1.Open your personal account for the Beeline Business cloud PBX. Go to section Numbers:

2. In section Subscribers select the number that will be used by the Telephone Operator and click on it:

3. In the section Services find and enable the option SIP account. Click the button Tune:

4. In the field Number of SIP lines set the value to 10. Thus, you can transfer 10 simultaneous incoming calls to the Telephone Operator. Click on the link Generate password. Copy the password from the field New Password , he will be needed soon. Click the button Save.

5. If you opened this instruction during Line and script settings, then copy to personal account in form fields PBX connections(SIP) these values:

• Copied in point #4 password in field Password
• SIP User ID in field User
• Domain in field SIP server
• Authorization User ID copy No need
• SIP proxy in field SIP proxy

If you are not studying now Setting up the line and script, then save above specified parameters: You will need them when setting up a new line

Connecting the "Secretary" line

Attention! Before connecting the line, create a new user

There are several ways to connect the line Secretary to your PBX:

1. Secretary, if in voice menu the caller did not dial extension number or dialed the secretary's internal number
2. Incoming call translated into Secretary, only if the caller dialed the secretary’s internal number in the voice menu
3. All incoming calls are immediately sent to Secretary.

We recommend using the first line connection method Secretary. This is exactly what is described in the instructions below. But by analogy with the described method, it is easy to configure other connection options.

1. Go to the section Numbers:

3. Press the button Add a button and select 0 . Mark the added button with a star. Select Transfer via ATS. Enter the extension number that you have configured for the Secretary line. Click the button Save:

Connecting the “Missed Missed Protection” line

Attention! Before connecting the line, create a new user. You cannot use the same user to connect different lines Telephone operators.

The "Missed Protection" line is needed so that a certain group of employees (for example, the sales department) does not miss a single incoming call. In Beeline PBX, groups of employees (departments) are included in the Calling Group. The main scenario for using the "Missed Missed Protection" line is as follows:

• an incoming call arrives to the dialing group
• no one from the group answers the call
• a call from the group is transferred after a certain timeout to the internal number to which the Telephone Operator is connected

1. Go to the section Numbers:

2. In section Services Click on the required ringing group:

3. In the field If the agent does not respond, move on to the next one in... set the value 3 :

4. Enable the option If the waiting time exceeds.... Specify the time in seconds after which the call will be transferred to the Telephone Operator. We recommend setting a value equal to the number of agents in List of agents multiplied by 10. For example, if you have 3 numbers in List of agents, then enter 30 seconds.

5. Enter the extension number that you configured for the Missed Missed Protection line. Click the button Save:

Recently for subscribers home internet Beeline appeared new service— Home digital phone. All users are offered a local city phone number, favorable rates and lack of monthly subscription fee.

You can make and receive calls on your computer using special program(X-Lite), on a VoIP phone, as well as on android device(since version 4.0). In this article we will figure out how to set up the service on smartphones with earlier android versions(including the popular 2.3).

For this we need free application CSipSimple. Download the application to your smartphone and then launch it. A page for creating an account will immediately open. Click on the button " Add account«.

You will be asked to select one of the presets. Since there is no Beeline digital phone in the list, we select extended ( Expert) list of parameters in the section " Settings Wizard«.

• Account name- any
• Account ID- enter , Where user— subscriber’s nickname (your login)
• URI registration- enter sip:sip.beeline.ru
• Realm (authentication domain)- enter * (star)
• Username- enter phone [email protected](country code, city code, telephone number). For example, 78121234567

• Password- enter your password
• Transport- choose UDP
• Default uri scheme- choose sip
• Allow overwriting- uncheck the box
• Proxy URI- enter:
  sip:spb.sip.beeline.ru— for residents of St. Petersburg
  sip:msk.sip.beeline.ru— for residents of Moscow
  sip:krs.sip.beeline.ru— for residents of Krasnoyarsk
  sip:ufa.sip.beeline.ru— for residents of Ufa
  sip:ekt.sip.beeline.ru— for residents of Yekaterinburg

There is no need to change any other settings! When the required fields are filled in, we save the account.

If everything is filled out correctly, the account will be registered online without errors. And you can make and receive calls.

Please note that you cannot be online from multiple devices! For example, if you are logged in via an account on your computer, an error will appear on your smartphone.

It all started with the fact that another communication competition for our hotline 8-800 was won by Beeline. At first, the service was provided simply by redirecting the 8-800 number to one of our external numbers, which we received from another operator. And according to the terms of the contract, we provide the service either via SIP through a dedicated channel, or by redirection to one of our numbers.

