18 Aug
2021
18 Aug
'21
7:53 p.m.
On Wednesday 18 August 2021 2021 5:18:51, Antony Stone wrote:
And again arguing ... your are asking for the FULL solution, and we refuse to
give it to you.
David have pointed you to a way of how to solve it, you just DON'T undestand
how to do it, but that's your fault, not our.
A B2BUA MAN! ... how many times you need to hear that?
And I have told you, that and Asterisk its perfectly capable as a client to do so,
the problem is that you NEED TO KOWN how to control Asterisk for that, and I have
told you how. CDF+AMI, there is no prebuild solution for that, so get your hand on
if you are willing to have a solution.
With FreeSwitch it's lot easier to do manipulations on A-Leg or B-Leg of the call
from
the ESL than with Asterisk, but both could do it.
Best regards
We can also
not provide a detailed solution to any problem.
I'm not asking for a detailed solution, however I find what Raúl has referred
to as "hints" to be so vague as not to be helpful, because they just say
"it
can be done with something like X". If I knew how to do it with X then I
would just get on with it. Nobody has pointed at any documentation suggesting
how to do it (David did refer me to FreeSwitch docs, but again, not for
something which actually does what I'm asking for). Regarding
about telling the PBX to put the call on hold, one way how to do
it is by sending a Re-INVITE with a changed SDP from one of the called
parties.
I believe that is precisely what I asked about in my original request:
"Specifically, this thing cannot send REINVITEs in order to put calls on hold,
nor can it handle anything to do with transfers (blind or attended).
"I'm looking for something which does have these SIP capabilities which I
could put in between this application and the SIP server" So, yes, I know that is one way to do it. What I do
not know is how to inject
this into an existing dialogue between client and server. That is why I
thought a SIP proxy would be a suitable solution, because it would naturally
be placed between client and server to start with.
A sip proxy, could not inject things in the dialog whitout braking it, that why
we told you, you need a B2BUA, but it's clear, that you don't get the point.
Maybe you can
get more detailed and useful answers for you at the Asterisk
or Freeswitch user lists.
The detailed answer from the Asterisk list is that Asterisk as a server is
perfectly capable of this, but Asterisk as a client cannot do it, and given
that I need to interface to a remote PBX as a SIP client, that basically means
that Asterisk is no solution to my problem.