I think - best integration, is - Kamailio handles users, Asterisk handles what it must handle - like IVR, registration with VoIP providers etc.... At least we do things this way - Asterisk dont know nothing about users. All calls from Kamailio are accepted, and forwarded either back to Kamailio, if call is to other user, or processed properly. If second user is not online - call drops, and Asterisk understand that second user is not registered. This of course is not best solution, but it works fine for me.
On Thu, May 16, 2013 at 12:44 PM, Barry Flanagan barry@flanagan.ie wrote:
On 16 May 2013 02:18, zhengyw zhengyw@neusoft.com wrote:
thank you very much. but I don't know What is the cause of the problem*（" Unresolvable destination (478/SL)" *）?
1.I was very confused, don't know what to do. can you give more hints ？
Not really. You have not confirmed that you have done what I suggested already, nor provided the log files I suggested.
2.When have you done that *Asterisk and Kamailio Realtime Integration using Asterisk Database*?
I have integrated Kamailio and Asterisk Realtime many times, although I have not followed the (very good) guide that you did, nor have I ever done it with both Asterisk and Kamailio on the same server.
-Barry
zhengyw
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users