I'd suggest opening a ticket at https://issues.asterisk.org, include
full debug logs and minimal test case for reproducing the issue. See
https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
for details about what to post on the new ticket.
Just to ask have you tested chan_pjsip? Not saying this issue shouldn't
be fixed but the chan_sip doxygen and sample sip.conf state that
chan_sip TCP/TLS support is experimental. Given the support status of
chan_sip combined with the declared state of the SIP over TCP feature I
would not use it (but that's just my choice, not trying to tell you what
to do).
One concern is if you are using TCP to communicate with an external
service provider it's possible this could inflate billing. When does
the provider recognize the call as ending if you don't send a BYE?
Probably not a big issue when the remote side hangs up too but you can't
always count on that.
On 01/19/2018 04:03 PM, Chris Jones wrote:
Hello all,
Can I get a little help to understand why I am receiving this error?
From a developer perspective, what Asterisk conditions would cause
this error to trigger?
At exactly 120 seconds after an ongoing call is setup, this pops up in
the console with heavy debugging enabled:
DEBUG[30015]: iostream.c:157 iostream_read: TLS clean shutdown alert
reading data
DEBUG[30015]: chan_sip.c:2905 sip_tcptls_read: SIP TCP/TLS server has
shut down
It doesn’t appear to have a negative effect on the ongoing call;
however, it appears to be keeping Asterisk from sending a BYE message
to the SIP provider at the conclusion of the call.
I have lots and lots of more details if you ask; however, I just need
a nudge in the right direction. v15.1.3, though I have used a version
in 13 and 14 train and had the same problem.
Cheers,
Chris
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