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

[George Joseph]

On Fri, Aug 20, 2021 at 8:33 AM Antony Stone <
Antony.Stone at asterisk.open.source.it> wrote:

> 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')?
>

On the outgoing pjsip endpoint, set "moh_passthrough = yes".   If you then
put
incoming call on hold, a reinvite with sendonly will be sent to the upstream
server.


>
>
> Thanks,
>
>
> Antony.
>
> --
> "The future is already here.   It's just not evenly distributed yet."
>
>  - William Gibson
>
>                                                    Please reply to the
> list;
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> me.
>
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