Shared Line Appearance /Shared hold QUESTIONS

Good morning FreePBX/ Asterisk Gurus.

I greatly appreciate the wealth of knowledge you guys post; although I’m still fairly new to freepbx - the forums have helped quite a bit over the last year.

My question / Situation is with Shared Line Appearance ( Forgive me if I’m using the wrong terminology as I am just getting my feet wet in telecom (involuntarily).

The company I work for had started putting in freepbx boxes at some of the locations when the old IT guy had to take an extended vacation with Buba in the slammer so now I’m finishing his project. The office ladies have complained non stop about loosing their ability to put calls on hold and retrieve the call from another extension.
I’ve worked around this by using the park feature and SNOM 720 handsets.

I’ve set the HOLD key to transfer the call to 70 and setting a group of “programmable keys” as BLF keys to monitor the parking lot extensions. I’ve also configured BLF keys for each extension on each phone so that now the receptionist can see every extension in use. The problem I’m having now is with the users determining which parking slot / button on the phone has the call that they placed on “hold”.

I’ve toyed around with changing the HOLD key from parking lot back HOLD. When I do this and then place a call on hold - the users are able to see that someone else has a call on hold however, they’re unable to retrieve that call… For example - I put extension 101 on hold. All the other extensions can see Ext101 blinking on their individual handsets however if they press Ext101 button from one of their phones; they get a busy signal- their handset tries to dial ext101 instead of retrieving the held call.

I’ve added a print screen of the phone configuration so you guys can see exactly what I’m doing - then maybe my explanation will make a little more sense.
(No custom configurations have been done on the pbx system itself. Clean install of the latest distro running asterisk 1.8 and a couple of registered trunks)
Any help, guidance or nudge in the right direction is greatly appreciated

If you switch to Asterisk 13 or higher you can transfer people right into a parking slot instead of a lot (which will choose the number for you). This way you choose the slot number and then you know where the person is parked. This simulates SLA in 99% of cased.

Basically upgrade FreePBX to version 12+ and start using Asterisk 11+ (preferably 13 for these features). Set Transfer to to “71” on other devices set it higher (maybe 72 or 73) so that they don’t collide.

I’ve just installed the latest ‘stable’ version 6.12.65, FreePBX 12, Linux 6.5 Asterisk 13. I like the look of the new gui but cannot seem to get the registered extensions to show up in the status graph and when I try “sip show peers” from the CLI - it shows Comedia ACL Port Status Description
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]

I’m 100% sure that the phone is registered; I dial 555 on the handset and can see all the typical chanspy info via CLI… Am I extra slow today and doing something wrong???

Are you sure this isn’t a chan PJSIP extension

I chose 'Generic PJSIP device" from the dropdown when creating the extensions.
I originally chose Chan Sip but as I was setting the information- i noticed it said it used 5061 instead of 5060; I never saved the extension I was creating as chan sip and created the generic pjsip device.

There wasn’t a “generic sip device” listed as in the older version of freepbx.

Also - THANK YOU so much for your assistance

Continuing the discussion from Shared Line Appearance /Shared hold QUESTIONS:

I’ve just installed another instance on a virtual machine this time, only difference is I used Asterisk 11 instead of 13. I used the same ChanSip extension. When I run SIP SHOW PEERS; I’m able to see that the extension I created shows up although it’s offline. Looks like the above problem is just asterisk 13…?? Is there a fix for this or should I hold off on deploying this version (especially considering it says that version 13 is experimental during the install). If I hold off on version 13, is there an alternative to get the SLA to work?

If the extensions are using pjsip then pjsip show (TAB-COMPLETE) is probably more appropriate than sip show (TAB-COMPLETE), no?

sip show peers

for sip

pjsip show endpoints

for pjsip

1 Like

Sweet; that makes sense and works!
Forgive me for not knowing; I’ve barely scraped the surface of asterisk 1.8 so version 13 is a totally new animal. :hamster:
Thank you soooo much for the clarification :smiley: