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

Antony Stone Antony.Stone at asterisk.open.source.it
Fri Aug 20 09:33:30 CDT 2021


On Friday 20 August 2021 at 16:14:44, George Joseph wrote:

> On Wed, Aug 18, 2021 at 3:33 AM Antony Stone wrote:
> > Hi.
> > 
> > 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).

Okay, that sounds like useful terminology.

> It also sounds like the 'SIP server" isn't Asterisk and you can't change
> that either right?

It *might* be Asterisk, but if it is, I have no access to it other than the 
SIP credentials a standard telephone would use to register to it.  Then again, 
I might not even *know* what it is - it's just a SIP-based PBX...

> You could actually use a tiny Asterisk instance to do this.

Hm, I'm very dubious about that, based on what I've seen in docs so far...

> The dumb client would call Asterisk and Asterisk would simply send the call
> to your existing SIP server.

Okay, so far, so good, I can get Asterisk to do that.

> 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.

So, let's just take the simplest example - how can I get Asterisk to tell the 
other server to put a call on hold and play that other server's hold music to 
the remote party?

> The big question is...  what triggers the action to take?

That's easy, I have a web interface which is on the same machine as the dumb 
SIP softphone, and that can talk to this "tiny Asterisk server" you speculate 
about, for example by sending in AMI Originate commands to it, which can 
trigger dial plan actions, which can do anything Asterisk is capable of.

My doubts are whether Asterisk as a SIP *client* is capable of this.

So, if I have Asterisk registered as a SIP client to some remote server, how 
can I get Asterisk to tell that remote server to put the call on hold (which a 
standard SIP telephone would normally do by sending a ReINVITE with the SDP 
parameter 'sendonly')?


Thanks,


Antony.

-- 
"The future is already here.   It's just not evenly distributed yet."

 - William Gibson

                                                   Please reply to the list;
                                                         please *don't* CC me.



More information about the asterisk-users mailing list