Webphone auto-disconnect after a call

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Webphone auto-disconnect after a call

Postby JAGT » Mon Jun 29, 2020 8:04 am

Hi everyone,

Please send help. We suddenly experience this issue, agents cant hear a ring after a manual call. When you logged in, there is no issue at all but after doing a manual call it suddenly disconnect on session. That's why you need to login again to connect to phone which is not appropriate. I already change rtpholdtimeout to 300 and nothing happend. Is there something to do with session or conference? Thanks in advance!

Im using vicibox9 and asterisk version 13.29.2-vici.

Here's cli
Code: Select all

[Jun 29 07:51:10]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 29 07:51:10]     -- Called 55558600052@default
[Jun 29 07:51:10]     -- Executing [55558600052@default:1] MeetMeAdmin("Local/55 558600052@default-00000018;2", "8600052,K") in new stack
[Jun 29 07:51:10] WARNING[14486][C-00000039]: app_meetme.c:5261 admin_exec: Conf erence number '8600052' not found!
[Jun 29 07:51:10]     -- Executing [55558600052@default:2] Hangup("Local/5555860 0052@default-00000018;2", "") in new stack
[Jun 29 07:51:10]   == Spawn extension (default, 55558600052, 2) exited non-zero  on 'Local/55558600052@default-00000018;2'
[Jun 29 07:51:10] WARNING[14486][C-00000039]: func_hangupcause.c:140 hangupcause _read: Unable to find information for channel
[Jun 29 07:51:10]     -- Executing [h@default:1] AGI("Local/55558600052@default- 00000018;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--- -----------------)") in new stack
[Jun 29 07:51:10]     -- <Local/55558600052@default-00000018;2>AGI Script agi:// 127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) c ompleted, returning 0
[Jun 29 07:51:11]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 29 07:51:11]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 29 07:51:15]   == WebSocket connection from '192.168.1.10:9740' for protoco l 'sip' accepted using version '13'
[Jun 29 07:51:15]     -- Registered SIP '50182129' at 192.168.1.10:9740
[Jun 29 07:51:27]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 29 07:51:27] ERROR[14509]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo ("vicibox9", "(null)", ...): Name or service not known
[Jun 29 07:51:27] WARNING[14509]: acl.c:835 resolve_first: Unable to lookup 'vic ibox9'
[Jun 29 07:51:27]   == Using SIP RTP CoS mark 5
[Jun 29 07:51:27]     -- Called 50182129
[Jun 29 07:51:27]     -- SIP/50182129-0000001f is ringing
[Jun 29 07:51:27]        > 0x7fc450028160 -- Strict RTP learning after remote ad dress set to: 192.168.1.180:60047
[Jun 29 07:51:27]     -- SIP/50182129-0000001f answered
[Jun 29 07:51:27]     -- Executing [8600052@default:1] MeetMe("SIP/50182129-0000001f ", "8600052,F") in new stack
[Jun 29 07:51:27]     -- Created MeetMe conference 1023 for conference '8600052'
[Jun 29 07:51:27]     -- <SIP/5011-0000001f> Playing 'conf-onlyperson.gsm' (lang uage 'en')
[Jun 29 07:51:27]        > 0x7fc450028160 -- Strict RTP learning after ICE compl etion
[Jun 29 07:51:27]        > 0x7fc450028160 -- Strict RTP switching to RTP target  address 192.168.1.180:60047 as source
[Jun 29 07:51:28]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 29 07:51:32]        > 0x7fc450028160 -- Strict RTP learning complete - Lock ing on source address 192.168.1.180:60047