And at some point, Beeline decided that “it’s enough for us to provide the service by redirect, let’s connect the subscriber with fiber optics and provide it via SIP.” In general, we successfully installed the optics, installed a media converter (and it’s not rack-mounted at all, and without a shelf - we had to adapt it in a rack). After about a week, we set up the channel, raised a SIP trunk from our asterisk to their server, and made sure the telephony was working. We switched 8-800 to this trunk. Everything worked successfully for 2 (two!) days.

After 2 days, it turned out that asterisk was failing to register on the Beeline SIP server. This is not surprising, because... at this time there was no ping from our SIP server to the Beeline SIP server. Naturally, we immediately created a request for Beeline technical support, and until the situation was clarified, we switched 8-800 to the old scheme.

Interestingly, during the proceedings it turned out that if we ping any IP in the beeline subnet from our SIP server (with the exception of the IP SIP server), then the connection with the SIP beeline is magically restored - the pings go through, asterisk is blocked. This all works for half an hour, then the connection disappears again.

I'll tell you a little about the network configuration. From the Beeline media converter, the patch cord goes to our switch to a port that is configured in access mode and belongs to the 21st vlan. Then this 21st vlan goes to the telephony server (already tagged) and then to the openvz container with asterisk. This container thus has 2 interfaces - one for communication with the world, and the second - purely for access to the Beeline network.

Approximate addressing to make it clear:

• 192.168.1.1 - beeline gateway
• 192.168.1.2 - Beeline SIP server
• 192.168.1.100 - our SIP server
• 255.255.255.0 - mask

So, an application was created to Beeline, they announced that they would conduct checks on their network. In general, we called back and forth for 2 weeks, Beeline tested its network several times, did not find any problems and answered in the spirit of “yes, that’s what you have there.” local problem, repair your equipment." It was almost impossible to prove anything. In general, I managed to go on and off vacation, but the problem remained the same.

I’ll tell you how typical communication with technical support was structured. My colleague Igor took a deep dive into this case while I was on vacation. We call technical support, we hang on the line for 10 to 60 minutes, we get to a polite 1st line employee who doesn’t understand anything about the technical part and can only create/close a request and add a comment. After explaining who we are and naming the application number, as a rule, a transfer to another specialist, this time technical, follows. But the chance that he will pick up the phone is not 100%. Yes, yes, you can hang on the line for an hour and still not get to the specialist.

And now about the qualifications of 2nd line network specialists. To say that she is weak is to say nothing. The "specialists" from Beeline have problems with a basic understanding of how IP networks work. So I found such a specialist, described the problem to him and this interesting trick with ping, after which everything suddenly starts working. This in itself did not interest him, but he wanted to check our settings. Is our IP like this? A mask like this? Is this the SIP server address? Is your gateway registered like this? This is where I say - we don’t have a gateway listed here at all, because... in your network we are only interested in the SIP server, and we do not send any packets to the gateway, because All other networks except Beeline's are connected to our router. This is where Beeline’s “specialist” got seriously excited. How can it be, what are you doing, how is it that you didn’t register our gateway, because of this, of course, everything doesn’t work! I tried to explain to him that for the interaction of hosts on the same network, in the same segment, no fucking gateway is needed, because The arp request left us, a response came from the desired host, the IP packet went to the right place, and still no packets are sent to the gateway. Fuck it, my friend doesn’t understand how the network works in general - he continues to insist that a gateway is simply necessary for such interaction! He doesn’t want to listen to anything, much less use his brain and understand the problem a little.

Well, to hell with you, there will be a gateway for you. We take a laptop with Linux, connect it directly to the media converter, configure the interface as needed, with a gateway. What would you think? Naturally, the problem is reproduced here and there. It works for half an hour - then that’s it, the pings disappear.

After that, I contacted the manager who supervises us, explained to him the situation and the “competence” of their specialists, he promised to do something about it.

Indeed, a few days later their specialist came to us with a VoIP gateway and a hardware phone. We connected, waited half an hour - the connection was lost, as expected. At the same time, the specialist kept in touch with someone “from the other side” who could poke the equipment, look at something, etc. In the process of these tinkering, it was suggested that perhaps in the Beeline network the IP-name that was issued to us is already in use somewhere. I suggested that they look at some of their equipment to see what kind of mac they see before and after.

Well and so it turned out - the IP license issued to us had already been issued to some client before. This is where all the network problems originated. The specialist who came was shocked that such a mess had happened. Immediately, as this guess was confirmed, I was simply the spitting image of Foreman.jpg.

Sandbox

Leonid Yakubovich April 27, 2016 at 2:01 pm

FMC for the poor or how to cross FreePBX and Cloud PBX from Beeline

• Asterisk,
• Development of communication systems

I became interested in integrating mobile network With office PBX based on FreePBX. Our company is not big. I estimated the number of mobile employees - there were 15 of them. Having studied the proposals of telecom operators for the FMC service, I was already thinking of abandoning this idea due to the high cost. But a solution was found that suited me.

