Asterisk Paging to Multiple Polycoms

Hey guys,

So I have an interesting issue it seems. I have an Asterisk 1.6 box with polycom phones talking to it. I can call multiple phones at once, I can page one phone at a time, but paging multiple phones at once does not work.

When I attempt to page multiple phones, all the phones get the alert tone but as soon as the auto answer happens it’s always just one phone that ends up staying on the call. It’s almost like when the auto answer occurs, the first phone to receive it gets the call and the rest get terminated.

It’s almost like I have to force all the phones into a meetme room or something. It would be great if anyone has any ideas, I’m not sure how I will get this working.

Here’s what my dialplan looks like…

;call both phones
exten => 3,1,Dial(SIP/Phone&SIP/Phone2)

;page both phones
exten => *3,1,SIPAddHeader(Alert-Info: Ring Answer)
exten => *3,2,Dial(SIP/Phone&SIP/Phone2)
exten => *3,3,Hangup

Here is what is in my sip.cfg for the actual polycom provisioning…

<alertInfo voIpProt.SIP.alertInfo.1.value="Ring Answer" voIpProt.SIP.alertInfo.1.class="4" />
<RING_ANSWER"Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="1000" se.rt.4.ringer="13" se.rt.4.callWait="6" se.rt.4.mod="1"/>
<RING_ANSWER"Ring Answer" se.rt.4.type="ring-answer" se.rt.4.timeout="2000" se.rt.4.ringer="2" se.rt.4.callWait="6" se.rt.4.mod="1" />

It would be awesome if anyone could give me any ideas or insight on this issue.


The paging is just auto-answer as you noted.

I had the same requirements and used ConfBridge with .Call files to handle. Basically when you call the extension, it copies the call files to asterisk’s area to initiate the calls and then places you into a ConfBridge room. The individual phones are called and when answered are placed into the ConfBridge room.

Make call files for the various phones and store them in a directory.

channel: SIP/1234
callerid: CodeBlue
context: codeblue
extension: s
priority: 1
archive: yes
waittime: 120

add this to your main dial plan for your phones.

; codeblue
exten => 333,1,Answer
exten => 333,n,playback(beep)
exten => 333,n,system(cp /etc/asterisk/codeblue/*.call /var/spool/asterisk/outgoing)
exten => 333,n,ConfBridge(codeblue,M)
exten => 333,n,hangup

add this context as well…when the call files get executed

exten => s,1,Answer
exten => s,n,playback(beep)
exten => s,n,ConfBridge(codeblue,M)
exten => s,n,hangup

Thank you very much for your input. Will these phones hangup after the person paging is done speaking?

Under ConfBridge, I don’t believe so. It is also important to note that ConfBridge is only available under 1.6.2.

You can change ConfBridge and use MeetMe instead. Under MeetMe if you add the A option to the initiator of the call and put and x option on the called people, then it will end the conference when initiator ends the call.

I realize that this topic is almost a year old but I may as well reply anyways.

I just figured out that confbridge does not support paging. At least that is what the Asterisk menuselect pre-install is telling me. app_page depends on app_meetme. If you are using app_confbridge you are sol as far as I can tell.

Look at Page application. I use it with Polycom phones and it works perfectly.

WOW ! This is old but this is exactly what I’m looking for.

I have asterisk 1.8.4 (or is it 1.8.6 not sure) and need to implement paging to polycom soundpoint 501 sets.

Could anyone point me to documentation on how to do this? Remeber, im not a asterisk guru, but i can get around if explained to correctly. Thanks


this post’s very old but this is what i need

in this 333 is it the number to dial to initite the process of calling? i think so but i’m not an expert

; codeblue
exten => 333,1,Answer
exten => 333,n,playback(beep)
exten => 333,n,system(cp /etc/asterisk/codeblue/*.call /var/spool/asterisk/outgoing)
exten => 333,n,ConfBridge(codeblue,M)
exten => 333,n,hangup