Calls are dropped after 60 seconds

[Joss2103]

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>


[Kabis]

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.

[carpenox]

change your rtpkeelalive to 30 and your rtptimeout to 600 on sip.conf and reload

[Joss2103]

Thank you

I made the changes on sip.conf and the problem is solved

[carpenox]

no problem, glad to help