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." Please reply to the list; please *don't* CC me. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users