when using asterisk real time database, when user A disconnects the call, the call is not getting terminated in user B.
The call gets disconneted properly within a minute. The issue occurs after a minute.
This issue does not occur when using static sip.conf file.
You’ve provided no accompanying logging for this, which would be needed.
which log do I need to share? Is it rtp set debug on
Connected to Asterisk 18.18.0 currently running on ngdcs (pid = 1514)
– Remote UNIX connection
– Registered SIP ‘9962’ at 10.203.0.6:42256
– Unregistered SIP ‘101001’
== WebSocket connection from ‘10.203.0.6:42260’ for protocol ‘sip’ accepted using version ‘13’
– Registered SIP ‘101001’ at 10.203.0.6:42260
== WebSocket connection from ‘10.203.0.6:42542’ for protocol ‘sip’ accepted using version ‘13’
[Oct 9 16:30:48] ERROR[2778]: res_http_websocket.c:567 ws_safe_read: Error reading from web socket: Success
[Oct 9 16:30:48] ERROR[2778]: iostream.c:552 ast_iostream_close: SSL_shutdown() failed: error:00000001:lib(0)::reason(1), Internal SSL error
== WebSocket connection from ‘10.203.0.6:42542’ forcefully closed due to fatal write error
– Unregistered SIP ‘9962’
== WebSocket connection from ‘10.203.0.6:33004’ for protocol ‘sip’ accepted using version ‘13’
– Registered SIP ‘9962’ at 10.203.0.6:33004
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
> 0x7f3e7802c380 – Strict RTP learning after remote address set to: 192.168.248.196:49666
– Executing [101001@Matrix:1] Dial(“SIP/9962-00000000”, “SIP/101001,30”) in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
– Called SIP/101001
– SIP/101001-00000001 is ringing
> 0x7f3e800096f0 – Strict RTP learning after remote address set to: 10.176.16.131:58718
– SIP/101001-00000001 requested media update control 26, passing it to SIP/9962-00000000
> 0x7f3e8001f890 – Strict RTP learning after remote address set to: 10.176.16.131:58720
> 0x7f3e800096f0 – Strict RTP switching to RTP target address 10.176.16.131:58718 as source
– SIP/101001-00000001 requested media update control 26, passing it to SIP/9962-00000000
– SIP/101001-00000001 answered SIP/9962-00000000
– Channel SIP/101001-00000001 joined ‘simple_bridge’ basic-bridge
– Channel SIP/9962-00000000 joined ‘simple_bridge’ basic-bridge
> 0x7f3e8001f890 – Strict RTP learning after ICE completion
> 0x7f3e800096f0 – Strict RTP learning after ICE completion
> 0x7f3e8001f890 – Strict RTP learning after remote address set to: 10.176.16.131:58720
> 0x7f3e8001f890 – Strict RTP switching to RTP target address 10.176.16.131:58720 as source
> 0x7f3e78045410 – Strict RTP learning after ICE completion
> 0x7f3e7802c380 – Strict RTP learning after ICE completion
> 0x7f3e78045410 – Strict RTP learning after remote address set to: 10.211.0.208:12264
> 0x7f3e7802c380 – Strict RTP learning after remote address set to: 10.211.0.208:12890
> 0x7f3e78045410 – Strict RTP switching to RTP target address 10.211.0.208:12264 as source
> 0x7f3e800096f0 – Strict RTP learning complete - Locking on source address 10.176.16.131:58718
> 0x7f3e8001f890 – Strict RTP learning complete - Locking on source address 10.176.16.131:58720
> 0x7f3e78045410 – Strict RTP learning complete - Locking on source address 10.211.0.208:12264
== WebSocket connection from ‘10.203.0.6:55230’ closed
== WebSocket connection from ‘10.203.0.6:42256’ closed
== WebSocket connection from ‘10.203.0.6:55212’ closed
== WebSocket connection from ‘10.203.0.6:42230’ closed
== WebSocket connection from ‘10.203.0.6:42234’ closed
== WebSocket connection from ‘10.203.0.6:42238’ closed
== WebSocket connection from ‘10.203.0.6:38688’ for protocol ‘sip’ accepted using version ‘13’
== WebSocket connection from ‘10.203.0.6:38690’ for protocol ‘sip’ accepted using version ‘13’
== WebSocket connection from ‘10.203.0.6:38704’ for protocol ‘sip’ accepted using version ‘13’
== WebSocket connection from ‘10.203.0.6:38720’ for protocol ‘sip’ accepted using version ‘13’
== WebSocket connection from ‘10.203.0.6:55214’ closed
– Registered SIP ‘201002’ at 10.203.0.6:38704
– Registered SIP ‘201002’ at 10.203.0.6:38688
== WebSocket connection from ‘10.203.0.6:55196’ closed
– Registered SIP ‘201102’ at 10.203.0.6:38690
== WebSocket connection from ‘10.203.0.6:38724’ for protocol ‘sip’ accepted using version ‘13’
== WebSocket connection from ‘10.203.0.6:38726’ for protocol ‘sip’ accepted using version ‘13’
== WebSocket connection from ‘10.203.0.6:42222’ closed
– Registered SIP ‘201101’ at 10.203.0.6:38720
== WebSocket connection from ‘10.203.0.6:38728’ for protocol ‘sip’ accepted using version ‘13’
== WebSocket connection from ‘10.203.0.6:38742’ for protocol ‘sip’ accepted using version ‘13’
== WebSocket connection from ‘10.203.0.6:42242’ closed
– Registered SIP ‘251002’ at 10.203.0.6:38724
– Registered SIP ‘251002’ at 10.203.0.6:38726
– Registered SIP ‘251002’ at 10.203.0.6:38728
– Registered SIP ‘251002’ at 10.203.0.6:38742
== WebSocket connection from ‘10.203.0.6:38758’ for protocol ‘sip’ accepted using version ‘13’
== WebSocket connection from ‘10.203.0.6:42254’ closed
– Registered SIP ‘211050’ at 10.203.0.6:38758
== WebSocket connection from ‘10.203.0.6:45304’ for protocol ‘sip’ accepted using version ‘13’
– Registered SIP ‘101001’ at 10.203.0.6:45304
== WebSocket connection from ‘10.203.0.6:45314’ for protocol ‘sip’ accepted using version ‘13’
== WebSocket connection from ‘10.203.0.6:42260’ closed
– Channel SIP/101001-00000001 left ‘simple_bridge’ basic-bridge
– Channel SIP/9962-00000000 left ‘simple_bridge’ basic-bridge
== Spawn extension (Matrix, 101001, 1) exited non-zero on ‘SIP/9962-00000000’
It has to do with SIP, so “sip set debug on”.
You also never mentioned you were using WebRTC. In general, not just for this post, the more information you provide, the more likely someone is to help or provide input.
yes, I’m using using webrtc
asterisk_log1.txt (18.4 KB)
User1 - 101001
User2 - 9962
Reregistration happens once a minite. How to increase the reregistration time?
Issue resolved by setting qualify=yes in sip.conf
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