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

[George Joseph]

On Fri, Aug 20, 2021 at 2:33 PM Eric Wieling <ewieling at nyigc.com> wrote:

>
>
> On 8/20/21 4:24 PM, Antony Stone wrote:
> > On Friday 20 August 2021 at 19:06:09, George Joseph wrote:
> >
> >> On Fri, Aug 20, 2021 at 8:33 AM Antony Stone wrote:
> >>>
> >>> 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.
> >
> > So... how do I put the incoming call on hold, when the dumb client I'm
> > starting from cannot do that bit?
> >
> > I already know (from this list) that Asterisk as a SIP client cannot do
> ore
> > than (a) place a call, (b) answer a call, and (c) hang up a call.
> >
> > So, I'm still intrigued as to how you think this might be possible.
> >
> > If it *is* possible, I'd be really interested, but all my researches so
> far
> > suggest that Asterisk, acting in the middle like this, just cannot add
> the
> > necessary "put call on hold" which the original client cannot do.
> >
>
> With Asterisk, keep Asterisk in the media path with direct_media=yes and
> use DTMF to hold, transfer, and other features using features.conf.
> Asterisk has to stay in the media path when NAT is involved anyway.
>

You need to set direct_media=no to keep Asterisk in the media path.


>
> I doubt anything except Asterisk or other B2BUA software can do what you
> want.
>
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