<div dir="ltr"><div dir="ltr"><br></div><br><div class="gmail_quote"><div dir="ltr" class="gmail_attr">On Fri, Aug 20, 2021 at 8:33 AM Antony Stone <<a href="mailto:Antony.Stone@asterisk.open.source.it">Antony.Stone@asterisk.open.source.it</a>> wrote:<br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">On Friday 20 August 2021 at 16:14:44, George Joseph wrote:<br>
<br>
> On Wed, Aug 18, 2021 at 3:33 AM Antony Stone wrote:<br>
> > Hi.<br>
> > <br>
> > Just to summarise: I have a SIP client talking to a SIP server, and I<br>
> > need something which can send commands to that server to put calls,<br>
> > which were created by the existing client, on hold (that's the simplest<br>
> > scenario). I do not want to build a SIP server / PBX myself which can<br>
> > itself perform call hold & transfer etc (I know how to do that with<br>
> > Asterisk) - I need those functions to be performed by the existing server.<br>
> <br>
> Sounds like you're looking for something to do 3rd Party Call Control<br>
> (3PCC).<br>
<br>
Okay, that sounds like useful terminology.<br>
<br>
> It also sounds like the 'SIP server" isn't Asterisk and you can't change<br>
> that either right?<br>
<br>
It *might* be Asterisk, but if it is, I have no access to it other than the <br>
SIP credentials a standard telephone would use to register to it. Then again, <br>
I might not even *know* what it is - it's just a SIP-based PBX...<br>
<br>
> You could actually use a tiny Asterisk instance to do this.<br>
<br>
Hm, I'm very dubious about that, based on what I've seen in docs so far...<br>
<br>
> The dumb client would call Asterisk and Asterisk would simply send the call<br>
> to your existing SIP server.<br>
<br>
Okay, so far, so good, I can get Asterisk to do that.<br>
<br>
> You could then use AMI or ARI to watch for the call events and tell<br>
> Asterisk to transfer to some other extension on your SIP server or whatever.<br>
<br>
So, let's just take the simplest example - how can I get Asterisk to tell the <br>
other server to put a call on hold and play that other server's hold music to <br>
the remote party?<br>
<br>
> The big question is... what triggers the action to take?<br>
<br>
That's easy, I have a web interface which is on the same machine as the dumb <br>
SIP softphone, and that can talk to this "tiny Asterisk server" you speculate <br>
about, for example by sending in AMI Originate commands to it, which can <br>
trigger dial plan actions, which can do anything Asterisk is capable of.<br>
<br>
My doubts are whether Asterisk as a SIP *client* is capable of this.<br>
<br>
So, if I have Asterisk registered as a SIP client to some remote server, how <br>
can I get Asterisk to tell that remote server to put the call on hold (which a <br>
standard SIP telephone would normally do by sending a ReINVITE with the SDP <br>
parameter 'sendonly')?<br></blockquote><div><br></div><div>On the outgoing pjsip endpoint, set "moh_passthrough = yes". If you then put</div><div>incoming call on hold, a reinvite with sendonly will be sent to the upstream</div><div>server.</div><div> <br></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left:1px solid rgb(204,204,204);padding-left:1ex">
<br>
<br>
Thanks,<br>
<br>
<br>
Antony.<br>
<br>
-- <br>
"The future is already here. It's just not evenly distributed yet."<br>
<br>
- William Gibson<br>
<br>
Please reply to the list;<br>
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<br>
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