PAGING Multiple Phones


i am trying to set up paging so that all the phones page out at the same time

exten => 101,1,SIPAddHeader("Alert-Info: ")
exten => 101,n,Dial(SIP/201&SIP/202&SIP/203&SIP/204 [etc…])

however it seems to ring on all phones - then one random phone picks up and only pages on that one speakerphone. What needs to be changed to have all the speakerphones activated at the same time ?

Set the auto answer header first, that depends on your phone’s model. take a look on

That’s what Alert-Info is, although, as you imply, it is non-standard, so some phones may not support it at all, and some may require a different header.

You may find that loss-leader soft phones don’t support it.

Also, because this has security implementations, I hope that there are no phones that will act on this header without having been explicitly configured to do so. Otherwise it could be used to bug a room.

The OP should probably be using the Page application.

thanks for the link

I used the page macro from the example - it works well and all phones not in use get paged

Thank you!

But I still can not make the Paging work.
I can not understand what is wrong. I consulted … ation_Page (but here is few documentation). Paging doesn’t work.

When I dial ‘113’, 1111 and 3333 must ring (CIsco 303):

  '113' =>          1. SIPAddHeader(Call-Info:;answer-after=0)    [pbx_config]
                    2. Page(SIP/1111&SIP/3333,i)                  [pbx_config]
                    3. Hangup()                                   [pbx_config]
-= 4 extensions (7 priorities) in 1 context. =-
  == Using SIP RTP CoS mark 5
    -- Executing [113@myphones:1] SIPAddHeader("SIP/2222-00000009", "Call-Info:;answer-after=0") in new stack
    -- Executing [113@myphones:2] Page("SIP/2222-00000009", "SIP/1111&SIP/3333,i") in new stack
  == Spawn extension (myphones, 113, 2) exited non-zero on 'SIP/2222-00000009'

Now the experiments are on Asterisk 11.3.0 .
I hope it doesn’t matter that I use ‘113’ and not '113’ like in the examples? Or '" symbol is mandatory, including configuration of features.conf? The example from says that it’s not so.

The header changes between Brands, so you may check the proper header for your phone and other issue maybe is that some phones need to enable the autoanswer feature first via the phone’s settings.

Yes, but I used the header for Linksys SPAXXX (I have Cisco 303) , both variants - with the IP address of the PBX and without it:

; test Page application exten => 113,1,SIPAddHeader(Call-Info:<sip:>\;answer-after=0) same => n,Page(SIP/1111&SIP/3333,i) same => n,Hangup

; test Page application exten => 113,1,SIPAddHeader(Call-Info:\;answer-after=0) same => n,Page(SIP/1111&SIP/3333,i) same => n,Hangup

; test Page application exten => 113,1,SIPAddHeader(Call-Info:;answer-after=0) same => n,Page(SIP/1111&SIP/3333,i) same => n,Hangup
Maybe the header for Cisco 303 must be another, though the phone is very alike Linksys SPA 94X. I’ll try to search…

The manual should have that info.

Hey guys - something broke ! :smile:

paging has been working well but recently we noticed a feedback during paging - turns out that the page is now two way - so while paging we can hear ongoing conversatiosn troughout the office -

As per the documentation - this should only work if “d” parameter is specified - but in this cast it is not

here is a new few lines i wrote just to test and it’s consistently failing.

exten => 1105,1,NoOP(Starting the page) same => n,Set(TIMEOUT(absolute)=15) same => n,SIPAddHeader("Alert-Info: <intercom>") same => n,Page(SIP/TestUser,qA(beep)n,15) same => n,Hangup

I am not sure if something changed in asterisk or in Digium Phone Firmware - but something did change … :frowning:

Any ideas - i looked trough firmware change log and didn’t see anything applicable

well i found a fix for this “full duplex”

yum upgrade asterisk

11.8 had this issues - 11.11 and 11.12 do not :smile: