Precedence of Emergency Call


(TheWebMachine Networks (Sangoma Software Development Partner)) #21

Set Whisper option on both. Phones on any call in-progress should receive Whisper in ear. Phones not on any calls will get normal Intercom announcement (auto-answer, assuming the phone supports it).


#22

Here my enpoints are Raspberry Pis configured with Linphone as Sip Extenstion.

Hope the above mentioned setting will work with those?


(TheWebMachine Networks (Sangoma Software Development Partner)) #23

Ah, then you may have a problem. I have yet to meet a softphone that supports the Intercom (PBX initiated auto-answer)…LinPhone included. To my knowledge, only proper ATA and endpoint devices support the proper intercom features. Not even high end softphone products like Bria Enterprise support Intercom.

In this case, Whisper will still work for speaking over a call in progress, because that is still mixed server side, but I don’t know how you’ll get around the intercom issue on softphones. Setting the softphone to autoanswer would mean ALL incoming calls would auto-answer, not just the page/intercom calls. I doubt this is what you want.

Softphones are rarely the answer for specialized setups, as they are quite limited compared to something like a proper physical ATA. In short, you probably won’t be able to be “cheap” in your setup by using a freeware softphone client.


(Urbnsr) #24

I’ll have to double check, but I though Mizutech has a softphone that can use Intercom. If I remember correctly, it didn’t work for me due to speaker audio getting into microphone. At least it has auto-answer - Maybe not the same thing.


#25

Also we are using those Pis for announcements not as intercom


(TheWebMachine Networks (Sangoma Software Development Partner)) #26

Your announcements are triggering the auto-answer function on endpoints via the intercom features provided by the Asterisk Page cmd. An announcement/page is effectively just a one-way, multi-party intercom call as far as Asterisk is concerned. This is why you also see the Duplex option in the Page config to allow for two-way talking.


#27

At the start of the emergency page, you might want to redirect any active non-emergency page members using Asterisk ChannelRedirect() dial plan application - place your edits into extensions_custom.conf file.