[Jun 29 07:51:45]   == WebSocket connection from '192.168.1.10:9722' closed
[Jun 29 07:51:45]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 29 07:51:45]     -- Called 8600052@default
[Jun 29 07:51:45]     -- Executing [8600052@default:1] MeetMe("Local/8600052@def ault-00000019;2", "8600052,F") in new stack
[Jun 29 07:51:45]     -- Local/8600052@default-00000019;1 answered
[Jun 29 07:51:45]     -- Executing [718662559215@default:1] AGI("Local/8600052@d efault-00000019;1", "agi://127.0.0.1:4577/call_log") in new stack
[Jun 29 07:51:45]     -- AGI Script Executing Application: (EXEC) Options: (Set( _CAMPCUST=NANOCBD))
[Jun 29 07:51:45]     -- <Local/8600052@default-00000019;1>AGI Script agi://127. 0.0.1:4577/call_log completed, returning 0
[Jun 29 07:51:45]     -- Executing [718662559215@default:2] Dial("Local/8600052@ default-00000019;1", "SIP/18662559215@nexvortex,,Tor") in new stack
[Jun 29 07:51:45]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 29 07:51:45]     -- Called 58600052@default
[Jun 29 07:51:45]     -- Executing [58600052@default:1] MeetMe("Local/58600052@d efault-0000001a;2", "8600052,Fmq") in new stack
[Jun 29 07:51:45]     -- Local/58600052@default-0000001a;1 answered
[Jun 29 07:51:45]     -- Executing [8309@default:1] Answer("Local/58600052@defau lt-0000001a;1", "") in new stack
[Jun 29 07:51:45]     -- Executing [8309@default:2] Monitor("Local/58600052@defa ult-0000001a;1", "wav,NANOCBD_20200629-075144_18662559215_50182129") in new stack
[Jun 29 07:51:45]     -- Executing [8309@default:3] Wait("Local/58600052@default -0000001a;1", "3600") in new stack
[Jun 29 07:51:45] ERROR[14538][C-0000003c]: netsock2.c:305 ast_sockaddr_resolve:  getaddrinfo("vicibox9", "(null)", ...): Name or service not known
[Jun 29 07:51:45] WARNING[14538][C-0000003c]: acl.c:835 resolve_first: Unable to  lookup 'vicibox9'
[Jun 29 07:51:45]   == Using SIP RTP CoS mark 5
[Jun 29 07:51:45]     -- Called SIP/18662559215@nexvortex
[Jun 29 07:51:46]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 29 07:51:46]   == Manager 'sendcron' logged off from 127.0.0.1
[Jun 29 07:51:47]        > 0x7fc4540114c0 -- Strict RTP learning after remote ad dress set to: 8.26.79.38:38208
[Jun 29 07:51:47]     -- SIP/nexvortex-00000020 is making progress passing it to  Local/8600052@default-00000019;1
[Jun 29 07:51:47]     -- SIP/nexvortex-00000020 answered Local/8600052@default-0 0000019;1
[Jun 29 07:51:47]     -- Channel SIP/nexvortex-00000020 joined 'simple_bridge' b asic-bridge <211cf631-1d55-49ee-8d24-b48dd978028a>
[Jun 29 07:51:47]     -- Channel Local/8600052@default-00000019;1 joined 'simple _bridge' basic-bridge <211cf631-1d55-49ee-8d24-b48dd978028a>
[Jun 29 07:51:47]        > 0x7fc4540114c0 -- Strict RTP switching to RTP target  address 8.26.79.38:38208 as source
[Jun 29 07:51:49]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 29 07:51:49]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel:  SIP/nexvortex-00000020
[Jun 29 07:51:49]     -- Channel SIP/nexvortex-00000020 left 'simple_bridge' bas ic-bridge <211cf631-1d55-49ee-8d24-b48dd978028a>
[Jun 29 07:51:49]     -- Channel Local/8600052@default-00000019;1 left 'simple_b ridge' basic-bridge <211cf631-1d55-49ee-8d24-b48dd978028a>
[Jun 29 07:51:49]   == Spawn extension (default, 718662559215, 2) exited non-zer o on 'Local/8600052@default-00000019;1'
