[asterisk-bugs] [JIRA] (ASTERISK-28596) 34/5000 Problem with the sip in asterisk 16

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28596?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=248503#comment-248503 ] 

Asterisk Team commented on ASTERISK-28596:
------------------------------------------

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.

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.

> 34/5000 Problem with the sip in asterisk 16
> -------------------------------------------
>
>                 Key: ASTERISK-28596
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28596
>             Project: Asterisk
>          Issue Type: New Feature
>      Security Level: None
>          Components: . I did not set the category correctly.
>    Affects Versions: 16.6.0
>            Reporter: David
>
> I have a problem, I am trying to use the sip in asterisk 16, and when I call it hangs up after 6 seconds.
> I enclose what I get when I analyze the results:
> [2019-10-23 11:08:49] WARNING[8752]: chan_sip.c:4119 retrans_pkt: Retransmission timeout reached on transmission nwzyJiDzSxozgsxxJF9tVw.. for seqno 2 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> Packet timed out after 6400ms with no response
> [2019-10-23 11:08:49] WARNING[8752]: chan_sip.c:4143 retrans_pkt: Hanging up call nwzyJiDzSxozgsxxJF9tVw.. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
>     -- Channel SIP/116-david-00000035 left 'native_rtp' basic-bridge <498ba006-0972-4736-b07d-f4545b05fc15>
>   == Spawn extension (davidcontext, 117, 1) exited non-zero on 'SIP/116-david-00000035'
>     -- Channel SIP/117-david-00000036 left 'native_rtp' basic-bridge <498ba006-0972-4736-b07d-f4545b05fc15>
>   == Extension Changed 116[hints-david] new state Idle for Notify User 115-david
>   == Extension Changed 117[hints-david] new state Idle for Notify User 115-david
>   == Extension Changed 117[hints-david] new state Idle for Notify User 115-david
> [2019-10-23 11:08:50] WARNING[29528]: res_pjsip_pubsub.c:3305 pubsub_on_rx_publish_request: No registered publish handler for event presence
> [2019-10-23 11:08:50] WARNING[29682]: res_pjsip_pubsub.c:776 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo



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