vicidial.org

by **antuan** » Tue Oct 26, 2021 6:10 am

Hello,

I am facing an issue when using the TRANSFER - CONF from an agent client screen, in detail:

- click TRANSFER - CONF
- write an internal extension in NUMBER TO CALL box
- click DIAL WITH CUSTOMER

internal extension is ringing
but when someone picks it up it instantly hangs up

This is not happening with older VICIBox ex. v6, in same environment

Mentioning that 3 way DIAL WITH CUSTOMER to an external number works ok
and directly BLIND TRANFER (even with internal extension) works ok too.

Maybe it is a missconfig, it is happening only at 3way DIAL WITH CUSTOMER to an internal extension.
Does it make any sense?

Thank you for any help you can offer.

VICIBox v9.0.3
VERSION: 2.14-830a
BUILD: 210920-2159
Asterisk 13.29.2-vici

Version: 2.14b0.5
SVN Version: 3529
DB Schema Version: 1645

by **carpenox** » Tue Oct 26, 2021 5:19 pm

try adding --cu3way to the keepalive script in crontab and see if that works for you

by **antuan** » Wed Oct 27, 2021 1:57 am

Added --cu3way but it did not make any difference.
Also tried /usr/share/astguiclient/ADMIN_keepalive_ALL.pl --cu3way --cu3way-delay=1

by **carpenox** » Wed Oct 27, 2021 4:32 pm

is iprelay running?

by **antuan** » Mon Nov 01, 2021 2:44 am

yes ip_relay is running

● vicidial.service - LSB: ViciDial Telephony Server
Loaded: loaded (/etc/init.d/vicidial; generated; vendor preset: disabled)
Active: active (running) since Mon 2021-11-01 06:01:04 EET; 3h 40min ago
Docs: man:systemd-sysv-generator(8)
Process: 1345 ExecStart=/etc/init.d/vicidial start (code=exited, status=0/SUCCESS)
Tasks: 119
CGroup: /system.slice/vicidial.service
├─1716 /usr/bin/SCREEN -S astshell20211101060059
├─1717 /bin/sh
├─1721 SCREEN -L -S asterisk
├─1722 /bin/sh
├─1812 /usr/sbin/asterisk -vvvvvvvvvvvvvvvvvvvvvgcT
├─1839 ip_relay 40569 127.0.0.1 4569 9999999
├─1840 ip_relay 41569 127.0.0.1 4569 9999999
└─1841 ip_relay 42569 127.0.0.1 4569 9999999

Nov 01 06:00:53 cti vicidial[1345]: Fallback to dahdi module... done.
Nov 01 06:00:58 cti vicidial[1345]: Initializing DAHDI Hardware... done.
Nov 01 06:00:58 cti vicidial[1345]: Checking Database connectivity... done.
Nov 01 06:00:59 cti vicidial[1345]: Resetting vars and rolling logs... done.
Nov 01 06:01:04 cti vicidial[1345]: Starting Asterisk... PID 1812, done.
Nov 01 06:01:04 cti vicidial[1345]: Giving asterisk -1 process priority... done.
Nov 01 06:01:04 cti vicidial[1345]: Starting ip_relay with OS specific version... done.
Nov 01 06:01:04 cti vicidial[1345]: ViciDial Telephony Server Started. Log at /var/log/vicidial.log
Nov 01 06:01:04 cti vicidial[1345]: ..done
Nov 01 06:01:04 cti systemd[1]: Started LSB: ViciDial Telephony Server.

by **carpenox** » Mon Nov 01, 2021 8:13 am

show me your asterisk cli when you try a transfer so i can see what happens

by **antuan** » Mon Nov 01, 2021 9:12 am

It is an inbound call to a DID,
agent picks it up
TRANSFER - CONF
NUMBER TO DIAL 253
DIAL WITH CUSTOMER

calling 253 (maybe inviting to conference is a better term) and it rings but instanlty is been hangup when 253 softphone picks it up
tried with voip phones too

