "sip-silence") in new stack [Jul 24 16:48:22] -- <Local/9610247223951@default-00000002;1> Playing 'sip-silence.gsm' (language 'en') [Jul 24 16:48:22] -- Executing [h@default:1] AGI("Local/9610247223951@default-00000002;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----14-----0") in new stack [Jul 24 16:48:22] -- Executing [8368@default:2] AGI("SIP/mytel1-00000002", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 16:48:22] -- <SIP/mytel1-00000002>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 16:48:22] -- Executing [8368@default:3] AGI("SIP/mytel1-00000002", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jul 24 16:48:22] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jul 24 16:48:22] -- <SIP/mytel1-00000002>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0 [Jul 24 16:48:22] -- Executing [8368@default:4] AGI("SIP/mytel1-00000002", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jul 24 16:48:22] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jul 24 16:48:22] -- <SIP/mytel1-00000002>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0 [Jul 24 16:48:22] -- Executing [8368@default:5] Hangup("SIP/mytel1-00000002", "") in new stack [Jul 24 16:48:22] == Spawn extension (default, 8368, 5) exited non-zero on 'SIP/mytel1-00000002' [Jul 24 16:48:22] -- Executing [h@default:1] AGI("SIP/mytel1-00000002", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack [Jul 24 16:48:22] -- <SIP/mytel1-00000002>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0 [Jul 24 16:48:23] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 16:48:23] -- <Local/9610247223951@default-00000002;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----14-----0 completed, returning 0 [Jul 24 16:48:23] == Spawn extension (default, 9610247223951, 2) exited non-zero on 'Local/9610247223951@default-00000002;2' [Jul 24 16:48:26] -- <SIP/mytel1-00000001>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 4 [Jul 24 16:48:26] == Spawn extension (default, 8368, 3) exited non-zero on 'SIP/mytel1-00000001' [Jul 24 16:48:26] -- Executing [h@default:1] AGI("SIP/mytel1-00000001", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack [Jul 24 16:48:26] -- <SIP/mytel1-00000001>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0 [Jul 24 16:48:26] == Manager 'sendcron' logged on from 127.0.0.1

My highly untrained eye only spots this as a difference from a successful call:

[Jul 24 16:48:23] == Spawn extension (default, 9610247223951, 2) exited non-zero on 'Local/9610247223951@default-00000002;2'
[Jul 24 16:48:26] -- <SIP/mytel1-00000001>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 4
[Jul 24 16:48:26] == Spawn extension (default, 8368, 3) exited non-zero on 'SIP/mytel1-00000001'

Posted: **Fri Jul 24, 2015 2:05 am**

You'll eventually find another difference before that line. Callerid, connection method, codec, something.

Perhaps you should consider sip debug and check the call handshake up through the connection. You may have a problem with the call you can't see in the dialplan log.

Posted: **Fri Jul 24, 2015 2:24 am**

Hahah. After installing vicidial/vicibox three times, you'd think I wouldn't be this dumb. I absolutely cannot see the problem.

Manual dialing is fine, I just can't get the autodialer to pass it on to the agent properly.

Just sits in queue until it drops.

A line picked up about halfway in this output:

