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

[Eric Wieling]

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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