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