Code: Select all: LargeDialer*CLI> LargeDialer*CLI> LargeDialer*CLI> [Jul 24 17:14:32] Really destroying SIP dialog '22a86b1741a578374b86997566fdea19@10.10.10.1' Method: OPTIONS [Jul 24 17:14:32] Really destroying SIP dialog '44665261107d77f33a08b8d563d4e3e1@10.10.10.1' Method: OPTIONS LargeDialer*CLI> LargeDialer*CLI> LargeDialer*CLI> LargeDialer*CLI> LargeDialer*CLI> [Jul 24 17:14:33] Really destroying SIP dialog '5c8d48c3496d4147426dfeeb6ef40b79@10.10.10.1' Method: NOTIFY [Jul 24 17:14:33] Really destroying SIP dialog '4afbc3a77bf6d80779075a6e230a6fac@10.10.10.1' Method: NOTIFY [Jul 24 17:14:39] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 17:14:39] -- Executing [9610296250635@default:1] AGI("Local/9610296250635@default-00000013;2", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 17:14:39] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=PDBUSNSW)) [Jul 24 17:14:39] -- <Local/9610296250635@default-00000013;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 17:14:39] -- Executing [9610296250635@default:2] Dial("Local/9610296250635@default-00000013;2", "SIP/mytel1/0296250635,,tTor") in new stack [Jul 24 17:14:39] == Using SIP RTP CoS mark 5 [Jul 24 17:14:39] Audio is at 14960 [Jul 24 17:14:39] Adding codec 0x4 (ulaw) to SDP [Jul 24 17:14:39] Adding non-codec 0x1 (telephone-event) to SDP [Jul 24 17:14:39] Reliably Transmitting (NAT) to 10.10.10.1:5060: INVITE sip:0296250635@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK687d0e96;rport Max-Forwards: 70 From: "V7241714390000000040" <sip:996@10.10.10.11>;tag=as4edef9d7 To: <sip:0296250635@10.10.10.1> Contact: <sip:996@10.10.10.11:5060> Call-ID: 0ba9136a61784b6446bc0c714cb6d8fa@10.10.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.32.3-vici Date: Fri, 24 Jul 2015 07:14:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "V7241714390000000040" <sip:0000000000@10.10.10.11>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 238 v=0 o=root 181451682 181451682 IN IP4 10.10.10.11 s=Asterisk PBX 1.8.32.3-vici c=IN IP4 10.10.10.11 t=0 0 m=audio 14960 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jul 24 17:14:39] -- Called SIP/mytel1/0296250635 [Jul 24 17:14:39] <--- SIP read from UDP:10.10.10.1:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK687d0e96;received=10.10.10.11;rport=5060 From: "V7241714390000000040" <sip:996@10.10.10.11>;tag=as4edef9d7 To: <sip:0296250635@10.10.10.1>;tag=as20cd3dac Call-ID: 0ba9136a61784b6446bc0c714cb6d8fa@10.10.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2ae6fce7" Content-Length: 0 <-------------> [Jul 24 17:14:39] --- (11 headers 0 lines) --- [Jul 24 17:14:39] Transmitting (NAT) to 10.10.10.1:5060: ACK sip:0296250635@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK687d0e96;rport Max-Forwards: 70 From: "V7241714390000000040" <sip:996@10.10.10.11>;tag=as4edef9d7 To: <sip:0296250635@10.10.10.1>;tag=as20cd3dac Contact: <sip:996@10.10.10.11:5060> Call-ID: 0ba9136a61784b6446bc0c714cb6d8fa@10.10.10.11:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.32.3-vici Content-Length: 0 --- [Jul 24 17:14:39] Audio is at 14960 [Jul 24 17:14:39] Adding codec 0x4 (ulaw) to SDP [Jul 24 17:14:39] Adding non-codec 0x1 (telephone-event) to SDP [Jul 24 17:14:39] Reliably Transmitting (NAT) to 10.10.10.1:5060: INVITE sip:0296250635@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK5ec2abd6;rport Max-Forwards: 70 From: "V7241714390000000040" <sip:996@10.10.10.11>;tag=as4edef9d7 To: <sip:0296250635@10.10.10.1> Contact: <sip:996@10.10.10.11:5060> Call-ID: 0ba9136a61784b6446bc0c714cb6d8fa@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.32.3-vici Proxy-Authorization: Digest username="996", realm="asterisk", algorithm=MD5, uri="sip:0296250635@10.10.10.1", nonce="2ae6fce7", response="f77d10d8e9e6909aafeefcd91ee42989" Date: Fri, 24 Jul 2015 07:14:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "V7241714390000000040" <sip:0000000000@10.10.10.11>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 238 v=0 o=root 181451682 181451683 IN IP4 10.10.10.11 s=Asterisk PBX 1.8.32.3-vici c=IN IP4 10.10.10.11 t=0 0 m=audio 14960 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jul 24 17:14:39] <--- SIP read from UDP:10.10.10.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK5ec2abd6;received=10.10.10.11;rport=5060 From: "V7241714390000000040" <sip:996@10.10.10.11>;tag=as4edef9d7 To: <sip:0296250635@10.10.10.1> Call-ID: 0ba9136a61784b6446bc0c714cb6d8fa@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:0296250635@10.10.10.1> Content-Length: 0 <-------------> [Jul 24 17:14:39] --- (11 headers 0 lines) --- [Jul 24 17:14:39] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 17:14:39] -- Executing [9610298330121@default:1] AGI("Local/9610298330121@default-00000014;2", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 17:14:39] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=PDBUSNSW)) [Jul 24 17:14:39] -- <Local/9610298330121@default-00000014;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 17:14:39] -- Executing [9610298330121@default:2] Dial("Local/9610298330121@default-00000014;2", "SIP/mytel1/0298330121,,tTor") in new stack [Jul 24 17:14:39] == Using SIP RTP CoS mark 5 [Jul 24 17:14:39] Audio is at 15848 [Jul 24 17:14:39] Adding codec 0x4 (ulaw) to SDP [Jul 24 17:14:39] Adding non-codec 0x1 (telephone-event) to SDP [Jul 24 17:14:39] Reliably Transmitting (NAT) to 10.10.10.1:5060: INVITE sip:0298330121@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK40198c03;rport Max-Forwards: 70 From: "V7241714390000000041" <sip:996@10.10.10.11>;tag=as180ac6d5 To: <sip:0298330121@10.10.10.1> Contact: <sip:996@10.10.10.11:5060> Call-ID: 58ed9e974d7af78529dd36336e5f5987@10.10.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.32.3-vici Date: Fri, 24 Jul 2015 07:14:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "V7241714390000000041" <sip:0000000000@10.10.10.11>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 240 v=0 o=root 1961391401 1961391401 IN IP4 10.10.10.11 s=Asterisk PBX 1.8.32.3-vici c=IN IP4 10.10.10.11 t=0 0 m=audio 15848 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jul 24 17:14:39] -- Called SIP/mytel1/0298330121 [Jul 24 17:14:39] <--- SIP read from UDP:10.10.10.1:5060 ---> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK40198c03;received=10.10.10.11;rport=5060 From: "V7241714390000000041" <sip:996@10.10.10.11>;tag=as180ac6d5 To: <sip:0298330121@10.10.10.1>;tag=as0a63e5aa Call-ID: 58ed9e974d7af78529dd36336e5f5987@10.10.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="6057078d" Content-Length: 0 <-------------> [Jul 24 17:14:39] --- (11 headers 0 lines) --- [Jul 24 17:14:39] Transmitting (NAT) to 10.10.10.1:5060: ACK sip:0298330121@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK40198c03;rport Max-Forwards: 70 From: "V7241714390000000041" <sip:996@10.10.10.11>;tag=as180ac6d5 To: <sip:0298330121@10.10.10.1>;tag=as0a63e5aa Contact: <sip:996@10.10.10.11:5060> Call-ID: 58ed9e974d7af78529dd36336e5f5987@10.10.10.11:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.32.3-vici Content-Length: 0 --- [Jul 24 17:14:39] Audio is at 15848 [Jul 24 17:14:39] Adding codec 0x4 (ulaw) to SDP [Jul 24 17:14:39] Adding non-codec 0x1 (telephone-event) to SDP [Jul 24 17:14:39] Reliably Transmitting (NAT) to 10.10.10.1:5060: INVITE sip:0298330121@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK52555ce8;rport Max-Forwards: 70 From: "V7241714390000000041" <sip:996@10.10.10.11>;tag=as180ac6d5 To: <sip:0298330121@10.10.10.1> Contact: <sip:996@10.10.10.11:5060> Call-ID: 58ed9e974d7af78529dd36336e5f5987@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.32.3-vici Proxy-Authorization: Digest username="996", realm="asterisk", algorithm=MD5, uri="sip:0298330121@10.10.10.1", nonce="6057078d", response="3dd222dd88519161b9cbff48d8464bd3" Date: Fri, 24 Jul 2015 07:14:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "V7241714390000000041" <sip:0000000000@10.10.10.11>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 240 v=0 o=root 1961391401 1961391402 IN IP4 10.10.10.11 s=Asterisk PBX 1.8.32.3-vici c=IN IP4 10.10.10.11 t=0 0 m=audio 15848 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv --- [Jul 24 17:14:39] <--- SIP read from UDP:10.10.10.1:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK52555ce8;received=10.10.10.11;rport=5060 From: "V7241714390000000041" <sip:996@10.10.10.11>;tag=as180ac6d5 To: <sip:0298330121@10.10.10.1> Call-ID: 58ed9e974d7af78529dd36336e5f5987@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:0298330121@10.10.10.1> Content-Length: 0 <-------------> [Jul 24 17:14:39] --- (11 headers 0 lines) --- [Jul 24 17:14:40] <--- SIP read from UDP:10.10.10.1:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK52555ce8;received=10.10.10.11;rport=5060 From: "V7241714390000000041" <sip:996@10.10.10.11>;tag=as180ac6d5 To: <sip:0298330121@10.10.10.1>;tag=as0921ebad Call-ID: 58ed9e974d7af78529dd36336e5f5987@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:0298330121@10.10.10.1> Content-Type: application/sdp Content-Length: 207 v=0 o=root 6229 6229 IN IP4 10.10.10.1 s=session c=IN IP4 10.10.10.1 t=0 0 m=audio 13048 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Jul 24 17:14:40] --- (12 headers 11 lines) --- [Jul 24 17:14:40] list_route: hop: <sip:0298330121@10.10.10.1> [Jul 24 17:14:40] Found RTP audio format 0 [Jul 24 17:14:40] Found RTP audio format 101 [Jul 24 17:14:40] Found audio description format PCMU for ID 0 [Jul 24 17:14:40] Found audio description format telephone-event for ID 101 [Jul 24 17:14:40] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jul 24 17:14:40] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jul 24 17:14:40] Peer audio RTP is at port 10.10.10.1:13048 [Jul 24 17:14:40] -- SIP/mytel1-00000011 is making progress passing it to Local/9610298330121@default-00000014;2 [Jul 24 17:14:41] <--- SIP read from UDP:10.10.10.1:5060 ---> SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK5ec2abd6;received=10.10.10.11;rport=5060 From: "V7241714390000000040" <sip:996@10.10.10.11>;tag=as4edef9d7 To: <sip:0296250635@10.10.10.1>;tag=as25686bc8 Call-ID: 0ba9136a61784b6446bc0c714cb6d8fa@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:0296250635@10.10.10.1> Content-Type: application/sdp Content-Length: 207 v=0 o=root 6229 6229 IN IP4 10.10.10.1 s=session c=IN IP4 10.10.10.1 t=0 0 m=audio 18562 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Jul 24 17:14:41] --- (12 headers 11 lines) --- [Jul 24 17:14:41] list_route: hop: <sip:0296250635@10.10.10.1> [Jul 24 17:14:41] Found RTP audio format 0 [Jul 24 17:14:41] Found RTP audio format 101 [Jul 24 17:14:41] Found audio description format PCMU for ID 0 [Jul 24 17:14:41] Found audio description format telephone-event for ID 101 [Jul 24 17:14:41] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jul 24 17:14:41] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jul 24 17:14:41] Peer audio RTP is at port 10.10.10.1:18562 [Jul 24 17:14:41] -- SIP/mytel1-00000010 is making progress passing it to Local/9610296250635@default-00000013;2 [Jul 24 17:14:45] Reliably Transmitting (NAT) to 10.10.10.1:5060: OPTIONS sip:10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK20f9cb7c;rport Max-Forwards: 70 From: "asterisk" <sip:996@10.10.10.11>;tag=as1c375a6a To: <sip:10.10.10.1> Contact: <sip:996@10.10.10.11:5060> Call-ID: 37404bd427eaa9c0641c757d4f76dccf@10.10.10.11:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.32.3-vici Date: Fri, 24 Jul 2015 07:14:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jul 24 17:14:45] <--- SIP read from UDP:10.10.10.1:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK20f9cb7c;received=10.10.10.11;rport=5060 From: "asterisk" <sip:996@10.10.10.11>;tag=as1c375a6a To: <sip:10.10.10.1>;tag=as7b55da70 Call-ID: 37404bd427eaa9c0641c757d4f76dccf@10.10.10.11:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Accept: application/sdp Content-Length: 0 <-------------> [Jul 24 17:14:45] --- (11 headers 0 lines) --- [Jul 24 17:14:45] Really destroying SIP dialog '37404bd427eaa9c0641c757d4f76dccf@10.10.10.11:5060' Method: OPTIONS [Jul 24 17:14:45] Reliably Transmitting (NAT) to 10.10.10.1:5060: OPTIONS sip:10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK194d34b6;rport Max-Forwards: 70 From: "asterisk" <sip:997@10.10.10.11>;tag=as1f447ac5 To: <sip:10.10.10.1> Contact: <sip:997@10.10.10.11:5060> Call-ID: 1111841d20b0abbf6ae4e4d97c14196d@10.10.10.11:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.32.3-vici Date: Fri, 24 Jul 2015 07:14:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 --- [Jul 24 17:14:45] <--- SIP read from UDP:10.10.10.1:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK194d34b6;received=10.10.10.11;rport=5060 From: "asterisk" <sip:997@10.10.10.11>;tag=as1f447ac5 To: <sip:10.10.10.1>;tag=as18369170 Call-ID: 1111841d20b0abbf6ae4e4d97c14196d@10.10.10.11:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Accept: application/sdp Content-Length: 0 <-------------> [Jul 24 17:14:45] --- (11 headers 0 lines) --- [Jul 24 17:14:45] Really destroying SIP dialog '1111841d20b0abbf6ae4e4d97c14196d@10.10.10.11:5060' Method: OPTIONS [Jul 24 17:14:46] <--- SIP read from UDP:10.10.10.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK5ec2abd6;received=10.10.10.11;rport=5060 From: "V7241714390000000040" <sip:996@10.10.10.11>;tag=as4edef9d7 To: <sip:0296250635@10.10.10.1>;tag=as25686bc8 Call-ID: 0ba9136a61784b6446bc0c714cb6d8fa@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:0296250635@10.10.10.1> Content-Type: application/sdp Content-Length: 207 v=0 o=root 6229 6230 IN IP4 10.10.10.1 s=session c=IN IP4 10.10.10.1 t=0 0 m=audio 18562 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Jul 24 17:14:46] --- (12 headers 11 lines) --- [Jul 24 17:14:46] Found RTP audio format 0 [Jul 24 17:14:46] Found RTP audio format 101 [Jul 24 17:14:46] Found audio description format PCMU for ID 0 [Jul 24 17:14:46] Found audio description format telephone-event for ID 101 [Jul 24 17:14:46] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jul 24 17:14:46] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jul 24 17:14:46] Peer audio RTP is at port 10.10.10.1:18562 [Jul 24 17:14:46] list_route: hop: <sip:0296250635@10.10.10.1> [Jul 24 17:14:46] set_destination: Parsing <sip:0296250635@10.10.10.1> for address/port to send to [Jul 24 17:14:46] set_destination: set destination to 10.10.10.1:5060 [Jul 24 17:14:46] Transmitting (NAT) to 10.10.10.1:5060: ACK sip:0296250635@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK182d37ac;rport Max-Forwards: 70 From: "V7241714390000000040" <sip:996@10.10.10.11>;tag=as4edef9d7 To: <sip:0296250635@10.10.10.1>;tag=as25686bc8 Contact: <sip:996@10.10.10.11:5060> Call-ID: 0ba9136a61784b6446bc0c714cb6d8fa@10.10.10.11:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.32.3-vici Content-Length: 0 --- [Jul 24 17:14:46] -- SIP/mytel1-00000010 answered Local/9610296250635@default-00000013;2 [Jul 24 17:14:46] > Channel Local/9610296250635@default-00000013;1 was answered. [Jul 24 17:14:46] -- Executing [8368@default:1] Playback("Local/9610296250635@default-00000013;1", "sip-silence") in new stack [Jul 24 17:14:46] -- <Local/9610296250635@default-00000013;1> Playing 'sip-silence.gsm' (language 'en') [Jul 24 17:14:46] -- Executing [h@default:1] AGI("Local/9610296250635@default-00000013;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----7-----0") in new stack [Jul 24 17:14:46] -- Executing [8368@default:2] AGI("SIP/mytel1-00000010", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 17:14:46] -- <SIP/mytel1-00000010>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 17:14:46] -- Executing [8368@default:3] AGI("SIP/mytel1-00000010", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jul 24 17:14:46] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jul 24 17:14:47] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 17:14:47] -- <Local/9610296250635@default-00000013;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----7-----0 completed, returning 0 [Jul 24 17:14:47] == Spawn extension (default, 9610296250635, 2) exited non-zero on 'Local/9610296250635@default-00000013;2' [Jul 24 17:14:47] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 17:14:56] <--- SIP read from UDP:10.10.10.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK52555ce8;received=10.10.10.11;rport=5060 From: "V7241714390000000041" <sip:996@10.10.10.11>;tag=as180ac6d5 To: <sip:0298330121@10.10.10.1>;tag=as0921ebad Call-ID: 58ed9e974d7af78529dd36336e5f5987@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:0298330121@10.10.10.1> Content-Type: application/sdp Content-Length: 207 v=0 o=root 6229 6230 IN IP4 10.10.10.1 s=session c=IN IP4 10.10.10.1 t=0 0 m=audio 13048 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv <-------------> [Jul 24 17:14:56] --- (12 headers 11 lines) --- [Jul 24 17:14:56] Found RTP audio format 0 [Jul 24 17:14:56] Found RTP audio format 101 [Jul 24 17:14:56] Found audio description format PCMU for ID 0 [Jul 24 17:14:56] Found audio description format telephone-event for ID 101 [Jul 24 17:14:56] Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw) [Jul 24 17:14:56] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) [Jul 24 17:14:56] Peer audio RTP is at port 10.10.10.1:13048 [Jul 24 17:14:56] list_route: hop: <sip:0298330121@10.10.10.1> [Jul 24 17:14:56] set_destination: Parsing <sip:0298330121@10.10.10.1> for address/port to send to [Jul 24 17:14:56] set_destination: set destination to 10.10.10.1:5060 [Jul 24 17:14:56] Transmitting (NAT) to 10.10.10.1:5060: ACK sip:0298330121@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK7737fb79;rport Max-Forwards: 70 From: "V7241714390000000041" <sip:996@10.10.10.11>;tag=as180ac6d5 To: <sip:0298330121@10.10.10.1>;tag=as0921ebad Contact: <sip:996@10.10.10.11:5060> Call-ID: 58ed9e974d7af78529dd36336e5f5987@10.10.10.11:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.32.3-vici Content-Length: 0 --- [Jul 24 17:14:56] -- SIP/mytel1-00000011 answered Local/9610298330121@default-00000014;2 [Jul 24 17:14:56] > Channel Local/9610298330121@default-00000014;1 was answered. [Jul 24 17:14:56] -- Executing [8368@default:1] Playback("Local/9610298330121@default-00000014;1", "sip-silence") in new stack [Jul 24 17:14:56] -- <Local/9610298330121@default-00000014;1> Playing 'sip-silence.gsm' (language 'en') [Jul 24 17:14:56] -- Executing [h@default:1] AGI("Local/9610298330121@default-00000014;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----17-----0") in new stack [Jul 24 17:14:56] -- Executing [8368@default:2] AGI("SIP/mytel1-00000011", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 17:14:56] -- <SIP/mytel1-00000011>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 17:14:56] -- Executing [8368@default:3] AGI("SIP/mytel1-00000011", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jul 24 17:14:56] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jul 24 17:14:56] -- <SIP/mytel1-00000011>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0 [Jul 24 17:14:56] -- Executing [8368@default:4] AGI("SIP/mytel1-00000011", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jul 24 17:14:56] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jul 24 17:14:56] -- <SIP/mytel1-00000011>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 0 [Jul 24 17:14:56] -- Executing [8368@default:5] Hangup("SIP/mytel1-00000011", "") in new stack [Jul 24 17:14:56] == Spawn extension (default, 8368, 5) exited non-zero on 'SIP/mytel1-00000011' [Jul 24 17:14:56] -- Executing [h@default:1] AGI("SIP/mytel1-00000011", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack [Jul 24 17:14:56] -- <SIP/mytel1-00000011>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0 [Jul 24 17:14:56] Scheduling destruction of SIP dialog '58ed9e974d7af78529dd36336e5f5987@10.10.10.11:5060' in 6400 ms (Method: INVITE) [Jul 24 17:14:56] set_destination: Parsing <sip:0298330121@10.10.10.1> for address/port to send to [Jul 24 17:14:56] set_destination: set destination to 10.10.10.1:5060 [Jul 24 17:14:56] Reliably Transmitting (NAT) to 10.10.10.1:5060: BYE sip:0298330121@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK377e1bae;rport Max-Forwards: 70 From: "V7241714390000000041" <sip:996@10.10.10.11>;tag=as180ac6d5 To: <sip:0298330121@10.10.10.1>;tag=as0921ebad Call-ID: 58ed9e974d7af78529dd36336e5f5987@10.10.10.11:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 1.8.32.3-vici Proxy-Authorization: Digest username="996", realm="asterisk", algorithm=MD5, uri="sip:0298330121@10.10.10.1", nonce="6057078d", response="76ec535beb30169fd9125034476dc11c" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jul 24 17:14:56] <--- SIP read from UDP:10.10.10.