Hi * gurus,
Please help I want my internet users to terminate calls via cisco gateway, I don’t want asterisk holding on to the call.
Is this possible?
What exactly are you trying to do?
Hi Malcolm,
I need to be able to receive large number of calls concurrently, possibly 1k++.
[quote=“alkaline”]Hi Malcolm,
I need to be able to receive large number of calls concurrently, possibly 1k++.[/quote]
Who is Malcolm? Also, this seems more like a question for your Cisco Gateway, and how it may send traffic to another SIP agent to play hold music, and take it back when it is ready.
not really, you see, if I get sip connections in asterisk and the audio stream keeps on coming, i may not be able to sustain good QoS when it hits more connections.
Well, if you want to provide MoH then of course there is an audio stream. What you are trying to do and what your issues are, are not clear. If you want to use a Cisco GW/Asterisk in the same way as proposed as a SER/Asterisk coupling, you may look here for some recommendations on how it works:
voip-info.org/wiki-Asterisk+at+large
But, once again, it would seem your questions are more around the Cisco GW and not Asterisk directly.
maybe i need to repharse my question sorry 
A) ua ----> ser ----> asterisk ----> gateway ----> ua
B) ua ----> ser ---------------------> gateway ----> ua
in the above scenario (A) was initiated for authentication and other specific validations, then after, if valid, ser connects to gateway bypassing asterisk (B). this way asterisk is freed-up, MoH not required.
is this possible?
With SER, yes, have a look at the link I posted above. I would recommend it for these types of apps, as it is great at scaling and deferring a lot of traffic that does not need to go through Asterisk for enhanced services.
that clears some cobwebs, thanks! i’ll start plugging these and see how it works.
some specific config would be as helpful i.e ser setup record route, ser rtpptoxy config. thanks for any help.
[quote=“alkaline”]:idea: that clears some cobwebs, thanks! i’ll start plugging these and see how it works.
some specific config would be as helpful i.e ser setup record route, ser rtpptoxy config. thanks for any help.[/quote]
I am not intimately familiar with SER so not able to assist directly. There may be others around here that know it well, but you could also go here:
voip-info.org/wiki/index.php … ess+Router
SER has its own Wiki, just like Asterisk.