The PBX Engineer: 2021

Friday, 6 August 2021

Mitel 5312, 5320, 5330 and 5340 BLF in SIP mode

A few years ago I posted a how to guide to getting Mitel dual mode phones into SIP mode and registering them on your SIP server. While my configuration on that particular job used the programable keys on the phone, there was no call for Busy Lamps.

Last week I was asked to use these phones and get the BLF working. I scoured the Internet and having patched together a couple of resources managed to get them to work.

On these phones you have the <User> tag section to define your user and also the <pkDescription> section to define the keys. Just as your phone wont work for you without both the user and the key setup, so the busy lamps wont work without a modification in both of those sections.

In the <User> section you need to define the users you want to subscribe to BLF updates for. User the BLFGroup="" to do this. Fill the quotes with a comma separated list of extensions to monitor . Then in the <pkDescription> section you define a key as Fea="27"

The below file is the MN_userID.cfg file in my TFTP server. Note how I use the BLFGroup= to setup a subscription for 1050, then in the key section, key 27 (third up from bottom) is the BLF key for 1050

<user_list>

</User>

</user_list>

</pkDescription>

<vlan_id>0</vlan_id>

<host_name>SIP13</host_name>

</Parameter>

Have a play, it worked for me.

John Rogers

www.oneszeros.biz

Friday, 2 April 2021

Killing stuck channels FreePBX Asterisk

I keep finding stuck channels in asterisk / FreePBX conferences, that I then have to manually hangup.

Here is how you find the calls

core show channels

This will bring up a list if channels in use. Identify the one you need to hangup i.e.

PJSIP/itsptrunk-00000002

Then issue command

channel request hangup PJSIP/itsptrunk-00000002

The call will then drop and you have killed the call.

If you need any help with a telephony project you can reach me at www.oneszeros.biz

Asterisk & FreePBX Feature Codes

I have tended to find that documentation for Asterisk and FreePBX based phone systems can be rather spread about. For the generic phone user what codes should they use? Here is a helpful list.

Call Forward

*72 – Call Forward All Activate
*73 – Call Forward All Deactivate
*93 - Call Forward All Prompting Activate
*74 – Call Forward All Prompting Deactivate

*90 – Call Forward Busy Activate
*91 – Call Forward Busy Deactivate
*94 - Call Forward Busy Promping Activate
*92 – Call Forward Busy Prompting Deactivate

*52 – Call Forward No Answer/Unavailable Activate
*53 – Call Forward No Answer/Unavailable Deactivate
*740 – Call Forward Toggle

So how do you call forward to voicemail? This is one point that seems to escape all the guides. Calls can be sent direct to a users voicemail by putting a * in front of their extension number, so you can forward direct to voicemail by dialling *72 *ext. e.g. if my extension is 1002 I can dial *72 *1002 to do the forward.

Turning the prompting on and off turns on and off the voice responses that walk you through it and confirm your actions.

Call Waiting

*70 – Call Waiting – Activate
*71 – Call Waiting – Deactivate

*8 – Asterisk General Call Pickup (When you are part of a call group)
** – Directed Call Pickup (When you are part of a call group)
*2 – (When in a call) Asterisk Attended Transfer
## – (When in a call) Asterisk Blind Transfer
** – (When in a call) Asterisk Disconnect Code
*1 – (When in a call) Asterisk Toggle Call Recording

If you need any help please contact me via www.oneszeros.biz

The PBX Engineer