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK377e1bae;received=10.10.10.11;rport=5060 From: "V7241714390000000041" <sip:996@10.10.10.11>;tag=as180ac6d5 To: <sip:0298330121@10.10.10.1>;tag=as0921ebad Call-ID: 58ed9e974d7af78529dd36336e5f5987@10.10.10.11:5060 CSeq: 104 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> [Jul 24 17:14:56] --- (10 headers 0 lines) --- [Jul 24 17:14:56] Really destroying SIP dialog '58ed9e974d7af78529dd36336e5f5987@10.10.10.11:5060' Method: INVITE [Jul 24 17:14:56] -- <SIP/mytel1-00000010>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 4 [Jul 24 17:14:56] == Spawn extension (default, 8368, 3) exited non-zero on 'SIP/mytel1-00000010' [Jul 24 17:14:56] -- Executing [h@default:1] AGI("SIP/mytel1-00000010", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack [Jul 24 17:14:56] -- <SIP/mytel1-00000010>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0 [Jul 24 17:14:56] Scheduling destruction of SIP dialog '0ba9136a61784b6446bc0c714cb6d8fa@10.10.10.11:5060' in 6400 ms (Method: INVITE) [Jul 24 17:14:56] set_destination: Parsing <sip:0296250635@10.10.10.1> for address/port to send to [Jul 24 17:14:56] set_destination: set destination to 10.10.10.1:5060 [Jul 24 17:14:56] Reliably Transmitting (NAT) to 10.10.10.1:5060: BYE sip:0296250635@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK59a0a59d;rport Max-Forwards: 70 From: "V7241714390000000040" <sip:996@10.10.10.11>;tag=as4edef9d7 To: <sip:0296250635@10.10.10.1>;tag=as25686bc8 Call-ID: 0ba9136a61784b6446bc0c714cb6d8fa@10.10.10.11:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 1.8.32.3-vici Proxy-Authorization: Digest username="996", realm="asterisk", algorithm=MD5, uri="sip:0296250635@10.10.10.1", nonce="2ae6fce7", response="3c92f802060349d07b1d33bd943ebdca" X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 --- [Jul 24 17:14:56] <--- SIP read from UDP:10.10.10.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK59a0a59d;received=10.10.10.11;rport=5060 From: "V7241714390000000040" <sip:996@10.10.10.11>;tag=as4edef9d7 To: <sip:0296250635@10.10.10.1>;tag=as25686bc8 Call-ID: 0ba9136a61784b6446bc0c714cb6d8fa@10.10.10.11:5060 CSeq: 104 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> [Jul 24 17:14:56] --- (10 headers 0 lines) --- [Jul 24 17:14:56] Really destroying SIP dialog '0ba9136a61784b6446bc0c714cb6d8fa@10.10.10.11:5060' Method: INVITE [Jul 24 17:14:56] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 17:14:57] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 17:14:57] -- <Local/9610298330121@default-00000014;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----17-----0 completed, returning 0 [Jul 24 17:14:57] == Spawn extension (default, 9610298330121, 2) exited non-zero on 'Local/9610298330121@default-00000014;2' [Jul 24 17:14:57] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 17:14:58] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 17:14:58] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 17:14:58] -- Executing [55558600051@default:1] MeetMeAdmin("Local/55558600051@default-00000015;2", "8600051,K") in new stack [Jul 24 17:14:58] -- Executing [55558600051@default:2] Hangup("Local/55558600051@default-00000015;2", "") in new stack [Jul 24 17:14:58] == Spawn extension (default, 55558600051, 2) exited non-zero on 'Local/55558600051@default-00000015;2' [Jul 24 17:14:58] -- Executing [h@default:1] AGI("Local/55558600051@default-00000015;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack [Jul 24 17:14:58] -- <SIP/mytel2-0000000f> Playing 'conf-kicked.gsm' (language 'en') [Jul 24 17:14:58] -- <Local/55558600051@default-00000015;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0 [Jul 24 17:14:59] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 17:14:59] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 17:15:00] <--- SIP read from UDP:10.10.10.1:5060 ---> OPTIONS sip:s@10.10.10.11:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK195c53e5;rport From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as60b8fd82 To: <sip:s@10.10.10.11:5060> Contact: <sip:asterisk@10.10.10.1> Call-ID: 1058648821d1a007593e662070f47bb2@10.10.10.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 24 Jul 2015 07:14:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> [Jul 24 17:15:00] --- (13 headers 0 lines) --- [Jul 24 17:15:00] Looking for s in trunkinbound (domain 10.10.10.11) [Jul 24 17:15:00] <--- Transmitting (NAT) to 10.10.10.1:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK195c53e5;received=10.10.10.1;rport=5060 From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as60b8fd82 To: <sip:s@10.10.10.11:5060>;tag=as7dd0ac56 Call-ID: 1058648821d1a007593e662070f47bb2@10.10.10.1 CSeq: 102 OPTIONS Server: Asterisk PBX 1.8.32.3-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jul 24 17:15:00] Scheduling destruction of SIP dialog '1058648821d1a007593e662070f47bb2@10.10.10.1' in 32000 ms (Method: OPTIONS) [Jul 24 17:15:00] <--- SIP read from UDP:10.10.10.1:5060 ---> OPTIONS sip:s@10.10.10.11:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK196e8b0e;rport From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as02a06b45 To: <sip:s@10.10.10.11:5060> Contact: <sip:asterisk@10.10.10.1> Call-ID: 76815d0d20eb77b37c724dd174e6f086@10.10.10.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 24 Jul 2015 07:14:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> [Jul 24 17:15:00] --- (13 headers 0 lines) --- [Jul 24 17:15:00] Looking for s in trunkinbound (domain 10.10.10.11) [Jul 24 17:15:00] <--- Transmitting (NAT) to 10.10.10.1:5060 ---> SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK196e8b0e;received=10.10.10.1;rport=5060 From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as02a06b45 To: <sip:s@10.10.10.11:5060>;tag=as4b17297b Call-ID: 76815d0d20eb77b37c724dd174e6f086@10.10.10.1 CSeq: 102 OPTIONS Server: Asterisk PBX 1.8.32.3-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 <------------> [Jul 24 17:15:00] Scheduling destruction of SIP dialog '76815d0d20eb77b37c724dd174e6f086@10.10.10.1' in 32000 ms (Method: OPTIONS) [Jul 24 17:15:00] -- Hungup 'DAHDI/pseudo-1389872196' [Jul 24 17:15:00] -- Executing [8600051@default:2] Hangup("SIP/mytel2-0000000f", "") in new stack [Jul 24 17:15:00] == Spawn extension (default, 8600051, 2) exited non-zero on 'SIP/mytel2-0000000f' [Jul 24 17:15:00] -- Executing [h@default:1] AGI("SIP/mytel2-0000000f", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack [Jul 24 17:15:00] -- <SIP/mytel2-0000000f>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0 [Jul 24 17:15:00] Scheduling destruction of SIP dialog '46ca083c46ae1281445b9432460ba3bf@10.10.10.11:5060' in 6400 ms (Method: INVITE) [Jul 24 17:15:00] set_destination: Parsing <sip:500@10.10.10.1> for address/port to send to [Jul 24 17:15:00] set_destination: set destination to 10.10.10.1:5060 [Jul 24 17:15:00] Reliably Transmitting (NAT) to 10.10.10.1:5060: BYE sip:500@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK21ef8895;rport Max-Forwards: 70 From: "S1507241713138600051" <sip:997@10.10.10.11>;tag=as3a1df9d2 To: <sip:500@10.10.10.1>;tag=as4e077afc Call-ID: 46ca083c46ae1281445b9432460ba3bf@10.10.10.11:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 1.8.32.3-vici Proxy-Authorization: Digest username="997", realm="asterisk", algorithm=MD5, uri="sip:500@10.10.10.1", nonce="652531c7", response="3e22b05712e32c12af82d4748f72d549" X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0 --- [Jul 24 17:15:00] <--- SIP read from UDP:10.10.10.1:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK21ef8895;received=10.10.10.11;rport=5060 From: "S1507241713138600051" <sip:997@10.10.10.11>;tag=as3a1df9d2 To: <sip:500@10.10.10.1>;tag=as4e077afc Call-ID: 46ca083c46ae1281445b9432460ba3bf@10.10.10.11:5060 CSeq: 104 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 <-------------> [Jul 24 17:15:00] --- (10 headers 0 lines) --- [Jul 24 17:15:00] Really destroying SIP dialog '46ca083c46ae1281445b9432460ba3bf@10.10.10.11:5060' Method: INVITE [Jul 24 17:15:01] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 17:15:01] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 17:15:01] == Manager 'sendcron' logged on from 127.0.0.1 LargeDialer*CLI> sip set debug off SIP Debugging Disabled [Jul 24 17:15:06] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 17:15:06] == Manager 'sendcron' logged off from 127.0.0.1 LargeDialer*CLI>

