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

[vivek Kumar shah (JIRA)]

vivek Kumar shah created ASTERISK-28330:
-------------------------------------------

             Summary: 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


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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