There are many occasions where you need to have some sort of communication between your Asterisk system and an external system, be it another telephony platform or external application. The beauty of Asterisk is it’s superb flexibility in terms of collaboration or integration with external phone systems or applications. One common scenario is integration to Avaya system. Here in this post I try to highlight some of the tips and tricks in order to establish a successful communication with Avaya.
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:
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:
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:
I wrote the following script to kick out logged in users to Asterisk. (FreePBX deviceanduser mode)
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
Different IP Phones require different methods of provisioning. Below I describe how to do this on Grandstream phones (tested on GXP285) with firmware: 220.127.116.11
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.