I might insert my mobile into a list and see what happens on the other end.

EDIT:

Inserted my mobile. No call came through. I wonder what was queuing then

ngrep-sip:

Code: Select all: LargeDialer:~ # ngrep-sip interface: any filter: (ip) and ( port 5060 ) # U 2015/07/24 17:34:45.220557 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK199db5fa;received=10.10.10.11;rport=5060 From: "asterisk" <sip:996@10.10.10.11>;tag=as1de15d26 To: <sip:10.10.10.1>;tag=as173d28b1 Call-ID: 13dde31d440fa7c805c422655ac885eb@10.10.10.11:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Accept: application/sdp Content-Length: 0 # U 2015/07/24 17:34:45.297916 10.10.10.11:5060 -> 10.10.10.1:5060 OPTIONS sip:10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK6ec69dfa;rport Max-Forwards: 70 From: "asterisk" <sip:997@10.10.10.11>;tag=as17401118 To: <sip:10.10.10.1> Contact: <sip:997@10.10.10.11:5060> Call-ID: 1d8fa52d5d46ed9b5e19420864f8f9d8@10.10.10.11:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.32.3-vici Date: Fri, 24 Jul 2015 07:34:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 # U 2015/07/24 17:34:45.298281 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK6ec69dfa;received=10.10.10.11;rport=5060 From: "asterisk" <sip:997@10.10.10.11>;tag=as17401118 To: <sip:10.10.10.1>;tag=as676b9f71 Call-ID: 1d8fa52d5d46ed9b5e19420864f8f9d8@10.10.10.11:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Accept: application/sdp Content-Length: 0 # U 2015/07/24 17:34:47.167786 10.10.10.11:5060 -> 10.10.10.1:5060 INVITE sip:0478805437@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK1546282a;rport Max-Forwards: 70 From: "V7241734470000005735" <sip:996@10.10.10.11>;tag=as5c51cb4b To: <sip:0478805437@10.10.10.1> Contact: <sip:996@10.10.10.11:5060> Call-ID: 0e77b373360b2a0a3ded61872d4e7186@10.10.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.32.3-vici Date: Fri, 24 Jul 2015 07:34:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "V7241734470000005735" <sip:0000000000@10.10.10.11>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 238 v=0 o=root 284327589 284327589 IN IP4 10.10.10.11 s=Asterisk PBX 1.8.32.3-vici c=IN IP4 10.10.10.11 t=0 0 m=audio 12082 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv # U 2015/07/24 17:34:47.168176 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK1546282a;received=10.10.10.11;rport=5060 From: "V7241734470000005735" <sip:996@10.10.10.11>;tag=as5c51cb4b To: <sip:0478805437@10.10.10.1>;tag=as260b1aa5 Call-ID: 0e77b373360b2a0a3ded61872d4e7186@10.10.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="72e556e3" Content-Length: 0 # U 2015/07/24 17:34:47.168277 10.10.10.11:5060 -> 10.10.10.1:5060 ACK sip:0478805437@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK1546282a;rport Max-Forwards: 70 From: "V7241734470000005735" <sip:996@10.10.10.11>;tag=as5c51cb4b To: <sip:0478805437@10.10.10.1>;tag=as260b1aa5 Contact: <sip:996@10.10.10.11:5060> Call-ID: 0e77b373360b2a0a3ded61872d4e7186@10.10.10.11:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.32.3-vici Content-Length: 0 # U 2015/07/24 17:34:47.168356 10.10.10.11:5060 -> 10.10.10.1:5060 INVITE sip:0478805437@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK2b3ed61e;rport Max-Forwards: 70 From: "V7241734470000005735" <sip:996@10.10.10.11>;tag=as5c51cb4b To: <sip:0478805437@10.10.10.1> Contact: <sip:996@10.10.10.11:5060> Call-ID: 0e77b373360b2a0a3ded61872d4e7186@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.32.3-vici Proxy-Authorization: Digest username="996", realm="asterisk", algorithm=MD5, uri="sip:0478805437@10.10.10.1", nonce="72e556e3", response="9c445b0ee040cebb23aa5ca7261c6365" Date: Fri, 24 Jul 2015 07:34:47 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "V7241734470000005735" <sip:0000000000@10.10.10.11>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 238 v=0 o=root 284327589 284327590 IN IP4 10.10.10.11 s=Asterisk PBX 1.8.32.3-vici c=IN IP4 10.10.10.11 t=0 0 m=audio 12082 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv # U 2015/07/24 17:34:47.168830 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK2b3ed61e;received=10.10.10.11;rport=5060 From: "V7241734470000005735" <sip:996@10.10.10.11>;tag=as5c51cb4b To: <sip:0478805437@10.10.10.1> Call-ID: 0e77b373360b2a0a3ded61872d4e7186@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:0478805437@10.10.10.1> Content-Length: 0 # U 2015/07/24 17:35:00.691619 10.10.10.11:5060 -> 10.10.10.1:5060 REGISTER sip:10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK64413d66;rport Max-Forwards: 70 From: <sip:997@10.10.10.1>;tag=as244fa405 To: <sip:997@10.10.10.1> Call-ID: 115250f524a977f4203363e920a00117@10.10.10.11 CSeq: 158 REGISTER User-Agent: Asterisk PBX 1.8.32.3-vici Authorization: Digest username="997", realm="asterisk", algorithm=MD5, uri="sip:10.10.10.1", nonce="0c482b0f", response="e041645bbc35a25e0dac4613400567ff" Expires: 120 Contact: <sip:s@10.10.10.11:5060> Content-Length: 0 # U 2015/07/24 17:35:00.691887 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK64413d66;received=10.10.10.11;rport=5060 From: <sip:997@10.10.10.1>;tag=as244fa405 To: <sip:997@10.10.10.1> Call-ID: 115250f524a977f4203363e920a00117@10.10.10.11 CSeq: 158 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 # U 2015/07/24 17:35:00.691895 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK64413d66;received=10.10.10.11;rport=5060 From: <sip:997@10.10.10.1>;tag=as244fa405 To: <sip:997@10.10.10.1>;tag=as0829646c Call-ID: 115250f524a977f4203363e920a00117@10.10.10.11 CSeq: 158 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="20f88ad0" Content-Length: 0 # U 2015/07/24 17:35:00.692089 10.10.10.11:5060 -> 10.10.10.1:5060 REGISTER sip:10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK0d7b2e45;rport Max-Forwards: 70 From: <sip:997@10.10.10.1>;tag=as244fa405 To: <sip:997@10.10.10.1> Call-ID: 115250f524a977f4203363e920a00117@10.10.10.11 CSeq: 159 REGISTER User-Agent: Asterisk PBX 1.8.32.3-vici Authorization: Digest username="997", realm="asterisk", algorithm=MD5, uri="sip:10.10.10.1", nonce="20f88ad0", response="9ae5c54494fcb9cfa4ad38f6ff3dc083" Expires: 120 Contact: <sip:s@10.10.10.11:5060> Content-Length: 0 # U 2015/07/24 17:35:00.692361 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK0d7b2e45;received=10.10.10.11;rport=5060 From: <sip:997@10.10.10.1>;tag=as244fa405 To: <sip:997@10.10.10.1> Call-ID: 115250f524a977f4203363e920a00117@10.10.10.11 CSeq: 159 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 # U 2015/07/24 17:35:00.694633 10.10.10.11:5060 -> 10.10.10.1:5060 REGISTER sip:10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK6f02f883;rport Max-Forwards: 70 From: <sip:996@10.10.10.1>;tag=as4b3a9e1a To: <sip:996@10.10.10.1> Call-ID: 5a25cc536dd0b4bc0fa92792675bd638@10.10.10.11 CSeq: 158 REGISTER User-Agent: Asterisk PBX 1.8.32.3-vici Authorization: Digest username="996", realm="asterisk", algorithm=MD5, uri="sip:10.10.10.1", nonce="6bbf9b1e", response="d32973440ab77bae776db834395e074a" Expires: 120 Contact: <sip:s@10.10.10.11:5060> Content-Length: 0 # U 2015/07/24 17:35:00.704039 10.10.10.1:5060 -> 10.10.10.11:5060 OPTIONS sip:s@10.10.10.11:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK004fe179;rport From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as5f433c2a To: <sip:s@10.10.10.11:5060> Contact: <sip:asterisk@10.10.10.1> Call-ID: 5eb4ab6d05d6861f51151ad627dc0903@10.10.10.