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

Asterisk Team (JIRA) noreply at issues.asterisk.org
Thu Dec 9 10:20:34 CST 2021


    [ 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.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/].

> 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)



More information about the asterisk-bugs mailing list