252 is the webphone extension asigned to agent

[Nov 1 15:54:13] -- Executing [8600051@default:1] MeetMe("SIP/252-00000000", "8600051,F") in new stack
[Nov 1 15:54:13] -- Created MeetMe conference 1023 for conference '8600051'
[Nov 1 15:54:13] -- <SIP/252-00000000> Playing 'conf-onlyperson.gsm' (language 'gr')
[Nov 1 15:54:13] > 0x7f2c9801eea0 -- Strict RTP learning after ICE completion
[Nov 1 15:54:13] > 0x7f2c9801eea0 -- Strict RTP switching to RTP target address 192.168.23.60:65149 as source
[Nov 1 15:54:14] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 1 15:54:18] > 0x7f2c9801eea0 -- Strict RTP learning complete - Locking on source address 192.168.23.60:65149
[Nov 1 15:54:36] -- Remote UNIX connection
[Nov 1 15:54:36] -- Remote UNIX connection disconnected
[Nov 1 15:54:36] -- Remote UNIX connection
[Nov 1 15:54:36] -- Remote UNIX connection disconnected
[Nov 1 15:54:39] == Using SIP RTP CoS mark 5
[Nov 1 15:54:39] > 0x7f2c940b1710 -- Strict RTP learning after remote address set to: 146.120.226.3:18438
[Nov 1 15:54:39] -- Executing [302170004582@trunkinbound:1] AGI("SIP/provider_test-00000001", "agi-DID_route.agi") in new stack
[Nov 1 15:54:39] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Nov 1 15:54:40] -- <SIP/provider_test-00000001>AGI Script agi-DID_route.agi completed, returning 0
[Nov 1 15:54:40] -- Executing [99909*7***DID@default:1] Answer("SIP/provider_test-00000001", "") in new stack
[Nov 1 15:54:40] > 0x7f2c940b1710 -- Strict RTP switching to RTP target address 146.120.226.3:18438 as source
[Nov 1 15:54:40] -- Executing [99909*7***DID@default:2] AGI("SIP/provider_test-00000001", "agi-VDAD_ALL_inbound.agi") in new stack
[Nov 1 15:54:40] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Nov 1 15:54:41] -- <SIP/provider_test-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'gr')
[Nov 1 15:54:41] -- <SIP/provider_test-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'gr')
[Nov 1 15:54:43] -- Started music on hold, class 'OnHold', on channel 'SIP/provider_test-00000001'
[Nov 1 15:54:44] > 0x7f2c940b1710 -- Strict RTP learning complete - Locking on source address 146.120.226.3:18438
[Nov 1 15:54:56] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 15:54:56] -- Called 192*168*023*019*78600051@default
[Nov 1 15:54:56] -- Executing [192*168*023*019*78600051@default:1] Goto("Local/192*168*023*019*78600051@default-00000000;2", "default,78600051,1") in new stack
[Nov 1 15:54:56] -- Goto (default,78600051,1)
[Nov 1 15:54:56] -- Executing [78600051@default:1] MeetMe("Local/192*168*023*019*78600051@default-00000000;2", "8600051,Fq") in new stack
[Nov 1 15:54:56] -- Local/192*168*023*019*78600051@default-00000000;1 answered
[Nov 1 15:54:56] -- Executing [83047777777777@vicidial-auto:1] Answer("Local/192*168*023*019*78600051@default-00000000;1", "") in new stack
[Nov 1 15:54:56] -- Executing [83047777777777@vicidial-auto:2] Playback("Local/192*168*023*019*78600051@default-00000000;1", "ding") in new stack
[Nov 1 15:54:56] -- <Local/192*168*023*019*78600051@default-00000000;1> Playing 'ding.gsm' (language 'gr')
[Nov 1 15:54:56] -- Executing [83047777777777@vicidial-auto:3] Hangup("Local/192*168*023*019*78600051@default-00000000;1", "") in new stack
[Nov 1 15:54:56] == Spawn extension (vicidial-auto, 83047777777777, 3) exited non-zero on 'Local/192*168*023*019*78600051@default-00000000;1'
[Nov 1 15:54:56] WARNING[11595][C-00000003]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Nov 1 15:54:56] -- Executing [h@vicidial-auto:1] AGI("Local/192*168*023*019*78600051@default-00000000;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Nov 1 15:54:56] -- <Local/192*168*023*019*78600051@default-00000000;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
[Nov 1 15:54:56] == Spawn extension (default, 78600051, 1) exited non-zero on 'Local/192*168*023*019*78600051@default-00000000;2'
[Nov 1 15:54:56] WARNING[11596][C-00000002]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Nov 1 15:54:56] -- Executing [h@default:1] AGI("Local/192*168*023*019*78600051@default-00000000;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Nov 1 15:54:56] -- <Local/192*168*023*019*78600051@default-00000000;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
[Nov 1 15:54:57] -- Stopped music on hold on SIP/provider_test-00000001
[Nov 1 15:54:57] -- <SIP/provider_test-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'gr')
[Nov 1 15:54:57] -- <SIP/provider_test-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'gr')
[Nov 1 15:54:57] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 1 15:54:57] -- <SIP/provider_test-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'gr')
[Nov 1 15:54:57] -- <SIP/provider_test-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'gr')
[Nov 1 15:54:57] -- <SIP/provider_test-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'gr')
[Nov 1 15:54:57] -- <SIP/provider_test-00000001> Playing 'sip-silence.gsm' (escape_digits=) (sample_offset 0) (language 'gr')
[Nov 1 15:54:57] -- <SIP/provider_test-00000001>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Nov 1 15:54:57] -- Executing [192*168*023*019*8600051@default:1] Goto("SIP/provider_test-00000001", "default,8600051,1") in new stack
[Nov 1 15:54:57] -- Goto (default,8600051,1)
[Nov 1 15:54:57] -- Executing [8600051@default:1] MeetMe("SIP/provider_test-00000001", "8600051,F") in new stack
[Nov 1 15:55:02] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 15:55:02] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 1 15:55:02] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 15:55:02] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 15:55:03] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 1 15:55:04] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 1 15:55:07] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 15:55:07] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 1 15:55:36] -- Remote UNIX connection
[Nov 1 15:55:36] -- Remote UNIX connection disconnected
[Nov 1 15:55:36] -- Remote UNIX connection
[Nov 1 15:55:36] -- Remote UNIX connection disconnected
[Nov 1 15:55:42] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 15:55:42] -- Called 253@default
[Nov 1 15:55:42] -- Executing [253@default:1] Dial("Local/253@default-00000001;2", "SIP/253,60,") in new stack
[Nov 1 15:55:42] == Using SIP RTP CoS mark 5
[Nov 1 15:55:42] -- Called SIP/253
[Nov 1 15:55:43] -- SIP/253-00000002 is ringing
[Nov 1 15:55:43] -- Local/253@default-00000001;1 is ringing
[Nov 1 15:55:45] > 0x7f2ca8023d90 -- Strict RTP learning after remote address set to: 192.168.23.88:8000
[Nov 1 15:55:45] -- SIP/253-00000002 answered Local/253@default-00000001;2
[Nov 1 15:55:45] -- Local/253@default-00000001;1 answered
[Nov 1 15:55:45] -- Executing [8600051@default:1] MeetMe("Local/253@default-00000001;1", "8600051,F") in new stack
[Nov 1 15:55:45] -- Channel SIP/253-00000002 joined 'simple_bridge' basic-bridge <af0fa014-bc26-4a9d-ae4f-beeace78233d>
[Nov 1 15:55:45] -- Channel Local/253@default-00000001;2 joined 'simple_bridge' basic-bridge <af0fa014-bc26-4a9d-ae4f-beeace78233d>
[Nov 1 15:55:46] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 1 15:55:47] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 15:55:47] -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/253-00000002
[Nov 1 15:55:47] -- Channel SIP/253-00000002 left 'simple_bridge' basic-bridge <af0fa014-bc26-4a9d-ae4f-beeace78233d>
[Nov 1 15:55:47] -- Channel Local/253@default-00000001;2 left 'simple_bridge' basic-bridge <af0fa014-bc26-4a9d-ae4f-beeace78233d>
[Nov 1 15:55:47] == Spawn extension (default, 253, 1) exited non-zero on 'Local/253@default-00000001;2'
[Nov 1 15:55:47] -- Executing [h@default:1] AGI("Local/253@default-00000001;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----4-----1-----SIP 200 OK)") in new stack
[Nov 1 15:55:47] -- <Local/253@default-00000001;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... -1-----SIP 200 OK) completed, returning 0
[Nov 1 15:55:47] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/253@default-00000001;1'
[Nov 1 15:55:47] WARNING[11942][C-00000005]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Nov 1 15:55:47] -- Executing [h@default:1] AGI("Local/253@default-00000001;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Nov 1 15:55:47] -- <Local/253@default-00000001;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
[Nov 1 15:55:48] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 1 15:55:50] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 15:55:50] -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/provider_test-00000001
[Nov 1 15:55:50] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/provider_test-00000001'
[Nov 1 15:55:50] WARNING[11571][C-00000001]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Nov 1 15:55:50] -- Executing [h@default:1] AGI("SIP/provider_test-00000001", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------)") in new stack
[Nov 1 15:55:50] -- <SIP/provider_test-00000001>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ----------) completed, returning 0
[Nov 1 15:55:50] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 15:55:50] NOTICE[11964]: manager.c:4458 action_hangup: Request to hangup non-existent channel: SIP/provider_test-00000001
[Nov 1 15:55:51] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 1 15:55:51] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 1 15:56:01] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 15:56:01] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 1 15:56:01] == Manager 'sendcron' logged off from 127.0.0.1
cti*CLI> exit
[Nov 1 15:56:03] Asterisk cleanly ending (0).
[Nov 1 15:56:03] Executing last minute cleanups

