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<p class="MsoNormal">Hello all,<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">SIP/UE (boghe or imsdroid) client registered to Kamailio makes call to an Asterisk registered SIP phone is successful:<o:p></o:p></p>
<p class="MsoNormal">[UE]-Kamailio--->INVITE--->Asterisk16-[phone]<o:p></o:p></p>
<p class="MsoNormal">[UE]-Kamailio--->200 OK with to_tag--->Asterisk16-[phone]<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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:<o:p></o:p></p>
<p class="MsoNormal">[phone]-Asterisk16 --->INVITE--->Kamailio<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">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.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Any insight or sample Kamailio configuration that Im lacking?
<o:p></o:p></p>
<p class="MsoNormal">Has anyone done this and could share the asterisk and Kamailio script snippets that make it possible.<o:p></o:p></p>
<p class="MsoNormal"><o:p> </o:p></p>
<p class="MsoNormal">Thanks,<o:p></o:p></p>
<p class="MsoNormal">_Martin<o:p></o:p></p>
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