Custom modifications to [ext-intercom]

On my FreePBX/Asterisk installation, FreePBX auto-generates the following in extensions_additional.conf:

include => ext-intercom-custom
exten => _*80.,1,Set(dialnumber=${EXTEN:3})
exten => _*80.,n,dbGet(user-intercom=AMPUSER/${dialnumber}/intercom)
exten => _*80.,n,GotoIf($["${user-intercom}" = “disabled” ]?nointercom)
exten => _*80.,n,Set(__SIPADDHEADER=Call-Info: ;answer-after=0)
exten => _*80.,n,Set(__ALERT_INFO=Ring Answer)
exten => _*80.,n,Set(__SIP_URI_OPTIONS=intercom=true)
----------------------< snip >---------------------------

The problem is that my Aastra phones won’t work with this. To get intercom to work, I have to manually replace:

exten => _*80.,n,Set(__ALERT_INFO=Ring Answer)


exten => _*80.,n,Set(__SIPADDHEADER=Alert-Info: ;info=alert-autoanswer)

I’m getting tired of having to manually patch extensions_additional.conf anytime I make a change in FreePBX. How can I change my FreePBX installation to use the code I need instead of the stock code? Is this stored in a template, module or database somewhere?

I’ve tried putting the correct context in extensions_custom.conf, but that doesn’t seem to work.

Thanks for any suggestions.

You’re welcome!

You are also correct, currently only a single method is supported regardless of which phones you use - as you’ve discovered, this isn’t selectable without modifying code.

I did read somewhere about this being changed in the future…



Eureka! That’s the fix I was looking for. Thanks very, very much!

That raises another question: if different phones handle intercom/paging differently, does this mean that FreePBX only supports a single option system-wide? In other words, does this mean that you can’t successfully use different phones using different methods on the Asterisk server?

Thanks very much again!


You need to modify this file
although these changes will be lost if/when you update your paging module.

As different phones handle intercom/paging differently, there should ideally be a setting indicating which method to use for each SIP extension. I believe there has been talk on this before.