[Jun 29 07:51:49]     -- Executing [h@default:1] AGI("Local/8600052@default-0000 0019;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----AN SWER-----4-----4-----SIP 200 OK)") in new stack
[Jun 29 07:51:49]     -- <Local/8600052@default-00000019;1>AGI Script agi://127. 0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----4-----4---- -SIP 200 OK) completed, returning 0
[Jun 29 07:51:49]   == Spawn extension (default, 8600052, 1) exited non-zero on  'Local/8600052@default-00000019;2'
[Jun 29 07:51:49] WARNING[14539][C-0000003b]: func_hangupcause.c:140 hangupcause _read: Unable to find information for channel
[Jun 29 07:51:49]     -- Executing [h@default:1] AGI("Local/8600052@default-0000 0019;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16------- -------------)") in new stack
[Jun 29 07:51:49]     -- <Local/8600052@default-00000019;2>AGI Script agi://127. 0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------) compl eted, returning 0
[Jun 29 07:51:49]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 29 07:51:49]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel:  SIP/50182129-0000001f
[Jun 29 07:51:49]   == Spawn extension (default, 8600052, 1) exited non-zero on  'SIP/50182129-0000001f'
[Jun 29 07:51:49]     -- Executing [h@default:1] AGI("SIP/50182129-0000001f", "agi:/ /127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16--------------------SI P 200 OK)") in new stack
[Jun 29 07:51:49]     -- <SIP/50182129-0000001f>AGI Script agi://127.0.0.1:4577/call _log--HVcauses--PRI-----NODEBUG-----16--------------------SIP 200 OK) completed,  returning 0
[Jun 29 07:51:49]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 29 07:51:49] NOTICE[14560]: manager.c:4458 action_hangup: Request to hangup  non-existent channel: Local/8600052@default-00000019;2
[Jun 29 07:51:49]   == Manager 'sendcron' logged on from 127.0.0.1
[Jun 29 07:51:49]     -- Manager 'sendcron' from 127.0.0.1, hanging up channel:  Local/58600052@default-0000001a;2
[Jun 29 07:51:49]     -- Hungup 'DAHDI/pseudo-156533489'
[Jun 29 07:51:49]   == Spawn extension (default, 58600052, 1) exited non-zero on  'Local/58600052@default-0000001a;2'
[Jun 29 07:51:49] WARNING[14543][C-0000003d]: func_hangupcause.c:140 hangupcause _read: Unable to find information for channel
[Jun 29 07:51:49]     -- Executing [h@default:1] AGI("Local/58600052@default-000 0001a;2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0------- -------------)") in new stack
[Jun 29 07:51:49]     -- <Local/58600052@default-0000001a;2>AGI Script agi://127 .0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) compl eted, returning 0
[Jun 29 07:51:49]   == Spawn extension (default, 8309, 3) exited non-zero on 'Lo cal/58600052@default-0000001a;1'
[Jun 29 07:51:49] WARNING[14542][C-0000003e]: func_hangupcause.c:140 hangupcause _read: Unable to find information for channel
[Jun 29 07:51:49]     -- Executing [h@default:1] AGI("Local/58600052@default-000 0001a;1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0------- -------------)") in new stack
[Jun 29 07:51:49]     -- <Local/58600052@default-0000001a;1>AGI Script agi://127 .0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------------) compl eted, returning 0
[Jun 29 07:51:50]   == Manager 'sendcron' logged off from 127.0.0.1
127.0.0.1

