On Monday 19 July 2021 at 11:59:34, Naveen Albert wrote: > I'm not sure I fully understand the entire question. > If you have full AMI access, why not send the ShowDialPlan AMI action to > see the dial plan?
1. How would seeing the dial plan help me to put a call on hold? 2. The result is pretty much useless to me because it's all done via AGI. Antony. > On 7/19/2021 4:02 AM, Antony Stone wrote: > > 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. -- “If code doesn’t receive constant love, it turns to shit.” - Brad Fitzpatrick, Google engineer 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