Hello everyone,
I have an error that I have not yet been able to solve and would like
the help of colleagues to indicate a correct path.
The problem that is occurring is that when the client disconnects the
call kamailio is not sending the BYE forward until arriving at the asterisk.
Both in the test scenario and in the production scenario the problem is
the same and the message I see in the capture is 404 Not here, msg this
coming from kamailio.
Production scenario.
PSTN <----------> Dialer --------->kamailio -----------> asterisk1
-----------> asterisk2
Test scenario.
sipp generated calls ------> kamailio -------> asterisk1
-------> asterisk2
When this occurs, the calls that are disconnected by the client are in a
"zombie" state in asterisk, and end up being terminated by timeout with
the following message in the asterisk CLI:
/[Apr 25 17:49:59] WARNING[2121]: chan_sip.c:4072 retrans_pkt:
Retransmission timeout reached on transmission 22-6073(a)10.110.7.242 for
seqno 1 (Critical Response) -- See
https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions//
//Packet timed out after 31999ms with no response/
In the sipp panel I see in the Retransmission column several
incrementing counters, as per the attachment.
If I take the kamailio from the move and point the sipp to only one of
the asterisk, the retransmissions do not happen and BYE follows normally.
My kamailio.cfg configuration file can be downloaded from this url:
https://drive.google.com/file/d/1bBj4GEZSPrp1iJXSLWQ4Z6g59tEFjnNT/view?usp=…
Thank you very much.
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