[SR-Users] Is this idea even feasible?
hw at skalatan.de
Wed Aug 18 19:01:50 CEST 2021
I just ACKed the stuck post from Raul in the list manager, should be end up on the list in some minutes.
Henning Westerholt - https://skalatan.de/blog/
Kamailio services - https://gilawa.com
From: sr-users <sr-users-bounces at lists.kamailio.org> On Behalf Of Raúl Alexis Betancor Santana
Sent: Wednesday, August 18, 2021 6:44 PM
To: Kamailio <sr-users at lists.kamailio.org>
Subject: Re: [SR-Users] Is this idea even feasible?
On Wednesday August 18, 2021 4:01:10, Antony Stone
> > 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.
Man, really, you don't know how things works so STOP saying that something is not what you need, only because you don't have the knowleage.
David, told you EXACTLY, what you need, send the API call to hold the B-Leg on the FreeSwitch.
YDSC ---- > FS ---- > PBX
If you send the HOLD api commando to FS, and told it to doit on the B-Leg of and outgoing call, FS will put the leg PBX on hold, EXACTLY as if you have a "hold"
button on your YDSC (Your Dumb SIP Client).
But as is more than clear, that you don't have a f**k idea how things works on the SIP world, you keep saying "that's not what I was asking for".
> > 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:
No, I was refering to the reply that still is not on the list, where I pointed you to and way to solve your issue using Asterisk, pointing you to the DOCS where it's explaixed how that AMI commands works and what you need to take into account on the dialplan to get it working.
> then this neither tells me how to do it, nor is it even what I need to
> do (see above about FreeSwitch).
That's point you to the docs WHERE it explained how it works. But you whant me to give you the full solution, that will not happen.
> > 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.
Again ... RTFM, I've told you more than tree times, that the solution is on the docs.
> Which ones, please? You seem to know, why not just tell me?
Because I do not work for free, and your attitude on the thread, not ever reading the docs we point you, don't give me any incentive to do it so.
> > 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.
I could count at least 5 people on this list, that have told you, that could be done, but will be easier do it using a B2BUA.
> > > 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.
Try to make me doing your job, and that will not happen, I've told you why.
I've point you to the docs where it's explained how to put a call on hold, in Asterisk throught the AMI and enabling some features on the dialplan manager.
I'm not going to give you a full step-by-step guide on how to do it.
> > 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 got it perfectly, It's just I don't want to give it to you, as simple as that.
> 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.
Your answers show to me, and the rest of the list, that what you call "knowing Asterisk"
it's probably nothing deeper than runing an Issabel or FreePBX, if you would REALLY know Asterisk and the SIP protocol, you will not insist that what we have told you is not what you need.
> 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.
Again an again ... YOU COULD DO THAT WITH A CUSTOMIZED B2BUA. Get it or leave it
> > 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.
So you say you know Asterisk and you don't know that on Asterisk and on any other SIP pbx, you could control the call flow using DTMF? ... Really?
Please read the [featuremap] section. You could do that control on the A-Leg or on the B-Leg, if your upstream PBX doesn't support DTMF call-control, THATS where your customized B2BUA take into place. If your still doesn't unsdestand that. Just give up on this.
> I'm looking for the textbook. (Oh, and if your response is "buy the
> Kamailio book", I have.)
And if you have it, why are you asking if Kamailio could do a B2BUA Role feature, like putting a call on hold? ... just because you don't undestand a word of what you have read from that book.
> I have tried to be polite in my responses to you; I would appreciate
> if you did the same.
I'm been very polite on my replies, better you don't get see my BOFH side ;-)
There is thousands ways of skinning a cat, and you insist on the only one that don't work ... ;-)
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