by **GenXOutsourcing** » Mon Nov 01, 2021 10:41 am

Do your agents have the CONSULTIVE box checked? I do believe when transferring internally, you need to use CONSULTIVE

by **ambiorixg12** » Mon Nov 01, 2021 8:52 pm

As soon as 253 enter to the conference room, Asterisk manager is terminating the call

[Nov 1 15:55:47] -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/253-00000002

by **carpenox** » Mon Nov 01, 2021 9:44 pm

why are your conferences 00000001 instead of 8600001 ?

by **antuan** » Tue Nov 02, 2021 2:56 am

conferences are in 8600 form

cti*CLI> meetme list
Conf Num Parties Marked Activity Creation Locked
8600051 0001 N/A 00:01:24 Static No

also tried Consultative option and it does not call/ring the extension ex. 253

by **antuan** » Tue Nov 02, 2021 9:03 am

ambiorixg12 wrote:As soon as 253 enter to the conference room, Asterisk manager is terminating the call

[Nov 1 15:55:47] -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/253-00000002

Yes i have trouble finding why this is happening

if someone can try a 3way conference procedure with DIAL WITH CUSTOMER to an internal extension to see if it is working,
maybe i have something missconfigured

tried on the release
VICIBox v9.0.3
VERSION: 2.14-830a
BUILD: 210920-2159
Asterisk 13.29.2-vici

Version: 2.14b0.5
SVN Version: 3529
DB Schema Version: 1645

by **ambiorixg12** » Tue Nov 02, 2021 12:34 pm

I suggest you do an upgrade to other version of V9 and try it again, I don't think this issue is related to any misconfiguration. I might be wrong but I dont think so

by **carpenox** » Tue Nov 02, 2021 2:16 pm

thats the vicidial conferences, but what about the regular conferences? go to admin > conferences > show conferences it should look like this:

by **antuan** » Wed Nov 03, 2021 3:08 am

carpenox wrote:thats the vicidial conferences, but what about the regular conferences? go to admin > conferences > show conferences it should look like this:

conferences

agent conferences

by **antuan** » Wed Nov 03, 2021 3:16 am

ambiorixg12 wrote:I suggest you do an upgrade to other version of V9 and try it again, I don't think this issue is related to any misconfiguration. I might be wrong but I dont think so

yes i have tried on v10 too but i get the same instant hangup
ViciBox v.10.0.0 210901
Version: 2.14b0.5
SVN Version: 3534
DB Schema Version: 1646
Asterisk Version 13.38.2-vici

by **antuan** » Wed Nov 03, 2021 3:38 am

i am posting a cli output of an old v6 version where is working ok
same environment network and siphones
maybe there is a critical difference in a way
vicibox v.6.0.4
Version: 2.14b0.5
SVN Version: 2722
DB Schema Version: 1498
Asterisk Version: 1.8.32.3-vici

same procedure
inbound call to a DID,
agent picks it up
TRANSFER - CONF
NUMBER TO DIAL 251
DIAL WITH CUSTOMER

