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. 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