Megafon offers FMC for 30 rubles per month from a SIM card + 3540 per month for a digital stream. MTS did not give specific numbers; they promised to provide test access, but they never provided it. Beeline was told that they cannot provide a connection to FMC via SIP and they need to extend fiber to our building, and this is a large expense that will result in a high subscription fee and high connection costs, but they suggested trying to implement it using Cloud PBX. To be fair, it must be said that Megafon also has such a service, but Beeline’s price is more attractive (950 rubles per month for 16 numbers connected to the Cloud PBX. There are other tariffs, but this one suited me best) - I decided to try it.

The initial task was as follows:

• Opportunity free translation conversation from the office to a mobile employee.
• Possibility of free transfer of a conversation from a mobile employee to the office or to another mobile employee.

To organize this, you need to enable a SIP account on one of the connected employee numbers, activate the “Colleagues” tariff without a subscription fee, increase the number of SIP lines on it (on the new tariff line“Everything for Business” you can activate only 2 lines, but on the “Colleagues” tariff the allowed number = 100).

We connect this number as a trunk in FreePBX:

We create rules for incoming and outgoing calls in FreePBX. Nothing complicated here:

In outgoing we indicate the dialing rules for internal numbers (I used 3XX numbering):

We enter the DID in the inbox:

We also add the “Call Transfer” service to the numbers used by mobile employees.

Well, now the disadvantages of this implementation:

• To transfer a conversation from a mobile employee, you need to perform a tricky manipulation:
  

  To use the option, you need to make/receive the first call, put it on hold, then make the second call, then depending on the phone model:

  • On push-button phones(BlackBerry, regular phones) just dial 4 and make a call (key with green handset)
  • On push-button phones (iPhone, Android), you need to go to the main menu, open the “Phone” application again, go to the “Dial” tab and dial 4 and the green button.

  The call is switched to another number, the switch initiator will disconnect from the conversation. If the conversation needs to be transferred to another mobile employee, then the second call will simply be the internal number of the mobile employee, and if someone is in the office, then you need to dial the number used as a trunk (in my case 300) and then dial the internal number from FreePBX

• When transferring a call to a mobile employee, the internal number used as a trunk in FreePBX is determined (in my case, number 300)

Beeline promised to implement the ability to connect trunks to its Cloud PBX in August. I think then it will be possible to organize a more “beautiful solution”.

Tags: asterisk, freepbx, beeline, cloud PBX, fmc, beeline

You and your employees can receive and make calls from desk phones.

Virtual PBX supports work with all models of popular SIP equipment vendors: AudioCodes, Yealink, Grandstream, Cisco/Linksys, D-Link, Panasonic, Escene, Aastra, Fanvil, Eltex and others.

If you have analog phones, you can connect them through a VoIP gateway.

Setting up a SIP phone

Setting up a SIP phone is quite simple and does not require special knowledge. Follow the instructions to install your phone. To complete the setup, you need to enter 3 parameters: server, login and password. Where can I get them?

These parameters are indicated when creating an employee:

In this example:

Login is alexey (usually you only need to enter what comes before @)

Password - if a SIP password is set, use it, otherwise use the normal password

The server is the address of your PBX format company.ats.beeline.kg, in our case msc.ats.beeline.kg

Enter your phone settings using this example. Save, the setup is complete and the phone should register.

Note! If you have already created an employee, but do not remember his password, then you can only change it by editing the employee. If you change a password that is already used in some devices (for example, in a softphone), then when you change the softphone will stop working and will require you to enter a new password. Be careful when changing your password.

Setting up a VoIP gateway

You can also connect analog phones through special VoIP gateways. Configuring VoIP gateway ports is similar to configuring SIP phones.

Ready-made instructions for models:

Nuances of work

All phones support all the functionality of the Virtual PBX. You can transfer and hold calls. Calls are recorded when the “Call Recording” option is enabled.

Please note that all devices connected to the employee can work parallel. For example, you configured redirection to mobile phone and installed a desk phone and softphone. An incoming call will be sent to a mobile phone, a desktop phone, or a softphone. simultaneously. This is very convenient: the employee can answer on any device available at a particular moment.

Fine tuning and possible problems

The STUN, NAT Traversal, and proxy server options must be disabled on the equipment.

The SIP ALG option must be disabled in your router settings.

Codecs: if possible, indicate the priority of use: PCMA (G711a, G711 a-law), PCMU (G711u, G711 u-law), G729, RTP Packet size (packetization time) 20 ms.

Must be allowed TCP ports(80, 443, 8080, 5060, 5061, 30000 - 65535) and UDP (5060, 30000-65535).