Yes, I am pretty sure that if a Polycom unit is set DND and you initiate a multicast page from another Polycom handset on a page or PTT channel that the DND handset is subscribed to (like the emergency channel), then you will hear audio on that handset.
BUT Polycom handsets cannot be configured to just listen to RTP being multicasted to a particular multicast IP like many other IP phones can...the signalling for Polycom multicast paging and PTT functionality is completely proprietary and not SIP-based, and in fact the audio itself is not RTP. It is a proprietary audio packet format that has a header prefixed to it containing signalling information, on every audio packet/frame. Therefore nothing else can initiate a multicast page except another Polycom phone on the same layer 2 broadcast domain...you cannot programmatically have Asterisk/FreePBX do this. Polycom has released an engineering advisory documenting the format, in case anyone in Asterisk land is interested in writing a channel driver that can interoperate with this. I for one think it would be very handy to be able to have Asterisk initiate group paging and push-to-talk on Polycom handsets. The document is here: http://support.polycom.com/global/documents/support/technical/products/voice/Audio_Packet_Format.pdf -- Nathan Anderson First Step Internet, LLC nath...@fsr.com On Wednesday, September 17, 2014 6:03 PM, David Wessell <> wrote: > Tim, > > I THINK but I'm not sure that you can do this with the Polycom multicast > page function. Have you attempted this yet? > > Thanks > david > > On Tue, Sep 16, 2014 at 10:07 PM, Tim Nelson <tnel...@rockbochs.com> > wrote: > > > Greetings- > > As many of your are Polycom "experienced", I was hoping some kind soul > could provide direction on a specific issue. > > On a system running Asterisk 11.11.0 (and latest FreePBX), I'm finding > an instance where, using intercom/paging functionality of FreePBX, I need > to override an end user's 'Do Not Disturb' selection on the handset. By > default, DND simply rejects all inbound SIP INVITEs. However, a > page/intercom needs to be allowed through. > > Any suggestions? I've read reports this is doable using Polycom config > options for call priorities, but I've had no such luck yet. > > Thanks! > > --Tim > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
