On Wednesday 18 August 2021 at 2021 2:33:42, Antony Stone wrote:
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.
- Throw that SIP Endpoint to the nearest trash bin you could find
Not an option, as already stated.
So you must go route 2, easy peasy
- 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, 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.
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