I’m setting up a new FreePBX system with close to 90 extensions. We’re special needs school and need “Page All” capabilities. Is there a way to have the page sound on all phones without adding all extensions to a paging group? My concern with having to explicitly add an extension to the paging group is that we’ll forget this step when adding new extensions.
We’re using Grandstream GXP21XX phones if it makes any difference.
If those phones support Multicast Paging, Paging Pro provides that in conjunction with Asterisk 13.
Apart from that, no. You need to add the phones to a paging group.
The Grandstream GXP2130/2140/2160 will in-fact support multicast paging. However, I’m running Asterisk 12 and from what I understand Asterisk 13 is still classified as experimental so that’s not currently an option for me.
The Paging and Intercom Module docs recommend “having more then 20-25 phones in a single page group”. I see that I can change the “PagingMaxParticipants” value in the advanced settings section to allow me to add all 90 extensions to a page group. However, I don’t understand why the recommended max extension count is 25. Will a page group with 90 extensions work reliably?
Thanks – Steve
Multicast paging is not actually dependent on any commercial modules it just makes it easier, if you set it up correctly it will work and there are plenty of recipes out there for Asterisk in google-land. The problem with +25 paged extensions is ultimately the aging Asterisk SIP stack and “Audio Factory” that need to maintain and synchronize 50 concurrent SIP/SDP connections all running pretty well asynchronously on the network , given enough bandwidth and processing power, it might possibly, perhaps work reliably, maybe , and perhaps PJSIP is better able, BUT for what you need just be economical on bandwidth aned resources and use multicasting ( it is of course only “one way” ), that’s why it is there. . .
Because each of those 25 extensions is an individual call. There’s no real limit as to how many devices you can page, except for at some point your machine and or network will run out of puff and not be able to service that many calls.
The reason why RTP Multicast paging is only available in Asterisk 13 and above was that there were a bunch of issues that we found it in PRIOR to 13. If you’re using Distro, you can move backwards and forwards to different asterisk versions trivially with the command ‘asterisk-version-switch’.
I, personally, think that 13 is good for production use, but there are still some issues with CDRs that give you an unexpected level of verbosity. Have a try!
12 is way more experimental than 13
FWIW, I have successfully used multicast paging since Asterisk 1.8
http://www.voip-info.org/wiki/view/Asterisk+MulticastRTP+channels
Thanks guys. The explanation for 25 max extension recommendation makes sense and is something that would be helpful if included in the Wiki. I just switched to Asterisk 13 using the ‘asterisk-version-switch’ command. It couldn’t have been easier–excellent feature.
Paging Pro has some features we can use so I’ll likely be buying the module but there’s nothing in the Paging Pro docs about multicast paging. Hopefully it’s a no brainer to set up.
Thanks again
Not looking to hijack a thread but this is a great find for me. We have always understood the reasoning for the paging limitation and since all our PBX servers are in the cloud its been an area that we have strictly adhered to paging group limitations.
RTP Multicast Paging is new to me so I am wondering if this works with cloud hosted servers as well?
Any here currently using Polycom VVX series with RTP Multicast paging? I’m trying to read-up and see if they support it.
Sadly no. Routing Multicast traffic is doable, but it needs some engineering work at both ends of the link.
Actually, it need engineering work on EVERY router between every one of your endpoints involved in a “page” (nomally IGMP, but few cloud services nor almost any ISP’s provide that nor an alternative)