251 rings and talks ok

250 is the extension asigned to agent

[Nov 3 09:54:12] -- Executing [302170004582@trunkinbound:1] AGI("SIP/provider_test-0000000d", "agi-DID_route.agi") in new stack
[Nov 3 09:54:12] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Nov 3 09:54:12] -- <SIP/provider_test-0000000d>AGI Script agi-DID_route.agi completed, returning 0
[Nov 3 09:54:12] -- Executing [99909*34***DID@default:1] Answer("SIP/provider_test-0000000d", "") in new stack
[Nov 3 09:54:12] -- Executing [99909*34***DID@default:2] AGI("SIP/provider_test-0000000d", "agi-VDAD_ALL_inbound.agi") in new stack
[Nov 3 09:54:12] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Nov 3 09:54:13] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Nov 3 09:54:13] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Nov 3 09:54:14] -- Started music on hold, class 'OnHold', on SIP/provider_test-0000000d
[Nov 3 09:54:23] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 3 09:54:23] -- Executing [192*168*023*016*78600051@default:1] Goto("Local/192*168*023*016*78600051@default-00000025;2", "default,78600051,1") in new stack
[Nov 3 09:54:23] -- Goto (default,78600051,1)
[Nov 3 09:54:23] -- Executing [78600051@default:1] MeetMe("Local/192*168*023*016*78600051@default-00000025;2", "8600051,Fq") in new stack
[Nov 3 09:54:23] > Channel Local/192*168*023*016*78600051@default-00000025;1 was answered.
[Nov 3 09:54:23] -- Executing [83047777777777@vicidial-auto:1] Answer("Local/192*168*023*016*78600051@default-00000025;1", "") in new stack
[Nov 3 09:54:23] -- Executing [83047777777777@vicidial-auto:2] Playback("Local/192*168*023*016*78600051@default-00000025;1", "ding") in new stack
[Nov 3 09:54:24] -- <Local/192*168*023*016*78600051@default-00000025;1> Playing 'ding.gsm' (language 'en')
[Nov 3 09:54:24] -- Executing [83047777777777@vicidial-auto:3] Hangup("Local/192*168*023*016*78600051@default-00000025;1", "") in new stack
[Nov 3 09:54:24] == Spawn extension (vicidial-auto, 83047777777777, 3) exited non-zero on 'Local/192*168*023*016*78600051@default-00000025;1'
[Nov 3 09:54:24] -- Executing [h@vicidial-auto:1] AGI("Local/192*168*023*016*78600051@default-00000025;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16---------------") in new stack
[Nov 3 09:54:24] -- <Local/192*168*023*016*78600051@default-00000025;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------- completed, returning 0
[Nov 3 09:54:24] == Spawn extension (default, 78600051, 1) exited non-zero on 'Local/192*168*023*016*78600051@default-00000025;2'
[Nov 3 09:54:24] -- Executing [h@default:1] AGI("Local/192*168*023*016*78600051@default-00000025;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Nov 3 09:54:24] -- <Local/192*168*023*016*78600051@default-00000025;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Nov 3 09:54:24] -- Stopped music on hold on SIP/provider_test-0000000d
[Nov 3 09:54:24] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Nov 3 09:54:24] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Nov 3 09:54:24] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 3 09:54:25] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Nov 3 09:54:25] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Nov 3 09:54:25] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Nov 3 09:54:25] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Nov 3 09:54:25] -- <SIP/provider_test-0000000d>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Nov 3 09:54:25] -- Executing [192*168*023*016*8600051@default:1] Goto("SIP/provider_test-0000000d", "default,8600051,1") in new stack
[Nov 3 09:54:25] -- Goto (default,8600051,1)
[Nov 3 09:54:25] -- Executing [8600051@default:1] MeetMe("SIP/provider_test-0000000d", "8600051,F") in new stack
[Nov 3 09:54:25] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 3 09:54:25] -- Executing [58600051@default:1] MeetMe("Local/58600051@default-00000026;2", "8600051,Fmq") in new stack
[Nov 3 09:54:25] > Channel Local/58600051@default-00000026;1 was answered.
[Nov 3 09:54:25] -- Executing [8309@default:1] Answer("Local/58600051@default-00000026;1", "") in new stack
[Nov 3 09:54:25] -- Executing [8309@default:2] Monitor("Local/58600051@default-00000026;1", "wav,20211103-095424_POSTADDR2_2182189218_889_user") in new stack
[Nov 3 09:54:25] -- Executing [8309@default:3] Wait("Local/58600051@default-00000026;1", "3600") in new stack
[Nov 3 09:54:26] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 3 09:54:31] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 3 09:54:31] -- Executing [251@default:1] Dial("Local/251@default-00000027;2", "SIP/251,60,") in new stack
[Nov 3 09:54:31] == Using SIP RTP CoS mark 5
[Nov 3 09:54:31] -- Called SIP/251
[Nov 3 09:54:32] -- SIP/251-0000000e is ringing
[Nov 3 09:54:34] NOTICE[16907]: res_rtp_asterisk.c:2382 ast_rtp_read: Unknown RTP codec 95 received from '192.168.23.88:8000'
[Nov 3 09:54:34] -- SIP/251-0000000e answered Local/251@default-00000027;2
[Nov 3 09:54:34] > Channel Local/251@default-00000027;1 was answered.
[Nov 3 09:54:34] -- Executing [8600051@default:1] MeetMe("Local/251@default-00000027;1", "8600051,F") in new stack
[Nov 3 09:54:35] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 3 09:54:50] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 3 09:54:50] == Spawn extension (default, 8600051, 1) exited non-zero on 'SIP/provider_test-0000000d'
[Nov 3 09:54:50] -- Executing [h@default:1] AGI("SIP/provider_test-0000000d", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Nov 3 09:54:50] -- <SIP/provider_test-0000000d>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Nov 3 09:54:50] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 3 09:54:50] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/251@default-00000027;1'
[Nov 3 09:54:50] -- Executing [h@default:1] AGI("Local/251@default-00000027;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Nov 3 09:54:50] -- <Local/251@default-00000027;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Nov 3 09:54:50] -- Executing [h@default:1] AGI("Local/251@default-00000027;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----19-----16") in new stack
[Nov 3 09:54:50] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 3 09:54:50] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 3 09:54:50] == Spawn extension (default, 58600051, 1) exited non-zero on 'Local/58600051@default-00000026;2'
[Nov 3 09:54:50] -- Executing [h@default:1] AGI("Local/58600051@default-00000026;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Nov 3 09:54:50] -- <Local/251@default-00000027;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----19-----16 completed, returning 0
[Nov 3 09:54:50] -- <Local/58600051@default-00000026;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Nov 3 09:54:50] == Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600051@default-00000026;1'
[Nov 3 09:54:50] -- Executing [h@default:1] AGI("Local/58600051@default-00000026;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Nov 3 09:54:50] -- <Local/58600051@default-00000026;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Nov 3 09:54:51] == Spawn extension (default, 251, 1) exited non-zero on 'Local/251@default-00000027;2'
[Nov 3 09:54:51] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 3 09:54:51] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 3 09:54:51] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 3 09:54:51] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 3 09:55:01] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 3 09:55:01] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 3 09:55:01] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 3 09:55:01] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 3 09:55:01] == Manager 'sendcron' logged off from 127.0.0.1
cti*CLI> exit
Executing last minute cleanups

by **carpenox** » Wed Nov 03, 2021 11:02 pm

msg me on skype ill see if i can fix it

by **antuan** » Thu Nov 04, 2021 5:26 am

carpenox wrote:msg me on skype ill see if i can fix it

ok thank you
sent invitation from vbouzas

by **antuan** » Fri Nov 05, 2021 2:58 am

i am posting a cli output with sip debug on
when used the DIAL WITH CUSTOMER
noticed in dialog
.
[Nov 5 09:41:57] Contact: <sip:192.168.23.155:54836>
[Nov 5 09:41:57] From: "asterisk"<sip:asterisk@192.168.23.150>;tag=as06aeed11
.

