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

[Asterisk Team (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-29799?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Asterisk Team closed ASTERISK-29799.
------------------------------------

    Resolution: Not A Bug

> 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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