[asterisk-bugs] [JIRA] (ASTERISK-28705) chan_sip: Phones loose abiltiy to work, core restarting asterisk fixes issue

[Dan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-28705?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=249625#comment-249625 ] 

Dan  commented on ASTERISK-28705:
---------------------------------

Had it happen again. 

I told it to do a sip set debug on   Whats interesting is this is one of a few users set to tcp.   99% are udp  not tcp.  Not sure if this has something to do with it.  



<--- SIP read from TCP:10.119.102.37:53305 --->
REGISTER sip:172.25.220.18;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 10.119.102.37:62384;branch=z9hG4bK-524287-1---10bb43e4dd97cbd1;rport
Max-Forwards: 70
Contact: <sip:tgarcia at 10.119.102.37:62384;transport=TCP;rinstance=08ed283c294cc901>
To: <sip:tgarcia at 172.25.220.18;transport=TCP>
From: <sip:tgarcia at 172.25.220.18;transport=TCP>;tag=312fe032
Call-ID: gcKoE6J2BIHTntHccZtTSg..
CSeq: 15 REGISTER
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.2.25 rv2.8.112
Allow-Events: presence, kpml, talk
Content-Length: 0

In sip show peers I had  
cia/tgarcia           10.119.102.37                            D  No         No          A  5060     OK (166 ms)                                  Cached RT 

My phone had an X on the display  and it was  showing as Daniel-Siemens/Daniel-Sie 10.119.220.166                           D  No         No          A  5060     OK (75 ms)                                   Cached RT




> chan_sip: Phones loose abiltiy to work,  core restarting asterisk fixes issue
> -----------------------------------------------------------------------------
>
>                 Key: ASTERISK-28705
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-28705
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/General
>    Affects Versions: 16.7.0
>         Environment: I have a Redhat HA cluster running with 2 Rhel 7 nodes with asterisk 16.7.0
>            Reporter: Dan 
>            Assignee: Unassigned
>
> Incoming calls on Chan sip still come into the sytem.    Calls in flight stay running.  
> Endpoints get a "x" on them being ploycom phone model vvx310, 311, soundpoint 430, 450's and  zoipe soft phones. 
> THere are no messages about phones loosing registration. 
> A core restart now on asterisk fixes the issue. 
> I think the problem started around asterisk 16.4.0     



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