[SR-Users] Is this idea even feasible?

[Antony Stone]

On Wednesday 18 August 2021 at 16:16:29, Raúl Alexis Betancor Santana wrote:

> On Wednesday 18 August 2021 at 2021 2:33:42, Antony Stone wrote:
> 
> > > 2) Put a B2BUA in front of that SIP Endpoint, and throught API, DTMF,
> > > RPC or witchever method that B2BUA gives you, you will have to emulate
> > > what your SIP Endpoint doesn't support
> > 
> > Precisely what I am asking how to do, thank you.
> 
> David told you how to do it with FS,

No, he told me how to get FS to put a call *which it is processing* on hold 
*inside FreeSwitch* - that is *not* what I need to do - I need to send a 
command to the *existing PBX* which is handling the call so that *it* puts it 
on hold, just the same as it would if I had standard SIP phone and could press 
the "hold" button.

> I told you, how to do it with Asterisk, (if you wait a couple of hours,
> message will be approved and posted on the list)

If you are referring to:

"The thing you need it's a B2BUA, and Asterisk, FreeSwitch, YATE, SEMS, either 
a simple python script using pjsua2 or any of the dozens SIP frameworks out 
there, will allow you to acomplish your goals

"If you go the Asterisk road, any of the AMI/ARI documentation will explain to 
you how to handle ongoing calls."

then this neither tells me how to do it, nor is it even what I need to do (see 
above about FreeSwitch).

> We give you hints about your options to solve your issue. And there is much
> more ways of solving the issue.

Please show me just one of them.

> > > You have been given with the hints about how to solve your problem,
> > 
> > Hints are all very well, but telling me "put a B2BUA in front of that SIP
> > Endpoint, and use API, DTMF, RPC or witchever method that B2BUA gives
> > you" doesn't exactly help when I've made it perfectly clear that I don't
> > know how to solve the problem.
> 
> So, if your are unable to follow a hint, read the docs,

Which ones, please?  You seem to know, why not just tell me?

> try the things on your own and ask the right questions on the right place,
> better you hire someone that could solve it for you.

If I find someone who says they can do it, that's definitely an option.

> > If it really is that simple, please just point me at one example of how
> > to actually do it.
> 
> Good try.

Try what?  I'm just saying that if you know how it can be done, please show me 
an example.

> It's really simple, for someone that knows how things works. With a minimal
> of dialplan programing knowleade of Asterisk, FreeSwitch, YATE, SEMS, etc.
> and how to interact with that B2BUA from outside the SIP channel.

You really don't get what it is I need to do.

I do NOT need to build myself a B2BUA using any of the above tools and get 
*that machine* to put calls on hold, transfer them, conference, etc.  I can, 
and have, done that perfectly well using Asterisk.  Despite your opinion, I do 
in fact know both SIP and Asterisk pretty well.

What I need is something which can tell *the existing server* to put the call 
on hold, resume it, or transfer it, in just the same way that a competent SIP 
telephone can tell the server to do that.

> On the other email I pointed you how you could solve it with Asterisk,
> using CDF+AMI PlayDTMF command

How does playing a DTMF tone down a channel tell some other server to put a 
call on hold?  That makes no sense at all.

> Giving you a hint, doesn't mean solving you the problem. Means, you must do
> you homework.

I'm looking for the textbook.  (Oh, and if your response is "buy the Kamailio 
book", I have.)


I have tried to be polite in my responses to you; I would appreciate if you 
did the same.


Regards,


Antony.

-- 
Ramdisk is not an installation procedure.

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