asterisk users - Jul 2021 - Call Hold / Transfer via AMI

Hi.

I have the following situation:

An Asterisk 16 server on which I have complete control of the dialplan, and 
which has (a) a SIP trunk to a PSTN gateway provider, and (b) several SIP 
credentials for accounts (extensions) on another Asterisk server.  For this 
example I have SIP username and password for extension 234 on that server.

That "other Asterisk server" is also Asterisk 16, but is a proprietary
PBX
which I cannot even see the dialplan of, let alone modify it.  I do, however, 
have full access to the AMI interface on that PBX, and I can write scripts 
(eg: in Perl) on my own server and connect to the PBX's AMI.

I get an inbound call on my SIP trunk, and I need to dial it on to extension 
456 on the PBX, from extension 234.  So far, so good, I can do all this.  The 
incoming call arrives, I use the SIP credentials for extension 234 to dial 
456, and the call gets answered by 456, who sees Caller ID 234.

Then I need to put 456 on hold, so that they get the hold music which is 
configured on the (proprietary) PBX, and perhaps I then need to dial to a 
different number from extension 234, and maybe ultimately transfer the call.  
Alternatively, I might simply want to resume the call that was put on hold.

Putting a call on hold and then having the option to transfer it to another 
number is easy with a SIP phone, but I need to do it for the call which my 
Asterisk server (acting as a SIP client to the PBX) has initiated.

I can't do the hold function on my own Asterisk server because that would
not
generate the correct hold music for the person on 456.

Also, if I transfer the call (so that 456 is now speaking to some other number 
which I dialled whilst on hold) using my own Asterisk server, I would end up 
with two calls in progress between my server and the PBX, whereas I want the 
PBX to completely handle the transferred call, and mine (which took the 
original incoming call on the SIP trunk) to have nothing further to do with 
it.

I can find AMI commands such as Atxfer, BlindTransfer, and Redirect.  All of 
these are fine if I want to actually transfer a call, but how do I simply tell 
the proprietary PBX to put a call on hold and play the configured music?

I hope this is clear - feel free to ask any questions if not.

Thanks,


Antony.

-- 
I have an excellent memory.
I can't think of a single thing I've forgotten.

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Hi.

Nobody has any ideas?

I have an AMI login to a proprietary PBX (ie: I can't see or modify the dial
plan).  I can subscribe to all events, get channel numbers when calls are set 
up, etc.  I can issue AMI commands, originate calls...

How can I use AMI to put a call on hold, and then either resume it, or perform 
a transfer?

On Thursday 15 July 2021 at 15:01:41, Antony Stone wrote:
> Hi.
> 
> I have the following situation:
> 
> An Asterisk 16 server on which I have complete control of the dialplan, and
> which has (a) a SIP trunk to a PSTN gateway provider, and (b) several SIP
> credentials for accounts (extensions) on another Asterisk server.  For this
> example I have SIP username and password for extension 234 on that server.
> 
> That "other Asterisk server" is also Asterisk 16, but is a
proprietary PBX
> which I cannot even see the dialplan of, let alone modify it.  I do,
> however, have full access to the AMI interface on that PBX, and I can
> write scripts (eg: in Perl) on my own server and connect to the PBX's
AMI.
> 
> I get an inbound call on my SIP trunk, and I need to dial it on to
> extension 456 on the PBX, from extension 234.  So far, so good, I can do
> all this.  The incoming call arrives, I use the SIP credentials for
> extension 234 to dial 456, and the call gets answered by 456, who sees
> Caller ID 234.
> 
> Then I need to put 456 on hold, so that they get the hold music which is
> configured on the (proprietary) PBX, and perhaps I then need to dial to a
> different number from extension 234, and maybe ultimately transfer the
> call. Alternatively, I might simply want to resume the call that was put
> on hold.
> 
> Putting a call on hold and then having the option to transfer it to another
> number is easy with a SIP phone, but I need to do it for the call which my
> Asterisk server (acting as a SIP client to the PBX) has initiated.
> 
> I can't do the hold function on my own Asterisk server because that
would
> not generate the correct hold music for the person on 456.
> 
> Also, if I transfer the call (so that 456 is now speaking to some other
> number which I dialled whilst on hold) using my own Asterisk server, I
> would end up with two calls in progress between my server and the PBX,
> whereas I want the PBX to completely handle the transferred call, and mine
> (which took the original incoming call on the SIP trunk) to have nothing
> further to do with it.
> 
> I can find AMI commands such as Atxfer, BlindTransfer, and Redirect.  All
> of these are fine if I want to actually transfer a call, but how do I
> simply tell the proprietary PBX to put a call on hold and play the
> configured music?
> 
> I hope this is clear - feel free to ask any questions if not.
> 
> Thanks,
> 
> 
> Antony.
-- 
"Can you keep a secret?"
"Well, I shouldn't really tell you this, but... no."


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