they are after BYE but dont know if they play a role

vicibox10*CLI> sip set debug on
SIP Debugging enabled
[Nov 5 09:41:52] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 5 09:41:52] -- Called 251@default
[Nov 5 09:41:52] -- Executing [251@default:1] Dial("Local/251@default-00000010;2", "SIP/251,60,") in new stack
[Nov 5 09:41:52] == Using SIP RTP CoS mark 5
[Nov 5 09:41:52] Audio is at 19564
[Nov 5 09:41:52] Adding codec alaw to SDP
[Nov 5 09:41:52] Adding codec ulaw to SDP
[Nov 5 09:41:52] Adding codec gsm to SDP
[Nov 5 09:41:52] Adding non-codec 0x1 (telephone-event) to SDP
[Nov 5 09:41:52] Reliably Transmitting (NAT) to 192.168.23.155:54836:
[Nov 5 09:41:52] INVITE sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663 SIP/2.0
[Nov 5 09:41:52] Via: SIP/2.0/UDP 192.168.23.150:5060;branch=z9hG4bK19086955;rport
[Nov 5 09:41:52] Max-Forwards: 70
[Nov 5 09:41:52] From: "DC098096W0000000051W" <sip:302170004582@192.168.23.150>;tag=as1c8fc7dc
[Nov 5 09:41:52] To: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>
[Nov 5 09:41:52] Contact: <sip:302170004582@192.168.23.150:5060>
[Nov 5 09:41:52] Call-ID: 4d5a383f641a35bc7b0983006edf4e97@192.168.23.150:5060
[Nov 5 09:41:52] CSeq: 102 INVITE
[Nov 5 09:41:52] User-Agent: Asterisk PBX 13.38.2-vici
[Nov 5 09:41:52] Date: Fri, 05 Nov 2021 07:41:52 GMT
[Nov 5 09:41:52] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 5 09:41:52] Supported: replaces, timer
[Nov 5 09:41:52] Remote-Party-ID: "DC098096W0000000051W" <sip:302170004582@192.168.23.150>;party=calling;privacy=off;screen=no
[Nov 5 09:41:52] Content-Type: application/sdp
[Nov 5 09:41:52] Content-Length: 304
[Nov 5 09:41:52]
[Nov 5 09:41:52] v=0
[Nov 5 09:41:52] o=root 26065135 26065135 IN IP4 192.168.23.150
[Nov 5 09:41:52] s=Asterisk PBX 13.38.2-vici
[Nov 5 09:41:52] c=IN IP4 192.168.23.150
[Nov 5 09:41:52] t=0 0
[Nov 5 09:41:52] m=audio 19564 RTP/AVP 8 0 3 101
[Nov 5 09:41:52] a=rtpmap:8 PCMA/8000
[Nov 5 09:41:52] a=rtpmap:0 PCMU/8000
[Nov 5 09:41:52] a=rtpmap:3 GSM/8000
[Nov 5 09:41:52] a=rtpmap:101 telephone-event/8000
[Nov 5 09:41:52] a=fmtp:101 0-16
[Nov 5 09:41:52] a=ptime:20
[Nov 5 09:41:52] a=maxptime:150
[Nov 5 09:41:52] a=sendrecv
[Nov 5 09:41:52]
[Nov 5 09:41:52] ---
[Nov 5 09:41:52] -- Called SIP/251
[Nov 5 09:41:52] Retransmitting #1 (NAT) to 192.168.23.155:54836:
[Nov 5 09:41:52] INVITE sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663 SIP/2.0
[Nov 5 09:41:52] Via: SIP/2.0/UDP 192.168.23.150:5060;branch=z9hG4bK19086955;rport
[Nov 5 09:41:52] Max-Forwards: 70
[Nov 5 09:41:52] From: "DC098096W0000000051W" <sip:302170004582@192.168.23.150>;tag=as1c8fc7dc
[Nov 5 09:41:52] To: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>
[Nov 5 09:41:52] Contact: <sip:302170004582@192.168.23.150:5060>
[Nov 5 09:41:52] Call-ID: 4d5a383f641a35bc7b0983006edf4e97@192.168.23.150:5060
[Nov 5 09:41:52] CSeq: 102 INVITE
[Nov 5 09:41:52] User-Agent: Asterisk PBX 13.38.2-vici
[Nov 5 09:41:52] Date: Fri, 05 Nov 2021 07:41:52 GMT
[Nov 5 09:41:52] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 5 09:41:52] Supported: replaces, timer
[Nov 5 09:41:52] Remote-Party-ID: "DC098096W0000000051W" <sip:302170004582@192.168.23.150>;party=calling;privacy=off;screen=no
[Nov 5 09:41:52] Content-Type: application/sdp
[Nov 5 09:41:52] Content-Length: 304
[Nov 5 09:41:52]
[Nov 5 09:41:52] v=0
[Nov 5 09:41:52] o=root 26065135 26065135 IN IP4 192.168.23.150
[Nov 5 09:41:52] s=Asterisk PBX 13.38.2-vici
[Nov 5 09:41:52] c=IN IP4 192.168.23.150
[Nov 5 09:41:52] t=0 0
[Nov 5 09:41:52] m=audio 19564 RTP/AVP 8 0 3 101
[Nov 5 09:41:52] a=rtpmap:8 PCMA/8000
[Nov 5 09:41:52] a=rtpmap:0 PCMU/8000
[Nov 5 09:41:52] a=rtpmap:3 GSM/8000
[Nov 5 09:41:52] a=rtpmap:101 telephone-event/8000
[Nov 5 09:41:52] a=fmtp:101 0-16
[Nov 5 09:41:52] a=ptime:20
[Nov 5 09:41:52] a=maxptime:150
[Nov 5 09:41:52] a=sendrecv
[Nov 5 09:41:52]
[Nov 5 09:41:52] ---
[Nov 5 09:41:52]
[Nov 5 09:41:52] <--- SIP read from UDP:192.168.23.155:54836 --->
[Nov 5 09:41:52] SIP/2.0 180 Ringing
[Nov 5 09:41:52] Via: SIP/2.0/UDP 192.168.23.150:5060;branch=z9hG4bK19086955;rport=5060
[Nov 5 09:41:52] Contact: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>
[Nov 5 09:41:52] To: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>;tag=7d6eb548
