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

[Joshua C. Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-29798?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua C. Colp closed ASTERISK-29798.
-------------------------------------

    Resolution: Not A Bug

> 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'



--
This message was sent by Atlassian JIRA
(v6.2#6252)