Moving telephony to the cloud, and keeping your number

It is more and more trending that companies move their application to a hosted a.k.a. Cloud environment and subscribe to them as services of which most common are email, file sharing, accounting and HR applications and a wide range of others as well. Cloud telephony services, commonly referred to as VaaS (Voice as a Service) […]

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Security considerations when choosing a Cloud Telephony provider

Just like email, many companies are now providing cloud based telephony which includes regular extensions to complex contact centers. A key component to lookout for when choosing your provider is of course security. Anything residing on the internet can possible pose risk to your organization, as you can clearly see these days. Choosing the right

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What the fax! Asterisk, Hylafax+, IAXModem and AvantFax for CentOS 7.x and FreePBX users

This guide attempts to walk you through installing a faxing solution on top of Asterisk that has FreePBX as it’s frontend GUI/dialplan generator. So for it to work, you need to be1) Using CentOS (this is a CentOS 7 guide btw). Tested on CentOS 7.7, 64bit2) Have a working Asterisk (tested on 13,16) either locally

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Limiting calls by DIDs for FreePBX users, with dynamic configurable parameters (Repost)

So, we had this challenge by our customer to do this as they are using PRI and supporting multiple customers. Each customer needs to be limited to n number of channels on PRI. When they were using analog that was simply straightforward, its a physical line, so nothing much you can do about “limiting” it

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Telekom Malaysia (TM) Multi-Line SIP setup with vanilla Asterisk or FreePBX over TEL URI

Happy to say that we’ve successfully set up Asterisk 11 or higher with TM’s Multi-Line SIP which basically uses IMS signalling on Huawei devices used by Telekom Malaysia. We had to modify chan_sip.c and parser  files to support TEL: URI for INVITE messages. Currently, we have enabled it to support incoming INVITES only. TM doesn’t

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