The pitfalls of creating a mini Contact Center on Asterisk
Our organization is engaged in the provision of services. Initially, we were based in the same 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 build up 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 organized battles, who need the phone now! (we did not want to produce a bunch of numbers and how we were mistaken we will show further). Need a PBX! As it should be, the system integrator was allocated for the server, Ubuntu was installed, Asterisk was collected from the source code (you need to optimize it somehow with your hardware!) And screwed the web interface (FreePBX! Who would doubt!). Initially, we used mobile communication from Megafon, and most of the calls were still made in our own region, then the solution quickly came: we put a 3G whistle with an unlimited sim card in the system manager. Pre-unlocked and left the mode only modem. The article Configuring Asterisk to optimize the cost of cellular communication shows how to determine the required modem using MCC / MNC, 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:
The solution, oddly enough, quickly came to mind: we take incoming calls through the SIP channel (Multifon taxis!), And we continue to let outgoing calls through the 3G modem! It turned out a hybrid. (Thanks to vp7 for the article Multiphone from Megaphone - we use an alternative SIP client )
What is most interesting, now we have a multichannel incoming line! We rubbed the handles and waited for calls from customers! But it was not there! Nobody was going to call us on a long distance! Then in each city was bought Simka Megaphone with a city number and bolted to the asterisk. Outgoing communication was still carried out via 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 the / dev / ttyUSB devices * !
There was still waiting for one pitfall. Periodically, numbers from Megaphone, which we received via SIP, began to fall off. No operator of technical support could not clearly explain the reason. At the beginning they referred to the fact that I was using a non-native client, then, when I started saying that it wasn’t working through their application, they created a bunch of tickets. It turned out that nothing!
I selected 2 rules for myself when working with Multifunction:
The number of branch cities by this time reached 10. Each city has its own SIM card with a number. The contact center is about 5 people. 3G modem could not cope with one channel on the outgoing line. I also had to transfer outgoing lines to SIP. And the cost of communication flew up sharply, because we made calls through SIP Megaphone, and his draconian tariffing is 1.5 rubles rounding minutely to any phone in Russia!
The search for a new communications provider took about 1 month and was crowned with success! A large provider was found who provided (I don’t remember exactly) 1.4 rubles per minute to mobile phones in Russia and less than 0.8 rubles to landlines with per-second billing. The most important thing is that they provide the Caller ID number identifier service: we can substitute any number to the caller during an outgoing call (in our case, these are Megafon numbers in city format, not federal, as was the case when calling via SIP from a multifon ).
Now everything is built on the basis of the Elastix distribution. Incoming calls are received via Megaphone, 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 drops, which ensures that we are not afraid of force majeure. There are plans to organize a ring between 2 contact centers (they opened another one) to ensure the distribution of the load on the operators, in case of an increased influx of calls.
Well, if someone is interested in the ups and downs of telephony costs:
At the end of 2011, one-by-one branches were opened in neighboring cities. We already have 2 operators! Now they periodically organized battles, who need the phone now! (we did not want to produce a bunch of numbers and how we were mistaken we will show further). Need a PBX! As it should be, the system integrator was allocated for the server, Ubuntu was installed, Asterisk was collected from the source code (you need to optimize it somehow with your hardware!) And screwed the web interface (FreePBX! Who would doubt!). Initially, we used mobile communication from Megafon, and most of the calls were still made in our own region, then the solution quickly came: we put a 3G whistle with an unlimited sim card in the system manager. Pre-unlocked and left the mode only modem. The article Configuring Asterisk to optimize the cost of cellular communication shows how to determine the required modem using MCC / MNC, 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 outgoing connection is crooked.
- If the operator is talking, then the second client listens to boring beeps, and not our cheerful welcome video!
The solution, oddly enough, quickly came to mind: we take incoming calls through the SIP channel (Multifon taxis!), And we continue to let outgoing calls through the 3G modem! It turned out a hybrid. (Thanks to vp7 for the article Multiphone from Megaphone - we use an alternative SIP client )
What is most interesting, now we have a multichannel incoming line! We rubbed the handles and waited for calls from customers! But it was not there! Nobody was going to call us on a long distance! Then in each city was bought Simka Megaphone with a city number and bolted to the asterisk. Outgoing communication was still carried out via 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 the / dev / ttyUSB devices * !
There was still waiting for one pitfall. Periodically, numbers from Megaphone, which we received via SIP, began to fall off. No operator of technical support could not clearly explain the reason. At the beginning they referred to the fact that I was using a non-native client, then, when I started saying that it wasn’t working through their application, they created a bunch of tickets. It turned out that nothing!
I selected 2 rules for myself when working with Multifunction:
- Balance should always be positive!
- Simka must register with the network at least 1 time per day.
The number of branch cities by this time reached 10. Each city has its own SIM card with a number. The contact center is about 5 people. 3G modem could not cope with one channel on the outgoing line. I also had to transfer outgoing lines to SIP. And the cost of communication flew up sharply, because we made calls through SIP Megaphone, and his draconian tariffing is 1.5 rubles rounding minutely to any phone in Russia!
The search for a new communications provider took about 1 month and was crowned with success! A large provider was found who provided (I don’t remember exactly) 1.4 rubles per minute to mobile phones in Russia and less than 0.8 rubles to landlines with per-second billing. The most important thing is that they provide the Caller ID number identifier service: we can substitute any number to the caller during an outgoing call (in our case, these are Megafon numbers in city format, not federal, as was the case when calling via SIP from a multifon ).
Now everything is built on the basis of the Elastix distribution. Incoming calls are received via Megaphone, 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 drops, which ensures that we are not afraid of force majeure. There are plans to organize a ring between 2 contact centers (they opened another one) to ensure the distribution of the load on the operators, in case of an increased influx of calls.
Well, if someone is interested in the ups and downs of telephony costs:
Source: https://habr.com/ru/post/152297/
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