[SR-Users] Kamailio I-CSCF not sending SIP:200 OK messages to Asterisk (to tag)

[Daniel-Constantin Mierla]

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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-- 
Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla

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