In Wireshark, do you see who is closing the connection? Which connection is
closed first: SSL or TCP?
In Asterisk CLI, do you see a SIP message at that time when "sip set debug on"
was enabled? Is it a SIP-REGISTER?
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Thank you, Alexander; great nudges.
Twilio has been less than helpful other than to repeatedly sending me their
configuration guide. The only thing in sip show settings with a timer at 120 is
the default registration, but I changed to 180 and the TLS server still stopped
at 120 seconds. And
Thank you, Corey. The SIP provider is Twilio and they prefer chan_sip. I will
open an issue this evening on 15.2 and see what happens. To answer your
question, if I stop the call post 2-minutes, then Twilio times out just about
300 seconds later. So total call time 7 minutes.
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> It doesn’t appear to have a negative effect on the ongoing call;
The SIP channel closed. RTP continues to flow. Therefore, the call itself stays
up but you cannot send any SIP messages anymore.
In Asterisk, the channel driver chan_sip and its SIP-over-TLS is used by many
folks out there,
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
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:
