18 Aug
2021
18 Aug
'21
5:16 p.m.
On Wednesday 18 August 2021 at 2021 2:33:42, Antony Stone wrote:
So you must go route 2, easy peasy
David told you how to do it with FS, 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)
We give you hints about your options to solve your issue. And there is much
more ways of solving the issue.
If you have a
DUMB SIP endpoint, as you have, that lacks the features to
put a call on hold, transfer a call, etc. YOU ONLY HAVE 2 WAYS of solving
that.
1) Throw that SIP Endpoint to the nearest trash bin you could find
Not an option, as already stated. 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. > Getting to this point, this is fully out of scope
of this list, as Kamailio
> it's not a B2BUA and will not (without TONS of work and hours) cover that
> special scenario you have.
Agreed, I realise now that Kamailio is not the
solution to my requirements,
however people here seem to believe they know what *would* be a solution to my
requirements, but so far nobody has pointed me at anything specific which I can
use.
As I told you before, could also be done with Kamailio, but with TONS of hours of work,
that no one will do it for you. Just because there are other tools out there (B2BUAs),
that better fit the needs.
> 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, 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 it really is that simple, please just point me at
one example of how to
actually do it.
Good try.
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.
On the other email I pointed you how you could solve it with Asterisk, using
CDF+AMI PlayDTMF command
Giving you a hint, doesn't mean solving you the problem. Means, you must do
you homework.
Best regards