Ian, First off, a belated welcome to the FreeSWITCH community! Apologies for the tardiness of my reply - it's been a bit nuts around here. :) Regarding your calls to multiple Polycom phones, I recommend that you look at the "mad boss" example in the default.xml dialplan file. I've used it with Polycoms without issue. I must confess, though, that I've never tried it on a phone w/ multiple line keys. I'd love to have you try it out and give us some feedback. Thanks! -Michael On Mon, Apr 23, 2012 at 7:12 PM, Ian McMaster <ian.mcmaster@gmail.com>wrote:
As this is my first question, please let me first praise the FreeSWITCH team for a fantastic product and all the efforts of the community. The two books really helped convert all my Asterisk AGI / AMI java code to FreeSWITCH. All IVR play/record/dtmf, etc is working perfectly with FreeSwitch now (and the IVR application is quite large). I came to FreeSwitch after limitations/frustrations with Asterisk. Happy to be aboard, and hope to contribute in time.
Use case: Controlling an IVR caller using the event socket; caller chooses an option which will park the caller, and page out to a group of phones for someone to pick up the parked caller. If the park is answered the script exits, or if the park times out then an alternate action is taken.
The valet_park command is working fine, and I can decipher through the ChannelExecuteComplete event whether the park was retrieved or not.
I am struggling a bit with the paging to multiple polycom phones.
- using something like 'originate {sip_auto_answer=true}user/1001,user/1002,... &playback(announce_page.PCMU) has limited success - my understanding is that the first phone to answer 'wins' and the other phones do not receive the playback. - I have better success with conference (a) Invite phones into the conference: conference page_group dial {sip_auto_answer=true}user/1001 ... etc (b) Play file: conference page_group play announce_page.PCMU (although I am having a codec problem here!) (c) End the conference: conference page_group hup all While the conference by nature has certain pre-stocked files (i.e. you are the first caller, and music on hold, etc), I believe I can silence them all with a custom conference.
My only show stopper is that if a page goes out to a phone with two line buttons, and is already in use (on one line), then things go bad. The incoming conference request is knocking the existing call on hold, the page is playing out, and the user must resume the call with a soft button.
Questions: 1) Is there a better approach to paging multiple phones that I have yet to try? 2) Is there a way to avoid paging a phone that already has a caller on it? 3) Is there conference variable to set the codec to allow the PCMU file to play cleanly (it does play cleanly using originate without setting any explicit codec).
Thank you in advance, Ian.