Hi group I am very interested in the inclusion of Kamailio into Astlinux. Just wondering if anyone has any use cases to describe where they are currently using it.
I have a multisite customer (and more to come Im sure), all with their own local Astlinux system and local trunk. This setup presents real problems transferring calls as the media is not released (please tell me if this should not be the case) when transferring between the systems. This is especially ugly when an incoming call is answered by a central console and then transferred back to an extension at the original site. Yes I realise that you can have a console that is registered to both systems and it rings on different lines but its not really seamless e.g. can’t do a Call Pickup etc from local phones when this line is ringing. And even if you could, you would end up with the same hairpin problem if you transferred it back. My understanding is that Kamailio will solve this problem. Asterisk remains the routing engine however all phones register to Kamailio and all RTP ends up going between the endpoints. If this is the case, then there should never be any hair pinning and only ever a single hop. Obviously you would need to remove all NAT on the network through VPN. Will this work as described? Is it a reasonable use case? The Kamailio config does look pretty scary. Regards Michael Knill
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