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 24 Jul 2015 07:34:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 # U 2015/07/24 17:35:00.704056 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK0d7b2e45;received=10.10.10.11;rport=5060 From: <sip:997@10.10.10.1>;tag=as244fa405 To: <sip:997@10.10.10.1>;tag=as0829646c Call-ID: 115250f524a977f4203363e920a00117@10.10.10.11 CSeq: 159 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Expires: 120 Contact: <sip:s@10.10.10.11:5060>;expires=120 Date: Fri, 24 Jul 2015 07:34:53 GMT Content-Length: 0 # U 2015/07/24 17:35:00.704212 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK6f02f883;received=10.10.10.11;rport=5060 From: <sip:996@10.10.10.1>;tag=as4b3a9e1a To: <sip:996@10.10.10.1> Call-ID: 5a25cc536dd0b4bc0fa92792675bd638@10.10.10.11 CSeq: 158 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 # U 2015/07/24 17:35:00.704231 10.10.10.11:5060 -> 10.10.10.1:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK004fe179;received=10.10.10.1;rport=5060 From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as5f433c2a To: <sip:s@10.10.10.11:5060>;tag=as49e390b4 Call-ID: 5eb4ab6d05d6861f51151ad627dc0903@10.10.10.1 CSeq: 102 OPTIONS Server: Asterisk PBX 1.8.32.3-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 # U 2015/07/24 17:35:00.704246 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK6f02f883;received=10.10.10.11;rport=5060 From: <sip:996@10.10.10.1>;tag=as4b3a9e1a To: <sip:996@10.10.10.1>;tag=as3f460b82 Call-ID: 5a25cc536dd0b4bc0fa92792675bd638@10.10.10.11 CSeq: 158 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="61a2ed72" Content-Length: 0 # U 2015/07/24 17:35:00.704538 10.10.10.11:5060 -> 10.10.10.1:5060 REGISTER sip:10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK70cdb777;rport Max-Forwards: 70 From: <sip:996@10.10.10.1>;tag=as4b3a9e1a To: <sip:996@10.10.10.1> Call-ID: 5a25cc536dd0b4bc0fa92792675bd638@10.10.10.11 CSeq: 159 REGISTER User-Agent: Asterisk PBX 1.8.32.3-vici Authorization: Digest username="996", realm="asterisk", algorithm=MD5, uri="sip:10.10.10.1", nonce="61a2ed72", response="8ca0b56d67bfb52388a5a03453d99c28" Expires: 120 Contact: <sip:s@10.10.10.11:5060> Content-Length: 0 # U 2015/07/24 17:35:00.704752 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK70cdb777;received=10.10.10.11;rport=5060 From: <sip:996@10.10.10.1>;tag=as4b3a9e1a To: <sip:996@10.10.10.1> Call-ID: 5a25cc536dd0b4bc0fa92792675bd638@10.10.10.11 CSeq: 159 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 # U 2015/07/24 17:35:00.707978 10.10.10.1:5060 -> 10.10.10.11:5060 OPTIONS sip:s@10.10.10.11:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK51d53222;rport From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as1804524d To: <sip:s@10.10.10.11:5060> Contact: <sip:asterisk@10.10.10.1> Call-ID: 786ef76e64389bbe6e7dd1be5aa8dc45@10.10.10.1 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 24 Jul 2015 07:34:53 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 # U 2015/07/24 17:35:00.708003 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK70cdb777;received=10.10.10.11;rport=5060 From: <sip:996@10.10.10.1>;tag=as4b3a9e1a To: <sip:996@10.10.10.1>;tag=as3f460b82 Call-ID: 5a25cc536dd0b4bc0fa92792675bd638@10.10.10.11 CSeq: 159 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Expires: 120 Contact: <sip:s@10.10.10.11:5060>;expires=120 Date: Fri, 24 Jul 2015 07:34:53 GMT Content-Length: 0 # U 2015/07/24 17:35:00.708093 10.10.10.11:5060 -> 10.10.10.1:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK51d53222;received=10.10.10.1;rport=5060 From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as1804524d To: <sip:s@10.10.10.11:5060>;tag=as38cc7aa7 Call-ID: 786ef76e64389bbe6e7dd1be5aa8dc45@10.10.10.1 CSeq: 102 OPTIONS Server: Asterisk PBX 1.8.32.3-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Accept: application/sdp Content-Length: 0 # U 2015/07/24 17:35:06.811397 10.10.10.1:5060 -> 10.10.10.11:5060 NOTIFY sip:s@10.10.10.11:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK762f8611;rport From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as3ee6db46 To: <sip:s@10.10.10.11:5060> Contact: <sip:asterisk@10.10.10.1> Call-ID: 05cdf5b671961f597fe697d248b0dda7@10.10.10.1 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 84 Messages-Waiting: no Message-Account: sip:*9@10.10.10.1 Voice-Message: 0/0 (0/0) # U 2015/07/24 17:35:06.811600 10.10.10.11:5060 -> 10.10.10.1:5060 SIP/2.0 489 Bad event Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK762f8611;received=10.10.10.1;rport=5060 From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as3ee6db46 To: <sip:s@10.10.10.11:5060>;tag=as52e074fd Call-ID: 05cdf5b671961f597fe697d248b0dda7@10.10.10.1 CSeq: 102 NOTIFY Server: Asterisk PBX 1.8.32.3-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 # U 2015/07/24 17:35:06.812038 10.10.10.1:5060 -> 10.10.10.11:5060 NOTIFY sip:s@10.10.10.11:5060 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK4bc6a4f8;rport From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as7ad685b4 To: <sip:s@10.10.10.11:5060> Contact: <sip:asterisk@10.10.10.1> Call-ID: 4b6917505960251c3611038b419b210f@10.10.10.1 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Content-Length: 84 Messages-Waiting: no Message-Account: sip:*9@10.10.10.1 Voice-Message: 0/0 (0/0) # U 2015/07/24 17:35:06.812135 10.10.10.11:5060 -> 10.10.10.1:5060 SIP/2.0 489 Bad event Via: SIP/2.0/UDP 10.10.10.1:5060;branch=z9hG4bK4bc6a4f8;received=10.10.10.1;rport=5060 From: "asterisk" <sip:asterisk@10.10.10.1>;tag=as7ad685b4 To: <sip:s@10.10.10.11:5060>;tag=as64ff1592 Call-ID: 4b6917505960251c3611038b419b210f@10.10.10.1 CSeq: 102 NOTIFY Server: Asterisk PBX 1.8.32.3-vici Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 # U 2015/07/24 17:35:07.185070 10.10.10.11:5060 -> 10.10.10.1:5060 CANCEL sip:0478805437@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK2b3ed61e;rport Max-Forwards: 70 From: "V7241734470000005735" <sip:996@10.10.10.11>;tag=as5c51cb4b To: <sip:0478805437@10.10.10.1> Call-ID: 0e77b373360b2a0a3ded61872d4e7186@10.10.10.11:5060 CSeq: 103 CANCEL User-Agent: Asterisk PBX 1.8.32.3-vici Content-Length: 0 # U 2015/07/24 17:35:07.185409 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK2b3ed61e;received=10.10.10.11;rport=5060 From: "V7241734470000005735" <sip:996@10.10.10.11>;tag=as5c51cb4b To: <sip:0478805437@10.10.10.1>;tag=as0eca845f Call-ID: 0e77b373360b2a0a3ded61872d4e7186@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 # U 2015/07/24 17:35:07.185429 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK2b3ed61e;received=10.10.10.11;rport=5060 From: "V7241734470000005735" <sip:996@10.10.10.11>;tag=as5c51cb4b To: <sip:0478805437@10.10.10.1>;tag=as0eca845f Call-ID: 0e77b373360b2a0a3ded61872d4e7186@10.10.10.11:5060 CSeq: 103 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 # U 2015/07/24 17:35:07.185589 10.10.10.11:5060 -> 10.10.10.1:5060 ACK sip:0478805437@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK2b3ed61e;rport Max-Forwards: 70 From: "V7241734470000005735" <sip:996@10.10.10.11>;tag=as5c51cb4b To: <sip:0478805437@10.10.10.1>;tag=as0eca845f Contact: <sip:996@10.10.10.11:5060> Call-ID: 0e77b373360b2a0a3ded61872d4e7186@10.10.10.11:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.32.3-vici Content-Length: 0 # U 2015/07/24 17:35:45.220437 10.10.10.11:5060 -> 10.10.10.1:5060 OPTIONS sip:10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK565c4ae2;rport Max-Forwards: 70 From: "asterisk" <sip:996@10.10.10.11>;tag=as711236f7 To: <sip:10.10.10.1> Contact: <sip:996@10.10.10.11:5060> Call-ID: 11038a15323b23d82530187a538b53c6@10.10.10.11:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.32.3-vici Date: Fri, 24 Jul 2015 07:35:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 # U 2015/07/24 17:35:45.220759 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK565c4ae2;received=10.10.10.11;rport=5060 From: "asterisk" <sip:996@10.10.10.11>;tag=as711236f7 To: <sip:10.10.10.1>;tag=as6806e3e2 Call-ID: 11038a15323b23d82530187a538b53c6@10.10.10.11:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Accept: application/sdp Content-Length: 0 # U 2015/07/24 17:35:45.298032 10.10.10.11:5060 -> 10.10.10.1:5060 OPTIONS sip:10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK7b2171ae;rport Max-Forwards: 70 From: "asterisk" <sip:997@10.10.10.11>;tag=as6ae192a3 To: <sip:10.10.10.1> Contact: <sip:997@10.10.10.11:5060> Call-ID: 528fc9e30abfa261750e20f554bf6792@10.10.10.11:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.32.3-vici Date: Fri, 24 Jul 2015 07:35:45 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 # U 2015/07/24 17:35:45.298392 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK7b2171ae;received=10.10.10.11;rport=5060 From: "asterisk" <sip:997@10.10.10.11>;tag=as6ae192a3 To: <sip:10.10.10.1>;tag=as60833329 Call-ID: 528fc9e30abfa261750e20f554bf6792@10.10.10.11:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Accept: application/sdp Content-Length: 0

