Calls are dropped after 60 seconds

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Calls are dropped after 60 seconds

Postby Joss2103 » Mon Apr 18, 2022 4:37 pm

VERSION: 2.14-848a
BUILD: 220218-1656
Asterisk 16.20.0-vici
SO: openSUSE Leap 15.3

Hi, I have a vicidial installed in clusters, a server for asterisk, another for the dialer, and another one for the db.

When I make a call manually outside from the agent screen from an extension the call is dropped after 60 seconds and there isn´t any sound. The calls made from the agent screen work fine. I disabled the firewall but the problem continues.

This is the console log

Code: Select all
 == Using SIP RTP CoS mark 5
[Apr 18 13:19:16]        > 0x7f2de80821f0 -- Strict RTP learning after remote address set to: XXX.XXX.XXX.XX:50236
[Apr 18 13:19:16]     -- Executing [441234567890@default:1] AGI("SIP/1000-00000008", "agi://127.0.0.1:4577/call_log") in new stack
[Apr 18 13:19:16]     -- <SIP/1000-00000008>AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Apr 18 13:19:16]     -- Executing [441234567890@default:2] Dial("SIP/1000-00000008", "SIP/SipTrunk/1234567890,,To") in new stack
[Apr 18 13:19:16]   == Using SIP RTP CoS mark 5
[Apr 18 13:19:16]     -- Called SIP/SipTrunk/1234567890
[Apr 18 13:19:17]        > 0x7f2de003d410 -- Strict RTP learning after remote address set to: XXX.XXX.XXX.XX:13270
[Apr 18 13:19:17]     -- SIP/SipTrunk-00000009 is making progress passing it to SIP/1000-00000008
[Apr 18 13:19:24]        > 0x7f2de003d410 -- Strict RTP learning after remote address set to: XXX.XXX.XXX.XX:13270
[Apr 18 13:19:24]     -- SIP/PruebaTrunk-00000009 answered SIP/1000-00000008
[Apr 18 13:19:24]     -- Channel SIP/SipTrunk-00000009 joined 'simple_bridge' basic-bridge <b6bd3c83-ba5c-49da-943b-57ae006faa57>
[Apr 18 13:19:24]     -- Channel SIP/1000-00000008 joined 'simple_bridge' basic-bridge <b6bd3c83-ba5c-49da-943b-57ae006faa57>
Joss2103
 
Posts: 88
Joined: Wed Jul 22, 2020 3:20 pm

Re: Calls are dropped after 60 seconds

Postby Kabis » Wed Apr 20, 2022 10:26 am

Hi,

It might be NAT issue or RTP Issue. Check You opened RTP Ports(10000-20000) in your firewall routes. Or check with Packet capture you can identify.
We are ready to help you,
Regards,
KABIS,
Email ID: kabisforu@gmail.com
Website: www.kabis.org.in
Skype: kabisforu
Kabis
 
Posts: 98
Joined: Sat Nov 03, 2018 12:08 am
Location: India

Re: Calls are dropped after 60 seconds

Postby carpenox » Thu Apr 21, 2022 10:17 am

change your rtpkeelalive to 30 and your rtptimeout to 600 on sip.conf and reload
Alma Linux 8.5 | Version: 2.14-858a | BUILD: 220513-0819 | SVN Version: 3602 | DB Schema Version: 1661 | Asterisk 16.17.0-vici
www.CyburDial.net -:- 725-22-CYBUR -:- My Blog: http://vicidial.blog -:- Whatsapp: +19549477572 -:- Skype: live:carpenox_3
carpenox
 
Posts: 1848
Joined: Wed Apr 08, 2020 2:02 am
Location: Coral Springs, FL

Re: Calls are dropped after 60 seconds

Postby Joss2103 » Fri Apr 22, 2022 10:02 am

Thank you

I made the changes on sip.conf and the problem is solved :)
Joss2103
 
Posts: 88
Joined: Wed Jul 22, 2020 3:20 pm

Re: Calls are dropped after 60 seconds

Postby carpenox » Sat Apr 23, 2022 3:41 pm

no problem, glad to help
Alma Linux 8.5 | Version: 2.14-858a | BUILD: 220513-0819 | SVN Version: 3602 | DB Schema Version: 1661 | Asterisk 16.17.0-vici
www.CyburDial.net -:- 725-22-CYBUR -:- My Blog: http://vicidial.blog -:- Whatsapp: +19549477572 -:- Skype: live:carpenox_3
carpenox
 
Posts: 1848
Joined: Wed Apr 08, 2020 2:02 am
Location: Coral Springs, FL


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