Hello,
not sure I understood your problem. The respective tutorial is for using
freeswitch as a media server for kamailio. That means services like
voicemail, conferencing, announcement, etc.
Now, from your email I understand that you want all calls to go to
freeswitch and it will do the routing, right?
Cheers,
Daniel
On 8/12/10 9:24 AM, Abhishek [ABK] Kumar wrote:
Hi SiP-Users,
I got kamailio installed and configured to work using mysql with
FreeSWiTCH using this HowTo wiki
freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms [Asipto - SIP and VoIP
Knowledge Base Site]
<http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms>
:
http://kb.asipto.com/freeswitch:kamailio-3.0.x-freeswitch-1.0.6d-ms
This is also reference on FreeSWITCH portal and was the only decent
tutorial I was able to find on integrating Kamailio with FreeSWITCH.
Followed everything successfully, but I have a slight
confusion/problem in its wonderful working...
From all tests I performed, I perceived following activities:
a.) it handled the extensions added to it very well, and carried over
the call to destination extension
b.) if destination extension is also added to it, call established
without intervention of FreeSWITCH
c.) if extension is not added to it but FreeSWITCH, the call reached
FreeSWITCH but was rejected there
d.) Red5Phone (developed over RED5 Server, @Google-Code) could dial
with success but didn't received any
Now,
/*for point (a)*/ Thank You!
/*for point (b)*/ it should be transferred to FreeSWITCH and that
should be allowed to establish call, otherwise all dial-plans,etc.
goes waste
/*for point (c)*/ may be some settings in Wiki are Obsolete OR New,
could you please point it out
/*for point (d)*/ it may be due to point (b)
--
--
Regards,
Abhishek Kumar
https://sites.google.com/site/abhikumar163/
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