I suppose I should look into what is different about a manually dialed call vs an auto dialed call.

Posted: **Fri Jul 24, 2015 3:01 am**

So, I removed our secondary trunk and disabled anything superfluous, the call comes in, and disconnects at precisely 10 seconds.

The interaction in ngrep-sip:

Code: Select all: # U 2015/07/24 17:54:02.366615 10.10.10.11:5060 -> 10.10.10.1:5060 INVITE sip:0478805437@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK20ddc3a6;rport Max-Forwards: 70 From: "V7241754020000005735" <sip:996@10.10.10.11>;tag=as67212156 To: <sip:0478805437@10.10.10.1> Contact: <sip:996@10.10.10.11:5060> Call-ID: 5960b7c07f4561ba12024ea5199a70a8@10.10.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 1.8.32.3-vici Date: Fri, 24 Jul 2015 07:54:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "V7241754020000005735" <sip:0000000000@10.10.10.11>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 238 v=0 o=root 377043220 377043220 IN IP4 10.10.10.11 s=Asterisk PBX 1.8.32.3-vici c=IN IP4 10.10.10.11 t=0 0 m=audio 17400 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv # U 2015/07/24 17:54:02.366947 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK20ddc3a6;received=10.10.10.11;rport=5060 From: "V7241754020000005735" <sip:996@10.10.10.11>;tag=as67212156 To: <sip:0478805437@10.10.10.1>;tag=as6162184b Call-ID: 5960b7c07f4561ba12024ea5199a70a8@10.10.10.11:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="37d7b6db" Content-Length: 0 # U 2015/07/24 17:54:02.367078 10.10.10.11:5060 -> 10.10.10.1:5060 ACK sip:0478805437@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK20ddc3a6;rport Max-Forwards: 70 From: "V7241754020000005735" <sip:996@10.10.10.11>;tag=as67212156 To: <sip:0478805437@10.10.10.1>;tag=as6162184b Contact: <sip:996@10.10.10.11:5060> Call-ID: 5960b7c07f4561ba12024ea5199a70a8@10.10.10.11:5060 CSeq: 102 ACK User-Agent: Asterisk PBX 1.8.32.3-vici Content-Length: 0 # U 2015/07/24 17:54:02.367164 10.10.10.11:5060 -> 10.10.10.1:5060 INVITE sip:0478805437@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK3682265a;rport Max-Forwards: 70 From: "V7241754020000005735" <sip:996@10.10.10.11>;tag=as67212156 To: <sip:0478805437@10.10.10.1> Contact: <sip:996@10.10.10.11:5060> Call-ID: 5960b7c07f4561ba12024ea5199a70a8@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX 1.8.32.3-vici Proxy-Authorization: Digest username="996", realm="asterisk", algorithm=MD5, uri="sip:0478805437@10.10.10.1", nonce="37d7b6db", response="aabcae7b9fbe3489519979d799cac49f" Date: Fri, 24 Jul 2015 07:54:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Remote-Party-ID: "V7241754020000005735" <sip:0000000000@10.10.10.11>;party=calling;privacy=off;screen=no Content-Type: application/sdp Content-Length: 238 v=0 o=root 377043220 377043221 IN IP4 10.10.10.11 s=Asterisk PBX 1.8.32.3-vici c=IN IP4 10.10.10.11 t=0 0 m=audio 17400 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv # U 2015/07/24 17:54:02.367567 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK3682265a;received=10.10.10.11;rport=5060 From: "V7241754020000005735" <sip:996@10.10.10.11>;tag=as67212156 To: <sip:0478805437@10.10.10.1> Call-ID: 5960b7c07f4561ba12024ea5199a70a8@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:0478805437@10.10.10.1> Content-Length: 0 # U 2015/07/24 17:54:04.014495 10.10.10.11:5060 -> 10.10.10.1:5060 OPTIONS sip:10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK759416d1;rport Max-Forwards: 70 From: "asterisk" <sip:996@10.10.10.11>;tag=as18941b5b To: <sip:10.10.10.1> Contact: <sip:996@10.10.10.11:5060> Call-ID: 083399ca2bfac2b437cd7dc36e884082@10.10.10.11:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.8.32.3-vici Date: Fri, 24 Jul 2015 07:54:04 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 # U 2015/07/24 17:54:04.014844 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 404 Not Found Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK759416d1;received=10.10.10.11;rport=5060 From: "asterisk" <sip:996@10.10.10.11>;tag=as18941b5b To: <sip:10.10.10.1>;tag=as4dadafe5 Call-ID: 083399ca2bfac2b437cd7dc36e884082@10.10.10.11:5060 CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Accept: application/sdp Content-Length: 0 # U 2015/07/24 17:54:06.648226 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK3682265a;received=10.10.10.11;rport=5060 From: "V7241754020000005735" <sip:996@10.10.10.11>;tag=as67212156 To: <sip:0478805437@10.10.10.1>;tag=as3e42ce1c Call-ID: 5960b7c07f4561ba12024ea5199a70a8@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:0478805437@10.10.10.1> Content-Type: application/sdp Content-Length: 207 v=0 o=root 6229 6229 IN IP4 10.10.10.1 s=session c=IN IP4 10.10.10.1 t=0 0 m=audio 10096 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv # U 2015/07/24 17:54:09.168078 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK3682265a;received=10.10.10.11;rport=5060 From: "V7241754020000005735" <sip:996@10.10.10.11>;tag=as67212156 To: <sip:0478805437@10.10.10.1>;tag=as3e42ce1c Call-ID: 5960b7c07f4561ba12024ea5199a70a8@10.10.10.11:5060 CSeq: 103 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Contact: <sip:0478805437@10.10.10.1> Content-Type: application/sdp Content-Length: 207 v=0 o=root 6229 6230 IN IP4 10.10.10.1 s=session c=IN IP4 10.10.10.1 t=0 0 m=audio 10096 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv # U 2015/07/24 17:54:09.168258 10.10.10.11:5060 -> 10.10.10.1:5060 ACK sip:0478805437@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK5cd767ce;rport Max-Forwards: 70 From: "V7241754020000005735" <sip:996@10.10.10.11>;tag=as67212156 To: <sip:0478805437@10.10.10.1>;tag=as3e42ce1c Contact: <sip:996@10.10.10.11:5060> Call-ID: 5960b7c07f4561ba12024ea5199a70a8@10.10.10.11:5060 CSeq: 103 ACK User-Agent: Asterisk PBX 1.8.32.3-vici Content-Length: 0 # U 2015/07/24 17:54:18.970424 10.10.10.11:5060 -> 10.10.10.1:5060 BYE sip:0478805437@10.10.10.1 SIP/2.0 Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK085ada65;rport Max-Forwards: 70 From: "V7241754020000005735" <sip:996@10.10.10.11>;tag=as67212156 To: <sip:0478805437@10.10.10.1>;tag=as3e42ce1c Call-ID: 5960b7c07f4561ba12024ea5199a70a8@10.10.10.11:5060 CSeq: 104 BYE User-Agent: Asterisk PBX 1.8.32.3-vici Proxy-Authorization: Digest username="996", realm="asterisk", algorithm=MD5, uri="sip:0478805437@10.10.10.1", nonce="37d7b6db", response="50636666b0fa3d424bea0cbad02a1e33" X-Asterisk-HangupCause: Unknown X-Asterisk-HangupCauseCode: 0 Content-Length: 0 # U 2015/07/24 17:54:18.970719 10.10.10.1:5060 -> 10.10.10.11:5060 SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK085ada65;received=10.10.10.11;rport=5060 From: "V7241754020000005735" <sip:996@10.10.10.11>;tag=as67212156 To: <sip:0478805437@10.10.10.1>;tag=as3e42ce1c Call-ID: 5960b7c07f4561ba12024ea5199a70a8@10.10.10.11:5060 CSeq: 104 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0

And what the asterisk CLI sees:

Code: Select all: [Jul 24 18:00:05] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 18:00:05] -- Executing [9610478805437@default:1] AGI("Local/9610478805437@default-00000023;2", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 18:00:05] -- AGI Script Executing Application: (EXEC) Options: (Set(_CAMPCUST=PDBUSNSW)) [Jul 24 18:00:05] -- <Local/9610478805437@default-00000023;2>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 18:00:05] -- Executing [9610478805437@default:2] Dial("Local/9610478805437@default-00000023;2", "SIP/mytel1/0478805437,,tTor") in new stack [Jul 24 18:00:05] == Using SIP RTP CoS mark 5 [Jul 24 18:00:05] -- Called SIP/mytel1/0478805437 [Jul 24 18:00:06] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 18:00:06] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 18:00:10] -- SIP/mytel1-0000001c is making progress passing it to Local/9610478805437@default-00000023;2 [Jul 24 18:00:12] -- SIP/mytel1-0000001c answered Local/9610478805437@default-00000023;2 [Jul 24 18:00:12] > Channel Local/9610478805437@default-00000023;1 was answered. [Jul 24 18:00:12] -- Executing [8368@default:1] Playback("Local/9610478805437@default-00000023;1", "sip-silence") in new stack [Jul 24 18:00:12] -- <Local/9610478805437@default-00000023;1> Playing 'sip-silence.gsm' (language 'en') [Jul 24 18:00:12] -- Executing [h@default:1] AGI("Local/9610478805437@default-00000023;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----7-----0") in new stack [Jul 24 18:00:12] -- Executing [8368@default:2] AGI("SIP/mytel1-0000001c", "agi://127.0.0.1:4577/call_log") in new stack [Jul 24 18:00:12] -- <SIP/mytel1-0000001c>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 [Jul 24 18:00:12] -- Executing [8368@default:3] AGI("SIP/mytel1-0000001c", "agi-VDAD_ALL_outbound.agi,NORMAL-----LB") in new stack [Jul 24 18:00:12] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi [Jul 24 18:00:13] == Manager 'sendcron' logged off from 127.0.0.1 [Jul 24 18:00:13] -- <Local/9610478805437@default-00000023;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----7-----0 completed, returning 0 [Jul 24 18:00:13] == Spawn extension (default, 9610478805437, 2) exited non-zero on 'Local/9610478805437@default-00000023;2' [Jul 24 18:00:22] -- <SIP/mytel1-0000001c>AGI Script agi-VDAD_ALL_outbound.agi completed, returning 4 [Jul 24 18:00:22] == Spawn extension (default, 8368, 3) exited non-zero on 'SIP/mytel1-0000001c' [Jul 24 18:00:22] -- Executing [h@default:1] AGI("SIP/mytel1-0000001c", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack [Jul 24 18:00:22] -- <SIP/mytel1-0000001c>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0 [Jul 24 18:00:22] == Manager 'sendcron' logged on from 127.0.0.1 [Jul 24 18:00:23] == Manager 'sendcron' logged off from 127.0.0.1 LargeDialer*CLI>

Posted: **Fri Jul 24, 2015 3:47 am**

asterisk version correct in admin->servers?

Posted: **Fri Jul 24, 2015 3:51 am**

Sure is. It's the first thing I always check. Eliminates most weird asterisk behaviour.

What is exit code 4 in /usr/share/asterisk/agi-bin/agi-VDAD_ALL_outbound.agi ?

Just curious, even if unrelated.