JAGT
 
Posts: 17
Joined: Thu Feb 27, 2020 1:53 pm

Re: Webphone auto-disconnect after a call

Postby carpenox » Mon Jun 29, 2020 12:41 pm

check if dahdi is running, on the server type this command:

dahdi_cfg -vvv

If its not then do this:

modprobe dahdi
Alma Linux 9.3 | Version: 2.14-911a | SVN Version: 3815 | DB Schema Version: 1710 | Asterisk 18.18.1
www.dialer.one -:- 1-833-DIALER-1 -:- https://linktr.ee/CyburDial -:- WhatsApp: +19549477572 -:- Skype: live:carpenox_3
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Re: Webphone auto-disconnect after a call

Postby JAGT » Mon Jun 29, 2020 10:07 pm

Here's the output. Is it normal?

Code: Select all

vicibox9:~ # dahdi_cfg -vvv
DAHDI Tools Version - 3.1.0

DAHDI Version: 3.1.0
Echo Canceller(s):
Configuration
======================


Channel map:


0 channels to configure.


JAGT
 
Posts: 17
Joined: Thu Feb 27, 2020 1:53 pm

Re: Webphone auto-disconnect after a call

Postby carpenox » Tue Jun 30, 2020 7:17 am

Yes that is correct. Try installing this webphone https://codeload.github.com/ccabrerar/V ... zip/master
Alma Linux 9.3 | Version: 2.14-911a | SVN Version: 3815 | DB Schema Version: 1710 | Asterisk 18.18.1
www.dialer.one -:- 1-833-DIALER-1 -:- https://linktr.ee/CyburDial -:- WhatsApp: +19549477572 -:- Skype: live:carpenox_3
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Re: Webphone auto-disconnect after a call

Postby ambiorixg12 » Tue Jun 30, 2020 8:29 pm

Your logs doesnt display any hangup related to RTP timeout , What I see the sessions channel is terminated directly by an

AMI request [Jun 29 07:51:49] -- Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/58600052@default-0000001a;2
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Re: Webphone auto-disconnect after a call

Postby carpenox » Tue Jun 30, 2020 9:24 pm

The neetme conference was never created which is why I thought is was dahdi related.
Alma Linux 9.3 | Version: 2.14-911a | SVN Version: 3815 | DB Schema Version: 1710 | Asterisk 18.18.1
www.dialer.one -:- 1-833-DIALER-1 -:- https://linktr.ee/CyburDial -:- WhatsApp: +19549477572 -:- Skype: live:carpenox_3
carpenox
 
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Location: Coral Springs, FL

Re: Webphone auto-disconnect after a call

Postby JAGT » Wed Jul 01, 2020 1:17 am

Okay @carpenox, I will try this webphone and update you later. By the way, thanks for the reply. I appreciate it so much.
JAGT
 
Posts: 17
Joined: Thu Feb 27, 2020 1:53 pm

Re: Webphone auto-disconnect after a call

Postby JAGT » Wed Jul 01, 2020 1:20 am

ambiorixg12 wrote:Your logs doesnt display any hangup related to RTP timeout , What I see the sessions channel is terminated directly by an

AMI request [Jun 29 07:51:49] -- Manager 'sendcron' from 127.0.0.1, hanging up channel: Local/58600052@default-0000001a;2


What are the factors that terminate the sessions channel? Sorry, I am clueless regarding this matter. I appreciate your help. Thank you!
JAGT
 
Posts: 17
Joined: Thu Feb 27, 2020 1:53 pm

Re: Webphone auto-disconnect after a call

Postby carpenox » Wed Jul 01, 2020 8:23 am

It was attempting to stop a channel that was never created to begin with

[Jun 29 07:51:10] WARNING[14486][C-00000039]: app_meetme.c:5261 admin_exec: Conference number '8600052' not found!

Do you have your vicidial conferences created correctly? This issue was something similar I had when I tried the webphone which is why i suggested to try the one I linked you too which fixed the problem I was having.
Alma Linux 9.3 | Version: 2.14-911a | SVN Version: 3815 | DB Schema Version: 1710 | Asterisk 18.18.1
www.dialer.one -:- 1-833-DIALER-1 -:- https://linktr.ee/CyburDial -:- WhatsApp: +19549477572 -:- Skype: live:carpenox_3
carpenox
 
Posts: 2230
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Location: Coral Springs, FL

Re: Webphone auto-disconnect after a call

Postby JAGT » Tue Jul 07, 2020 4:22 am

Hi guys! Sorry if it takes days for me to reply. Just to give you an update, I was instructed to rebuild or reinstall it. I created a bootable drive using the ViciBox_v9.x86_64-9.0.3-md.iso. But I'm stuck on a new problem which is time synchronization. Can you guide me on this?
JAGT
 
Posts: 17
Joined: Thu Feb 27, 2020 1:53 pm

Re: Webphone auto-disconnect after a call

Postby carpenox » Tue Jul 07, 2020 4:29 pm

what sockets are running? use the command "screen -ls" and report back what you have.
Alma Linux 9.3 | Version: 2.14-911a | SVN Version: 3815 | DB Schema Version: 1710 | Asterisk 18.18.1
www.dialer.one -:- 1-833-DIALER-1 -:- https://linktr.ee/CyburDial -:- WhatsApp: +19549477572 -:- Skype: live:carpenox_3
carpenox
 
Posts: 2230
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