[SR-Users] [EXT] Re: Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)
Martin W Woscek
mwoscek at mitre.org
Wed Jun 24 17:21:11 CEST 2020
Hi,
The issue is the P should be sending to the UE not back to the I like the failed flow:
Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->I-CSCF ---> [604 HSS user unknown] to Asterisk16[phone]
The successful call should look like:
Asterisk16-[phone] --->ICSCF--->S-CSCF--->P-CSCF--->Kamailio-[UE]
Does anyone have a example P-cscf scripts including the mt.cfg or other script file that make this work?
The reverse calls work just fine.
Thanks,
_Martin
From: Daniel-Constantin Mierla <miconda at gmail.com>
Sent: Wednesday, June 24, 2020 4:53 AM
To: Kamailio (SER) - Users Mailing List <sr-users at lists.kamailio.org>; Martin W Woscek <mwoscek at mitre.org>
Subject: [EXT] Re: [SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)
Hello,
Kamailio is not generating the 200ok for INVITE (calls), it just sends out what it received from end point, after consuming own Via header (of course, other headers can be changed based on config rules, but Kamailio is not in charge of setting To-tag).
Cheers,
Daniel
On 19.06.20 21:19, Martin W Woscek wrote:
Hello all,
SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful:
[UE]-Kamailio--->INVITE--->Asterisk16-[phone]
[UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone]
But in reverse direction for a call, Kamailio does not return the SIP OK so no to_tag is sent so call fails to ring and complete:
[phone]-Asterisk16 --->INVITE--->Kamailio
Where the UE device never receives the INVITE, Asterisk never gets and 200 OK message with the to_tag from the I-CSCF, the call flow itself gets lost in Kamailio, where the P-CSCF sends final INVITE to I-CSCF and and ultimately a 604 HSS user unknown message is sent back to Asterisk from the I-CSCF.
Basically Im using the default "sample" configs for both the P and the I-CSCF. Our sauce is in the S-CSCF for out going calls that originate by a registered UE.
Any insight or sample Kamailio configuration that Im lacking?
Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible.
Thanks,
_Martin
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