[asterisk-bugs] [JIRA] (ASTERISK-29798) the call cut between two extensions

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29798?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=257324#comment-257324 ] 

Asterisk Team commented on ASTERISK-29798:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

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> the call cut between two extensions
> -----------------------------------
>
>                 Key: ASTERISK-29798
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29798
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 18.8.0
>            Reporter: ayoub
>            Severity: Major
>
> 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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