Pitfalls of creating a mini Contact Center at Asterisk

Our organization provides services. Initially, we were based in one city and our contact center consisted of one girl and a mobile phone. As the number of cities increased, there was an urgent need to implement a solution that would quickly increase power. And, of course, the choice fell on IP telephony.

At the end of 2011, one by one branches were opened in neighboring cities. We already have 2 operators! Now they periodically staged battles, who needs the phone now! (we did not want to produce a bunch of numbers and how we were mistaken will show later). Need a PBX! As expected, we allocated the system for the server, installed Ubuntu, collected Asterisk from the sources (you need to optimize it somehow for your hardware!) And screw the web interface (FreePBX! Who would doubt it!) initially we used mobile communication from Megafon, and most of the calls were still made in our native region, the solution quickly came: we insert a 3G whistle with an unlimited SIM card into the system. Pre-unlocked and left the modem only mode. In Configuring Asterisk to Optimize Cellular Coststhe method for determining the desired modem using MCC / MNC is indicated, but we did it easier and created such a trunk:

Configuration => Trunks => Add special trunk => Outgoing settings Special set => datacard / i: 123456789012345 / $ OUTNUM $, where 123456789012345 ( IMEI modem)

As usual, outbound routing is created and our trunk is selected.

What was our surprise that:
  • Multichannel outbound communication is crooked
  • If the operator is talking, then the second client is listening to boring beeps, not our fun welcome video!


The solution, oddly enough, quickly came to mind: we pick up incoming calls through the SIP channel (Multifon drives!), And outgoing calls still continue to be launched through the 3G modem! The result is a hybrid. (Thanks to vp7 for the article Multifon from Megafon - we use an alternative SIP client )
What is most interesting, now we have a multi-channel incoming line! We rubbed our hands and began to wait for calls from customers! But it was not there! Nobody was going to call us on intercity! Then, in each city, a SIM card of Megaphone with a city number was bought and screwed to the asterisk. Outgoing communication was still carried out through a 3G modem. I agree that this is cheaper, but if there is a power failure on the USB port, I constantly had to plug it in and give the asterisk user rights to / dev / ttyUSB * devices again !
There was still waiting for one pitfall. From time to time, numbers from the Megaphone that we received via SIP began to fall off. Not a single tech support operator could clearly explain the reason. At the beginning they referred that I was using a non-native client, then when I already started saying that their application was not working, they created a bunch of tickets. It turned out that in vain!
For myself, I identified 2 rules when working with Multifon:
  • The balance must always be positive!
  • Simka must be registered on the network at least 1 time per day.


The number of branch cities by this time reached 10 pieces. Each city has its own SIM card with a number. The contact center is already about 5 people. 3G modem could not cope with one channel on the outgoing line. I had to also transfer outgoing lines to SIP. And communication costs skyrocketed, as we made calls through the SIP Megaphone, and he has draconian tariffing - 1.5 rubles rounded up per minute to any phone in Russia!

The search for a new communications provider took about 1 month and succeeded! A large provider was found that provided (I do not remember exactly) 1.4 rubles per minute for mobile phones in Russia and less than 0.8 rubles for landline phones with per-second billing. The most important thing is that they provide the Outbound Caller ID service: we can substitute any number with an identifier for an outgoing call (in our case, these are numbers from Megafon in the city format, and not in the federal one, as was the case when calling via SIP from the multi-phone )

Now everything is based on the Elastix distribution. Incoming calls come through Megafon, outgoing calls through the second provider. In the cloud, a backup copy of Elastix with mirrored settings in case the main server or the Internet channel crashes, which ensures that force majeure is not a problem for us. It is planned to organize a ring between 2 contact centers (opened another one) to ensure the distribution of load on operators, in case of an increased influx of calls.

Well, if someone is interested in the ups and downs in telephony costs:

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