Based on a customer request, we needed to have some virtual extensions (Voice Mail enabled) but the problem was that they wanted the incoming calls to extensions to ring 4 times before falling back to voice mail. Since by default FreePBX will get the extension status as UNAVAILABLE and I couldn’t get virtual extension running with this version of freepbx, i did this trick:
Today we had a customer who wanted to use the phone unlock feature for their virtual extension which had no physical phones. Normally phone lock feature works directly from the extension physical phone. So I had to do some customisation in the asterisk dialplan. Here is the code:
Configuring SPA3102 (or SPA3000) is not very straightforward and requires alot of configurations on both the Gateway and Asterisk PBX. Follow this instruction step by step.
Note that CLID and MTMF in IVR is successfully checked with this gateway.
It appears that the mail server (SendMail) is not so straightforward in terms of configuration.
so in systems that have SendMail by default, you might have to remove sendmail using the following command:
apt-get --purge remove sendmail
Trunk to a SIP switch using FreePBX
Below is the instruction:
Asterisk Cisco Call Manager (CCM) Integration
Why integrate Cisco CallManager and Asterisk?
- Features: Asterisk provides features that CallManager by itself does not.
- Migration: Allow a gradual migration from a closed source PBX to open source PBX.
This is the instruction on how to install a2billing with FreePBX:
basically, 1 extension in freepbx is 1 card in a2billing. so 1 extension in freepbx represented as a card in a2billing a card holds value of call-plan a call plan contains one or several ratecards a ratecard contains one or several rates a rate is where you configure rate per destination, by time, whether or not its progressive in a rate also configured which trunks will be used so: card –> call plan –> ratecard –> rate –> trunk to configure it, usually from bottom to top, that is from providers –> trunks –> rates –> ratecards –> callplans –> assign call plan to a card
SIP trunk to ATCOM PBX from FreePBX
smaRtPBX IP: 192.168.1.200 SIP extensions: 100, 101, 102, 103, 104, 105
ATCOM IP: 192.168.1.100 SIP extensions: 200, 201, 202, 203, 204, 205
How to SIP trunk from one FreePBX to another?
Here is the instruction on how to create a peer to peer trunk between 2 IP-PBX using FreePBX (Elastix, Trixbox, etc.)