[Nov 5 09:41:52] From: "DC098096W0000000051W"<sip:302170004582@192.168.23.150>;tag=as1c8fc7dc
[Nov 5 09:41:52] Call-ID: 4d5a383f641a35bc7b0983006edf4e97@192.168.23.150:5060
[Nov 5 09:41:52] CSeq: 102 INVITE
[Nov 5 09:41:52] User-Agent: X-Lite release 1011s stamp 41150
[Nov 5 09:41:52] Content-Length: 0
[Nov 5 09:41:52]
[Nov 5 09:41:52] <------------->
[Nov 5 09:41:52] --- (9 headers 0 lines) ---
[Nov 5 09:41:52] sip_route_dump: route/path hop: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>
[Nov 5 09:41:52] -- SIP/251-00000011 is ringing
[Nov 5 09:41:52] -- Local/251@default-00000010;1 is ringing
[Nov 5 09:41:52]
[Nov 5 09:41:52] <--- SIP read from UDP:192.168.23.155:54836 --->
[Nov 5 09:41:52] SIP/2.0 180 Ringing
[Nov 5 09:41:52] Via: SIP/2.0/UDP 192.168.23.150:5060;branch=z9hG4bK19086955;rport=5060
[Nov 5 09:41:52] Contact: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>
[Nov 5 09:41:52] To: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>;tag=7d6eb548
[Nov 5 09:41:52] From: "DC098096W0000000051W"<sip:302170004582@192.168.23.150>;tag=as1c8fc7dc
[Nov 5 09:41:52] Call-ID: 4d5a383f641a35bc7b0983006edf4e97@192.168.23.150:5060
[Nov 5 09:41:52] CSeq: 102 INVITE
[Nov 5 09:41:52] User-Agent: X-Lite release 1011s stamp 41150
[Nov 5 09:41:52] Content-Length: 0
[Nov 5 09:41:52]
[Nov 5 09:41:52] <------------->
[Nov 5 09:41:52] --- (9 headers 0 lines) ---
[Nov 5 09:41:52] sip_route_dump: route/path hop: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>
[Nov 5 09:41:52] -- SIP/251-00000011 is ringing
[Nov 5 09:41:53]
[Nov 5 09:41:53] <--- SIP read from UDP:192.168.23.155:54836 --->
[Nov 5 09:41:53] SIP/2.0 200 OK
[Nov 5 09:41:53] Via: SIP/2.0/UDP 192.168.23.150:5060;branch=z9hG4bK19086955;rport=5060
[Nov 5 09:41:53] Contact: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>
[Nov 5 09:41:53] To: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>;tag=7d6eb548
[Nov 5 09:41:53] From: "DC098096W0000000051W"<sip:302170004582@192.168.23.150>;tag=as1c8fc7dc
[Nov 5 09:41:53] Call-ID: 4d5a383f641a35bc7b0983006edf4e97@192.168.23.150:5060
[Nov 5 09:41:53] CSeq: 102 INVITE
[Nov 5 09:41:53] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
[Nov 5 09:41:53] Content-Type: application/sdp
[Nov 5 09:41:53] User-Agent: X-Lite release 1011s stamp 41150
[Nov 5 09:41:53] Content-Length: 188
[Nov 5 09:41:53]
[Nov 5 09:41:53] v=0
[Nov 5 09:41:53] o=- 7 2 IN IP4 192.168.23.155
[Nov 5 09:41:53] s=CounterPath X-Lite 3.0
[Nov 5 09:41:53] c=IN IP4 192.168.23.155
[Nov 5 09:41:53] t=0 0
[Nov 5 09:41:53] m=audio 1632 RTP/AVP 8 0 101
[Nov 5 09:41:53] a=fmtp:101 0-15
[Nov 5 09:41:53] a=rtpmap:101 telephone-event/8000
[Nov 5 09:41:53] a=sendrecv
[Nov 5 09:41:53] <------------->
[Nov 5 09:41:53] --- (11 headers 9 lines) ---
[Nov 5 09:41:53] Got SDP version 2 and unique parts [- 7 IN IP4 192.168.23.155]
[Nov 5 09:41:53] Found RTP audio format 8
[Nov 5 09:41:53] Found RTP audio format 0
[Nov 5 09:41:53] Found RTP audio format 101
[Nov 5 09:41:53] Found audio description format telephone-event for ID 101
[Nov 5 09:41:53] Capabilities: us - (alaw|ulaw|gsm), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (alaw|ulaw)
[Nov 5 09:41:53] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Nov 5 09:41:53] > 0x7fe934072ed0 -- Strict RTP learning after remote address set to: 192.168.23.155:1632
[Nov 5 09:41:53] Peer audio RTP is at port 192.168.23.155:1632
[Nov 5 09:41:53] sip_route_dump: route/path hop: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>
[Nov 5 09:41:53] Transmitting (NAT) to 192.168.23.155:54836:
[Nov 5 09:41:53] ACK sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663 SIP/2.0
[Nov 5 09:41:53] Via: SIP/2.0/UDP 192.168.23.150:5060;branch=z9hG4bK49697fe1;rport
[Nov 5 09:41:53] Max-Forwards: 70
[Nov 5 09:41:53] From: "DC098096W0000000051W" <sip:302170004582@192.168.23.150>;tag=as1c8fc7dc
[Nov 5 09:41:53] To: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>;tag=7d6eb548
[Nov 5 09:41:53] Contact: <sip:302170004582@192.168.23.150:5060>
[Nov 5 09:41:53] Call-ID: 4d5a383f641a35bc7b0983006edf4e97@192.168.23.150:5060
[Nov 5 09:41:53] CSeq: 102 ACK
[Nov 5 09:41:53] User-Agent: Asterisk PBX 13.38.2-vici
[Nov 5 09:41:53] Content-Length: 0
[Nov 5 09:41:53]
[Nov 5 09:41:53]
