[asterisk-bugs] [JIRA] (ASTERISK-29799) the call cut between two extensions

[ayoub (JIRA)]

ayoub created ASTERISK-29799:
--------------------------------

             Summary: the call cut between two extensions
                 Key: ASTERISK-29799
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29799
             Project: Asterisk
          Issue Type: Information Request
      Security Level: None
          Components: pjproject/pjsip
    Affects Versions: 18.9.0
            Reporter: ayoub


hi guys; so my issue is, when i call from extension 100 to other 103, the call ring but when i answer, the call cut on first seconde
this is the log

WebSocket connection from '196.41.231.78:47242' for protocol 'sip' accepted using version '13'
– Registered SIP '103' at 196.41.231.78:47242
[Dec 9 17:15:17] NOTICE[7998]: chan_sip.c:28813 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 103
== DTLS ECDH initialized (automatic), faster PFS enabled
== DTLS ECDH initialized (automatic), faster PFS enabled
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x7facdc025490 – Strict RTP learning after remote address set to: 196.41.231.78:35468
– Executing [100 at from-extensions:1] Gosub("SIP/103-00000009", "dial-extension,s,1,(100)") in new stack
– Executing [s at dial-extension:1] NoOp("SIP/103-00000009", "Calling: 100") in new stack
– Executing [s at dial-extension:2] Set("SIP/103-00000009", "JITTERBUFFER(adaptive)=default") in new stack
– Executing [s at dial-extension:3] Dial("SIP/103-00000009", "SIP/100,30") in new stack
== DTLS ECDH initialized (automatic), faster PFS enabled
== DTLS ECDH initialized (automatic), faster PFS enabled
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called SIP/100
– SIP/100-0000000a is ringing
– SIP/100-0000000a redirecting info has changed, passing it to SIP/103-00000009
– SIP/100-0000000a is busy
== Everyone is busy/congested at this time (1:1/0/0)
– Executing [s at dial-extension:4] Hangup("SIP/103-00000009", "") in new stack
== Spawn extension (dial-extension, s, 4) exited non-zero on 'SIP/103-00000009'



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