We have recently used Digium gateway as alternative to other existing PRI interfaces and it’s been working like a charm. However, we had a client who needed to monitor the PRI connections and send an email alert whenever the PRI go down. Back then we had no issue with PRI PCI cards as we had access to DAHDI and could simply write a daemon to monitor “pri show spans”. With this gateway however, we cannot do it as simple as that as we have no access to the PRI and there is no SNMP tool whatsoever. The way I could think of was to push the syslog to Asterisk server and parse the syslog looking for anything suspicious. Now here is how I worked that out:
Category Archives: IP-PBX
Cisco conference phone 7937 with Asterisk
Recently we had a customer with this cisco conference phone (7937) which doesn’t support SIP protocol. To get it to work with Asterisk we need to either use chan_skinny or chan_sccp. As the latter is a lot more solid is brief you how it work here:
Insert BLOB into MSSQL using PHP
Recently I had a customer who wanted to push all VoiceMail voice files into a table in MSSQL database. After going through a lot of searching and trying I managed to get it running like this:
kickout script for Asterisk Hot desking
I wrote the following script to kick out logged in users to Asterisk. (FreePBX deviceanduser mode)
Grandstream IP Phone auto-provisioning Template
Here is a php file that will read the list of IP phones from a CSV file including MAC address, host IP address, Extension and Secret. Then it will generate a binary for each cfgMAC.temp file.
Download the file here
IP Phone provisioning option 66 and 43
Different IP Phones require different methods of provisioning. Below I describe how to do this on Grandstream phones (tested on GXP285) with firmware: 1.2.5.3
A2Billing configuration on FreePBX
Here is the step by step instruction on how to configure a2billing with FreePBX 2.4:
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Hotline calls (PLAR) with Asterisk
Hotline/Warmline, also known as Private Line Automatic Ringdown (PLAR), is a line used for priority telephone service. When using with a SIP phone you must make sure the phone supports this feature. As of this writing I know that Polycom SIP phones support sending a signal to Asterisk upon the offhook event.
Cisco 7912 with Asterisk
Asterisk phone lock/unlock feature
Below is a simple and intuitive dialplan by which you can add the phone lock feature to your Asterisk system. Take note that the provided dialplan only locks the phone from calling out to PSTN.

