Valcom Paging with an FXO - I am missing something

I have a Valcom paging system set up behind a Sangoma 7/FreePBX 14 instance using a Digium AEX-410 with a single FXO port that is working fine - but I want to Virtualize.

So I set up a GrandStream HT503 with an FXO port to take over for the Digium Card.

The weird thing is the Valcom requires DTMF to select the zone to be paged before the page will go out - this was easily done with the Digium card - in this case the All-Page code is #10 - I have had that working for years with the Digium, but when I try to do it with the Grandstream, it fails.

It’s a School and they use it for the Class Bells with a CRON Job and a Call-File that pages and plays a beep.

Where I am stuck is if I set up the FXO port as an extension, it won’t pass (and dial) the #10 - but if you call the extension and press #10 yourself, it works fine - so Paging is working fine, but the Class-Bell system is failing.

Has anyone else tried anything similar and gotten it to work? I think it should be possible, but I just can’t get the coding correct.

If I set up the FXO as a Trunk, I get an Auth Reject when I try to pass the #10 in the call file - Here is the Call-File that is working for the Digium Card:

Channel: DAHDI/1/#10
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: custom-class-bell
Extension: 10

So my assumption was to just change it to the Extension that I had created:

Channel: SIP/2323/#10
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: custom-class-bell
Extension: 2323

But like I said, it doesn’t like the #10 as the Dial String.

Any ideas?

Most Valcom systems also have a way to do it hard wired without relying on dial tone via an FXO. I generally recommend to clients to use the Snom-PA1 or Cyberdata Paging Adapter for this purpose.

Because of that I don’t use call files for paging. I just use the asterisk command in the cron job. Ext 5197 is the paging extension in this example.

0 10 * * 1-5 /usr/sbin/asterisk -x "channel originate local/[email protected] application playback en/custom/buzzer-1_GJ9hRbVu"

Maybe try using an originate command to replace your call file?

Hmmm…that is a cool way of doing it - and simpler than my Call-File. I really like the PA1’s too - we use them all over the place - just in this case it was already set up for the FXO handoff and they are about 300 miles away from me - but I may plan a trip out there and just switch to a PA1 - it would probably be simpler.

It is a Multi-Zone Valcom though, so the DTMF has to be passed through to select the All-Page. No matter how I connect, I have to pass that sequence to get it to work correctly.

Test using my extension where I am right now…

you can use senddtmf as the application, but I never hear it.

channel originate local/[email protected] application dial #10
channel originate local/[email protected] application dial 10

might need a pause. because when I tell it to say goodbye, I don’t hear entire word. I head dbye

channel originate local/[email protected] application playback goodbye

Ahhh…now we are getting somewhere - I am going to set up a tester here - Thanks!

If you set up the HT503 correctly as a trunk, replacing the DAHDI trunk, everything else should work as before.

IMO this is easier with chan_sip; there are many examples to follow. On the Asterisk side, HT503 is just like SPA3102 or OBi110.

Test by dialing the prefix and page code from an extension (no separate DTMF).

Yeah, that is what I thought too - More experimenting to come.

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