[Nov 5 09:41:53] ---
[Nov 5 09:41:53] -- SIP/251-00000011 answered Local/251@default-00000010;2
[Nov 5 09:41:53] -- Local/251@default-00000010;1 answered
[Nov 5 09:41:53] -- Executing [8600051@default:1] MeetMe("Local/251@default-00000010;1", "8600051,F") in new stack
[Nov 5 09:41:53] -- Channel SIP/251-00000011 joined 'simple_bridge' basic-bridge <c0e39aa4-2039-45b9-8245-f9e5e25be48f>
[Nov 5 09:41:53] > 0x7fe934072ed0 -- Strict RTP switching to RTP target address 192.168.23.155:1632 as source
[Nov 5 09:41:53] -- Channel Local/251@default-00000010;2 joined 'simple_bridge' basic-bridge <c0e39aa4-2039-45b9-8245-f9e5e25be48f>
[Nov 5 09:41:54] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 5 09:41:55] == Manager 'sendcron' logged on from 127.0.0.1
[Nov 5 09:41:55] -- Manager 'sendcron' from 127.0.0.1, hanging up channel: SIP/251-00000011
[Nov 5 09:41:55] -- Channel SIP/251-00000011 left 'simple_bridge' basic-bridge <c0e39aa4-2039-45b9-8245-f9e5e25be48f>
[Nov 5 09:41:55] Scheduling destruction of SIP dialog '4d5a383f641a35bc7b0983006edf4e97@192.168.23.150:5060' in 6528 ms (Method: INVITE)
[Nov 5 09:41:55] Reliably Transmitting (NAT) to 192.168.23.155:54836:
[Nov 5 09:41:55] BYE sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663 SIP/2.0
[Nov 5 09:41:55] Via: SIP/2.0/UDP 192.168.23.150:5060;branch=z9hG4bK1bce9d44;rport
[Nov 5 09:41:55] Max-Forwards: 70
[Nov 5 09:41:55] From: "DC098096W0000000051W" <sip:302170004582@192.168.23.150>;tag=as1c8fc7dc
[Nov 5 09:41:55] To: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>;tag=7d6eb548
[Nov 5 09:41:55] Call-ID: 4d5a383f641a35bc7b0983006edf4e97@192.168.23.150:5060
[Nov 5 09:41:55] CSeq: 103 BYE
[Nov 5 09:41:55] User-Agent: Asterisk PBX 13.38.2-vici
[Nov 5 09:41:55] X-Asterisk-HangupCause: Normal Clearing
[Nov 5 09:41:55] X-Asterisk-HangupCauseCode: 16
[Nov 5 09:41:55] Content-Length: 0
[Nov 5 09:41:55]
[Nov 5 09:41:55]
[Nov 5 09:41:55] ---
[Nov 5 09:41:55] -- Channel Local/251@default-00000010;2 left 'simple_bridge' basic-bridge <c0e39aa4-2039-45b9-8245-f9e5e25be48f>
[Nov 5 09:41:55] == Spawn extension (default, 251, 1) exited non-zero on 'Local/251@default-00000010;2'
[Nov 5 09:41:55] -- Executing [h@default:1] AGI("Local/251@default-00000010;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----3-----2-----SIP 200 OK)") in new stack
[Nov 5 09:41:55] -- <Local/251@default-00000010;2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----3-----2-----SIP 200 OK) completed, returning 0
[Nov 5 09:41:55] == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/251@default-00000010;1'
[Nov 5 09:41:55] WARNING[17873][C-00000026]: func_hangupcause.c:140 hangupcause_read: Unable to find information for channel
[Nov 5 09:41:55] -- Executing [h@default:1] AGI("Local/251@default-00000010;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------)") in new stack
[Nov 5 09:41:55] -- <Local/251@default-00000010;1>AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) completed, returning 0
[Nov 5 09:41:55] Retransmitting #1 (NAT) to 192.168.23.155:54836:
[Nov 5 09:41:55] BYE sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663 SIP/2.0
[Nov 5 09:41:55] Via: SIP/2.0/UDP 192.168.23.150:5060;branch=z9hG4bK1bce9d44;rport
[Nov 5 09:41:55] Max-Forwards: 70
[Nov 5 09:41:55] From: "DC098096W0000000051W" <sip:302170004582@192.168.23.150>;tag=as1c8fc7dc
[Nov 5 09:41:55] To: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>;tag=7d6eb548
[Nov 5 09:41:55] Call-ID: 4d5a383f641a35bc7b0983006edf4e97@192.168.23.150:5060
[Nov 5 09:41:55] CSeq: 103 BYE
[Nov 5 09:41:55] User-Agent: Asterisk PBX 13.38.2-vici
[Nov 5 09:41:55] X-Asterisk-HangupCause: Normal Clearing
[Nov 5 09:41:55] X-Asterisk-HangupCauseCode: 16
[Nov 5 09:41:55] Content-Length: 0
[Nov 5 09:41:55]
[Nov 5 09:41:55]
[Nov 5 09:41:55] ---
[Nov 5 09:41:55]
[Nov 5 09:41:55] <--- SIP read from UDP:192.168.23.155:54836 --->
[Nov 5 09:41:55] SIP/2.0 200 OK
[Nov 5 09:41:55] Via: SIP/2.0/UDP 192.168.23.150:5060;branch=z9hG4bK1bce9d44;rport=5060
[Nov 5 09:41:55] Contact: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>
[Nov 5 09:41:55] To: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>;tag=7d6eb548
[Nov 5 09:41:55] From: "DC098096W0000000051W"<sip:302170004582@192.168.23.150>;tag=as1c8fc7dc
