VoIP hardware is developing fast - this is where you ask all those “how do I make my SIP Telephone, Adapter or Asterisk box work with my voip provider?” questions.
User avatar
By p.vladi
#4414
I hope someone can help, been banging my head on this one, and FreePBX forums cannot help.

Scenario:
Bob's extension is 101
Pete's extension is 102
Bob and Pete are on same call group, so can pick up each other's calls by dialling **101 and **102

Now comes the fun part
BLF is set up on Panasonic phones (KX-HDV230) as user extension, BLF working as intended, it is red when that extension is busy, and flashes when that extension is being called, and calls that extension when pushed.

however, Pete's extension rings, and Bob wants to pick that call up, but instead of pickup, a parallel call is placed from Bob to Pete, so now Pete has two calls.

One can set up the button as one touch pickup, and that works as intended, however, it does not indicate if that extension is busy or being called...

Many thanks for any advise!
User avatar
By WelshPaul
#4415
To achieve what your asking on a Yealink phone "Dialog-Info Call Pickup" (It enables or disables the IP phone to pick up a call according to the SIP header of dialog-info for account X.) must be DISABLED and both the "Directed Call Pickup Code" and "Group Call Pickup Code" parameter fields be configured to use ** and *8.

Unfortunately I have limited experience with the Panasonic range of IP phones and from what little past experience I have, I can tell you that the Panasonic phones I have used in the past had very limited configuration options, so I doubt there is any way to configure the phones like you can with a Yealink. I will look into this more tomorrow.
User avatar
By WelshPaul
#4418
Yea, I found that in a search myself last night. I was going to give it a try this morning on a test box before recommending you try it on a live server. :)

Edit: If you do try and implement this, edit the custom file equivalents and not the files named in the guide. E.g. sip_custom.conf and extensions_custom.conf.

What version of Asterisk are you running? The patches listed here https://issues.asterisk.org/jira/browse/ASTERISK-19516 may need to be applied too. Be careful though, the patches are for an old version of asterisk so no doubt won't work with later releases...

Beginning in Asterisk 13, Asterisk supports RFC 4662 resource list subscriptions in its PJSIP-based SIP implementation. https://wiki.asterisk.org/wiki/pages/vi ... d=30278158
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