[asterisk-users] Between a dumb client and a capable server...

[George Joseph]

On Wed, Aug 18, 2021 at 3:33 AM Antony Stone <
Antony.Stone at asterisk.open.source.it> wrote:

> Hi.
>
> I wonder if anyone has some helpful advice or suggestions for me?
>
>
snip

I had thought that Kamailio might be what I was looking for, but I've asked
> on
> their mailing list and people are telling me that it isn't, and that I
> need
> something like Asterisk to do this.  I'm trying to get some specifics from
> them
> about *how* I would get Asterisk to do this (because I personally can't
> see
> how Asterisk could sit between a SIP client and a SIP server, and generate
> commands to manipulate the RTP stream and send them to the server, which
> is
> what the Kamailio people are saying I should do), but I thought it was
> worth
> asking here just in case what they're telling me is in fact quite easy
> when
> you only know enough about Asterisk.
>
> So, if someone here thinks this is possible using Asterisk, please could
> you
> point me at some documentation showing what commands I would use or the
> basics
> of how I should go about it?
>
> If anyone thinks there is another, perhaps better, way of achieving this,
> then
> I'm quite open to alternative solutions (as I say, I was initially
> thinking
> that Kamailio might be the way forward), so anything that shows me *how*
> such
> a thing might be achieved, with any tool at all, would be very welcome.
>
> Just to summarise: I have a SIP client talking to a SIP server, and I need
> something which can send commands to that server to put calls, which were
> created by the existing client, on hold (that's the simplest scenario).  I
> do
> not want to build a SIP server / PBX myself which can itself perform call
> hold
> & transfer etc (I know how to do that with Asterisk) - I need those
> functions
> to be performed by the existing server.
>
>
Sounds like you're looking for something to do 3rd Party Call Control
(3PCC).
It also sounds like the 'SIP server" isn't Asterisk and you can't change
that either
right?

You could actually use a tiny Asterisk instance to do this. The dumb client
would
call Asterisk and Asterisk would simply send the call to your existing SIP
server.
You could then use AMI or ARI to watch for the call events and tell
Asterisk to
transfer to some other extension on your SIP server or whatever.
The big question is...  what triggers the action to take?
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