[Nov 5 09:41:55] Call-ID: 4d5a383f641a35bc7b0983006edf4e97@192.168.23.150:5060
[Nov 5 09:41:55] CSeq: 103 BYE
[Nov 5 09:41:55] User-Agent: X-Lite release 1011s stamp 41150
[Nov 5 09:41:55] Content-Length: 0
[Nov 5 09:41:55]
[Nov 5 09:41:55] <------------->
[Nov 5 09:41:55] --- (9 headers 0 lines) ---
[Nov 5 09:41:55] Really destroying SIP dialog '4d5a383f641a35bc7b0983006edf4e97@192.168.23.150:5060' Method: INVITE
[Nov 5 09:41:55]
[Nov 5 09:41:55] <--- SIP read from UDP:192.168.23.155:54836 --->
[Nov 5 09:41:55] SIP/2.0 200 OK
[Nov 5 09:41:55] Via: SIP/2.0/UDP 192.168.23.150:5060;branch=z9hG4bK1bce9d44;rport=5060
[Nov 5 09:41:55] Contact: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>
[Nov 5 09:41:55] To: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>;tag=7d6eb548
[Nov 5 09:41:55] From: "DC098096W0000000051W"<sip:302170004582@192.168.23.150>;tag=as1c8fc7dc
[Nov 5 09:41:55] Call-ID: 4d5a383f641a35bc7b0983006edf4e97@192.168.23.150:5060
[Nov 5 09:41:55] CSeq: 103 BYE
[Nov 5 09:41:55] User-Agent: X-Lite release 1011s stamp 41150
[Nov 5 09:41:55] Content-Length: 0
[Nov 5 09:41:55]
[Nov 5 09:41:55] <------------->
[Nov 5 09:41:55] --- (9 headers 0 lines) ---
[Nov 5 09:41:56] == Manager 'sendcron' logged off from 127.0.0.1
[Nov 5 09:41:57] Reliably Transmitting (NAT) to 192.168.23.155:54836:
[Nov 5 09:41:57] OPTIONS sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663 SIP/2.0
[Nov 5 09:41:57] Via: SIP/2.0/UDP 192.168.23.150:5060;branch=z9hG4bK59e434cc;rport
[Nov 5 09:41:57] Max-Forwards: 70
[Nov 5 09:41:57] From: "asterisk" <sip:asterisk@192.168.23.150>;tag=as06aeed11
[Nov 5 09:41:57] To: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>
[Nov 5 09:41:57] Contact: <sip:asterisk@192.168.23.150:5060>
[Nov 5 09:41:57] Call-ID: 47a04a474adae2f137563210348b8933@192.168.23.150:5060
[Nov 5 09:41:57] CSeq: 102 OPTIONS
[Nov 5 09:41:57] User-Agent: Asterisk PBX 13.38.2-vici
[Nov 5 09:41:57] Date: Fri, 05 Nov 2021 07:41:57 GMT
[Nov 5 09:41:57] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[Nov 5 09:41:57] Supported: replaces, timer
[Nov 5 09:41:57] Content-Length: 0
[Nov 5 09:41:57]
[Nov 5 09:41:57]
[Nov 5 09:41:57] ---
[Nov 5 09:41:57]
[Nov 5 09:41:57] <--- SIP read from UDP:192.168.23.155:54836 --->
[Nov 5 09:41:57] SIP/2.0 200 OK
[Nov 5 09:41:57] Via: SIP/2.0/UDP 192.168.23.150:5060;branch=z9hG4bK59e434cc;rport=5060
[Nov 5 09:41:57] Contact: <sip:192.168.23.155:54836>
[Nov 5 09:41:57] To: <sip:251@192.168.23.155:54836;rinstance=10a5943c1affa663>;tag=836cc30a
[Nov 5 09:41:57] From: "asterisk"<sip:asterisk@192.168.23.150>;tag=as06aeed11
[Nov 5 09:41:57] Call-ID: 47a04a474adae2f137563210348b8933@192.168.23.150:5060
[Nov 5 09:41:57] CSeq: 102 OPTIONS
[Nov 5 09:41:57] Accept: application/sdp
[Nov 5 09:41:57] Accept-Language: en
[Nov 5 09:41:57] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
[Nov 5 09:41:57] User-Agent: X-Lite release 1011s stamp 41150
[Nov 5 09:41:57] Content-Length: 0
[Nov 5 09:41:57]
[Nov 5 09:41:57] <------------->
[Nov 5 09:41:57] --- (12 headers 0 lines) ---
[Nov 5 09:41:57] Really destroying SIP dialog '47a04a474adae2f137563210348b8933@192.168.23.150:5060' Method: OPTIONS
[Nov 5 09:41:57]
[Nov 5 09:41:57] <--- SIP read from UDP:192.168.23.155:54836 --->
[Nov 5 09:41:57]
[Nov 5 09:41:57]
[Nov 5 09:41:57] <------------->
vicibox10*CLI> sip set debug off
SIP Debugging Disabled
vicibox10*CLI> exit

by **ambiorixg12** » Fri Nov 05, 2021 1:10 pm

In this case a SIP trace won't help as there is no issue on the SIP session establishment, as you know the issue is manager hangup a channel, and that action for asterisk is Normal Clearing

[Nov 5 09:41:55] X-Asterisk-HangupCause: Normal Clearing

by **callcentertech** » Mon Dec 13, 2021 3:28 pm

Is it fixed yet..??

Try updating to latest SVN and see if that issue goes away..

by **antuan** » Wed Dec 15, 2021 6:58 am

callcentertech wrote:Is it fixed yet..??

Try updating to latest SVN and see if that issue goes away..

Hello,
i have upgraded to 3545 and did some calls but the problem still exists, internal extension is ringing but when someone picks it up it instantly hangs up.

vicidial.org

3way conference

3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Re: 3way conference

Who is online