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