From agi-out:

Code: Select all: 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|Perl Environment Dump: 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|NORMAL-----LB 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|AGI Environment Dump: 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- accountcode = 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- arg_1 = NORMAL-----LB 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- callerid = 996 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- calleridname = LargeDialer 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- callingani2 = 0 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- callingpres = 0 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- callingtns = 0 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- callington = 0 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- channel = SIP/mytel1-0000001e 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- context = default 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- dnid = unknown 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- enhanced = 0.0 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- extension = 8368 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- language = en 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- priority = 3 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- rdnis = unknown 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- request = agi-VDAD_ALL_outbound.agi 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- threadid = 140404623193856 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- type = SIP 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- uniqueid = 1437728521.113 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi| -- version = 1.8.32.3-vici 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|AGI Variables: |1437728521.113|SIP/mytel1-0000001e|8368|SIP|996|LargeDialer|3| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|+++++ VDAD START : |0|2015-07-24 19:02:08|1.8.32.3-vici|3| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_live_agents where callerid='996';| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where callerid='996' and status IN('LIVE','XFER');| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|-- VDAD : |0E0|update of vac table: 996 |UPDATE vicidial_auto_calls set uniqueid='1437728521.113', channel='SIP/mytel1-0000001e',status='LIVE',stage='LIVE-0' where callerid='996' order by call_time desc limit 1;| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|-- NO VDAC FOUND!!!!!: 996 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|-- NO VDM FOUND!!!!!!!!!!: 996 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|1|VDAC-reinsert|INSERT INTO vicidial_auto_calls (server_ip,campaign_id,status,lead_id,uniqueid,callerid,channel,phone_code,phone_number,call_time,call_type,stage,queue_priority) values('10.10.10.11','','LIVE','0','1437728521.113','996','SIP/mytel1-0000001e','','','2015-07-24 19:02:08','OUT','LIVE-0.25','')| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|0|SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";| 2015-07-24 19:02:08|agi-VDAD_ALL_outbound.agi|WWWWWWWW VDAD XFER BALANCE WAIT: |0||SIP/mytel1-0000001e|996|1437728521.113| 2015-07-24 19:02:18|agi-VDAD_ALL_outbound.agi||UPDATE vicidial_log FORCE INDEX(lead_id) set status='DROP',end_epoch='1437728538',length_in_sec='9',term_reason='QUEUETIMEOUT' where lead_id = '0' and uniqueid LIKE "1437728521%";| 2015-07-24 19:02:18|agi-VDAD_ALL_outbound.agi|-- VDAD vicidial_log update: |0E0|1437728521.113 2015-07-24 19:02:18|agi-VDAD_ALL_outbound.agi||UPDATE vicidial_list set status='DROP' where lead_id = '0';| 2015-07-24 19:02:18|agi-VDAD_ALL_outbound.agi|-- VDAD vicidial_list update: |0E0|0 2015-07-24 19:02:18|agi-VDAD_ALL_outbound.agi|-- VDAD vac record deleted: |1|| 2015-07-24 19:02:18|agi-VDAD_ALL_outbound.agi|-- VDCL call_hungup timout: |0|VH0724190218||SIP/mytel1-0000001e|insert to vicidial_manager

Posted: **Fri Jul 24, 2015 9:35 pm**

Is there any other log files I could offer?

I'm really stumped.

Posted: **Mon Jul 27, 2015 4:32 am**

Just removed the lot, grabbed the preload ISO and gave it a shot as barebones as possible, still no luck. Call sits in queue for 10 seconds and drops. Autocalls do not pass to agents.

I enabled the outbound queue in the agent screen to see what's coming through.

It appears that every field is blank. No phone number or lead information.

Posted: **Tue Jul 28, 2015 9:24 pm**

Looks like the most recent version is a no-go.

I reinstalled for a 5th time after 4 other variations, this time I didn't perform an upgrade to 2347.

Same settings all work on SVN 2192, but not on SVN 2347.

In the past I have installed and upgraded fine, but this one revision didn't cooperate. Not sure why.

Posted: **Wed Jul 29, 2015 7:24 am**

We just did an install with the non-preload ISO and it worked fine. I'm not really sure what the problem is that you are having.

Posted: **Thu Jul 30, 2015 8:40 pm**

When you performed these clean installs, you obviously had to put in some settings to make the server work. Even barebones has to be configured. Networking and carrier settings are required, and are different on each machine.

Perhaps you should share your carrier settings and internet connection configuration. And any other settings you had to personalized during setup.

Posted: **Thu Aug 06, 2015 2:13 am**

Hi Matt,

Is this problem seen in pre load version of 6.0.3.? or you tested only in non-preload ISO.

@Arfeh,

You mean on same platform SVN 2192 is working? but not SVN 2347.

Can we downgrade from higher svn to lower ?

Posted: **Thu Aug 06, 2015 6:16 am**

We don't really test the preload ISO very often. The one we use in production is the non-preload ISO.

Posted: **Thu Aug 06, 2015 10:06 am**

Thanks Matt for the input.

will try non preload and migrate the DB and check.

Posted: **Sat Aug 08, 2015 11:06 pm**

Hi Matt,

I think only reason for this issue is unknown callerID in autocalls. Below are the MYQL logs . Script is failing to get live agent and campaign information from where clause --CallerID = 'unknown'.

Please help us to understand who /which script is setting this callerid for auto calls. I believe its some long alpha numeric character.

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

2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01002|1|||SELECT count(*) FROM vicidial_live_agents where callerid='unknown';|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01003|1|||SELECT count(*) FROM vicidial_auto_calls where callerid='unknown' and status IN('LIVE','XFER');|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01004|0E0|||UPDATE vicidial_auto_calls set uniqueid='1439090234.763522', channel='DAHDI/i1/0989898989-23845',status='LIVE',stage='LIVE-0' where callerid='unknown' order by call_time desc limit 1;|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01070|0|||SELECT cmd_line_k,entry_date,cmd_line_j FROM vicidial_manager where callerid='unknown' and action='Originate' order by entry_date,cmd_line_k desc limit 1;|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01047|1|||INSERT INTO vicidial_auto_calls (server_ip,campaign_id,status,lead_id,uniqueid,callerid,channel,phone_code,phone_number,call_time,call_type,stage,queue_priority) values('192.168.1.10','','LIVE','0','1439090234.763522','unknown','DAHDI/i1/0989898989-23845','','','2015-08-09 08:47:23','OUT','LIVE-0.25','')|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
2015-08-09 08:47:23|agi-VDAD_ALL_outbound.agi|01040|1|||SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and lead_id != '0' and campaign_id = '' and call_time < "";|unknown|0|
@
-----------------------------------------------------------------------

Posted: **Thu Aug 27, 2015 9:35 pm**

williamconley wrote:Perhaps you should share your carrier settings ...

This can be caused by improper carrier settings. Do you include the AGI and Hangup lines in your carrier settings which you have not yet posted? 8-)

Posted: **Wed Sep 02, 2015 10:38 pm**

carrier setting is there in very first post--extensions.conf.

Posted: **Wed Sep 30, 2015 8:28 am**

Hi

I added below lines in the dial plan and to VDAD transfer number 8368, issue resolved... calls are landing happily

))))

exten => _91X.,1,AGI(agi://127.0.0.1:4577/call_log);
exten => _91X.,n,Set(_Missing_CLID1=${CALLERID(all)});
exten => _91X.,n,Dial(${ogtrunk}/0${EXTEN:2},,tToR)
exten => _91X.,n,Hangup()

exten => 8368,1,Playback(sip-silence)
exten => 8368,n,Set(CALLERID(all)=${Missing_CLID1});
exten => 8368,n,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,Hangup()

Posted: **Mon May 04, 2020 12:52 pm**

lvish wrote:Hi

I added below lines in the dial plan and to VDAD transfer number 8368, issue resolved... calls are landing happily ))))

exten => _91X.,1,AGI(agi://127.0.0.1:4577/call_log);
exten => _91X.,n,Set(_Missing_CLID1=${CALLERID(all)});
exten => _91X.,n,Dial(${ogtrunk}/0${EXTEN:2},,tToR)
exten => _91X.,n,Hangup()

exten => 8368,1,Playback(sip-silence)
exten => 8368,n,Set(CALLERID(all)=${Missing_CLID1});
exten => 8368,n,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,n,Hangup()

Puffff Man this topic should be PINNED somewhere. 5 Years old and save my ass big time. I was completely loss debugging the same problem and it was hard to find your answer. How can I buy you a beer ? like for real. Thanks!

Posted: **Mon May 04, 2020 1:07 pm**

Unfortunately this solution will not work properly under load. Asterisk is known to lose dialplan variables when it transitions from Local channels to the underlying SIP/IAX/DAHDI channel when under load. The _91X. extension is for Local channels. The 8368 extension is for the underlying channel.

When setup properly Asterisk 13+ should resolve this problem. If you are having this issue with Asterisk 13 it means your dialplan / server settings are not setup properly.

vicidial.org

Autodial calls not transferring to agent, Manual working fin

Autodial calls not transferring to agent, Manual working fin

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent,-resolved

Re: Autodial calls not transferring to agent, Manual working

Re: Autodial calls not transferring to agent, Manual working