[asterisk-bugs] [JIRA] (ASTERISK-28330) call gets disconnected every 30 sec after answer

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28330?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=246492#comment-246492 ] 

Asterisk Team commented on ASTERISK-28330:
------------------------------------------

We appreciate the difficulties you are facing, however information request type issues would be better served in a different forum.

The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.

If this issue is actually a bug please use the Bug issue type instead.

Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

> call gets disconnected every 30 sec after answer
> ------------------------------------------------
>
>                 Key: ASTERISK-28330
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28330
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>          Components: pjproject/pjsip
>    Affects Versions: 16.2.1
>         Environment: centos 7
>            Reporter: vivek Kumar shah
>            Severity: Minor
>              Labels: pjsip
>
> Hi ,
> I am using asterisk 16.2 and PJsip driver . When the phone is answered , the call get disconnected with bye every 30 secs. Please help.
> I am getting multiple  following message before bye.
> <--- Transmitting SIP response (793 bytes) to UDP:10.9.10.114:35032 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 10.9.10.114:35032;rport=35032;received=10.9.10.114;branch=z9hG4bK-524287-1---76e95a686b04e46c
> Call-ID: 2LJiP5hVJcRW4yhOgD3V-Q..
> From: <sip:phone1 at 10.40.111.48>;tag=9d7c0517
> To: <sip:251 at 10.40.111.48>;tag=9b1afd32-127c-4924-bb3b-9715d7a807b5
> CSeq: 2 INVITE
> Server: Asterisk PBX 16.2.0
> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REFER
> Contact: <sip:10.33.111.48:5060>
> Supported: 100rel, replaces, norefersub
> Content-Type: application/sdp
> Content-Length:   228
> v=0
> o=- 1139927487 2 IN IP4 10.40.111.48
> s=Asterisk
> c=IN IP4 10.40.111.48
> t=0 0
> m=audio 12354 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv



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