1.Open your personal account of the Beeline Business cloud PBX. Go to section Rooms:
2. Section Subscribers select the number to be used by the Operator and click on it:
3. 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 Operator. Click 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 manual during Line and scenario settings then copy to personal account into form fields PBX connections(SIP) these values:
- Copied in paragraph #4 password in field Password
- SIP UserID in field User
- Domain in field SIP server
- Authorization User ID copy no need
- SIP proxy in field SIP proxy
If you are not working now Line and script setup then save above specified parameters: you will need them when setting up a new line Connection of the "Secretary" line
Attention! Create a new user before connecting the line
There are several ways to connect the line Secretary to your PBX:
- Secretary, if in voice menu the caller did not dial the extension number or dialed the secretary's extension number
- Incoming call is transferred to Secretary, only if the caller dialed the secretary's internal number in the voice menu
- All incoming calls go directly to Secretary.
1. Go to the section Rooms:
3. Press the button Add button and choose 0 . Mark the added button with an asterisk. Select PBX transfer. Enter the extension number you configured for the Secretary line. Click the button Save:
Connecting the "Protection from missed" line
Attention! Before connecting the line, create a new user. You cannot use the same user to connect different lines Telephone operators.
The "Protection from missed" 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 "Protection from Missed" line is as follows:
- incoming call goes to call group
- no one from the group answers the call
- the call from the group after a certain timeout is transferred to the extension number to which the Operator is connected
1. Go to the section Rooms:
2. Section Services click on the desired call group:
3. In the field If the agent does not respond, go to the next one in... set the value 3 :
4. Enable the option If the waiting time has exceeded.... Specify the time in seconds after which the call will be transferred to the Operator. We recommend setting a value equal to the number of agents in Agent List multiplied by 10. For example, if you have 3 numbers in Agent List, enter 30 seconds.
5. Set the extension number you configured for the Missed Protection line. Click the button Save:
Sandbox
iron Man April 27, 2016 at 02:01 pmFMC for the poor or how to cross FreePBX and Cloud PBX from Beeline
- Asterisk ,
- Development of communication systems
I am interested in integrating mobile network with PBX based on FreePBX. Our company is not big. I estimated the number of mobile employees - they turned out to be 15 people. Having studied the offers 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 figures, they promised to provide test access, but they never did. Beeline was told that they cannot provide a connection to the FMC via SIP and they need to extend the optics to our building, and this is a big cost that will result in a high monthly fee and high cost connection, but offered to try to implement using the "Cloud PBX". In fairness, it must be said that Megafon also has such a service, but Beeline has a more attractive price (950 rubles per month for 16 numbers connected to the Cloud PBX. There are other tariffs, but it suited me the most) - I decided to try.
The initial task was the following:
- Possibility free translation conversation from the office mobile employee.
- Possibility of free transfer of a conversation from a mobile employee to the office or to another mobile employee.
We connect this number as a trunk in FreePBX:
Creating rules for incoming and outgoing calls in FreePBX. There is nothing complicated here:
In outgoing, specify the dialing rules for internal numbers (I used 3XX numbering):
In the incoming prescribe DID:
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 do a tricky manipulation:
To use the option, you need to make/accept the first call, put it on hold, then make the second call, then depending on the phone model:
- On the push-button phones(blackberry, regular phones) just dial 4 and send a call (a key with a green tube)
- On keyless phones (iPhone, Android), you need to exit to the main menu, open the "Phone" application again, go to the "Dial" tab and dial 4 and the green button.
- When transferring a call to a mobile employee, the extension number used as a trunk in FreePBX is determined (in my case, number 300)
Tags: asterisk, freepbx, beeline, cloud PBX, fmc, beeline
As the English say: If you can't beat em, join em". Which translates into Russian as: "You can't fight, then lead." It was this English proverb that Beeline and Megafon probably meant when they stopped fighting Skype, and offered their VoIP telephony solutions based on SIP. Today I will tell about them.
As you can see, everything is very simple. It is also easy to get access to the "Multifon" service. You just need to activate this service with one of the 3 possible ways(Read the details here: "How to activate the MultiFon service?"). Yes, I completely forgot! To do this, you need to be a Megafon subscriber. Sorry, I completely forgot to mention that! Money for calls made will be debited from your phone account. Exist specialized programs for PC (for Windows and linux), as well as apps for Android and iOS. In addition, you can send SMS from desktop programs. For smartphones, this service is not available from the Multiphone application. But there is a way out - an application for UMS services, which allows you to receive and send SMS from any device on Android and iOS. Read the settings for any SIP program or application here: "How to set up the MultiFon profile in alternative software clients or hardware SIP phones?". As is customary in SIP, calls (including video) between Multifon users are absolutely free.
For me, it turned out to be a very nice service for Megafon. It is strange that others Russian Operators Cellular they don’t offer anything like this (from analogues I know only “baZa” from Centel, but there the offer is more likely for organizations than for individuals). I also note a fairly convenient and well-filled "Help" section on the service website.
Home phone from "Beeline"
But Beeline is not like that. Firstly, only users of the home Internet or television can use the VoIP solution via the SIP protocol from Beeline. Secondly, far from everything, but only (as I understand it) in Moscow, St. Petersburg, Yekaterinburg, Krasnoyarsk and Ufa. Thirdly, you can use the service only and exclusively from the Beeline network. Moreover, from the Beeline home Internet network.
But not everything is so bad, there are also advantages. The biggest thing: you get a "direct" Moscow (St. Petersburg, etc.) number absolutely free of charge. Now about tariffs. As far as I understand, the rates are different for different cities. Therefore, I ask you to forgive me for bringing here the Moscow tariffs.
There are already 2 tariffs: "Pay only for a minute" and "Unlimited + Beeline RF".
Tariff "Pay only per minute"
- Cost of outgoing calls to landlines Moscow: 0.44 rubles. in a minute;
- The cost of outgoing calls to Beeline numbers: 0.8 rubles. in a minute;
- The cost of outgoing calls to numbers of other operators in Moscow: 1.5 rubles. in a minute;
- The cost of outgoing calls to landline phones in the Moscow region: 1.7 rubles. in a minute.
There is no subscription fee. To use the service, you just need to have more than 50 rubles on your account " home internet or "home television".
Tariff "Unlimited + "Beeline" RF"
This tariff has subscription fee: 300 rubles per month (and if someone does not remember, a month in Beeline is 30 days). For this money you are offered the following:
- Cost of outgoing calls to landline phones in Moscow: 0 rub. in a minute;
- The cost of outgoing calls to Beeline numbers in Moscow: 0 rubles. in a minute;
- The cost of outgoing calls to Beeline numbers in Russia: 0 rubles. per minute (except for subscribers of the Republic of Crimea and Sevastopol);
- The cost of calls to other destinations is the same as in the previous tariff.
Thus, for 300 rubles a month you get unlimited calls to city numbers in Moscow and to all mobile numbers"Beeline" of Russia (so far, except for the Crimean ones). Very profitable, especially compared to MGTS tariffs. But the real "icing on the cake" is the ability to enable free forwarding of incoming calls to the Beeline mobile phone.
Almost any SIP hardware or program/application can be used for use. The main thing is to set it up correctly. How exactly what to set up can be with some difficulty, but you can find it on the Beeline website: Help and support - Home phone. If you have any questions about this topic, ask them in the comments. It seems like he almost learned how to set up some Android applications to work with a Beeline home phone.
That's actually all I wanted to tell you today. As I see, the topic of SIP-telephony is not very popular among my readers. But you have to be patient a little. Soon everything I wanted to talk about this topic will be over.
Virtual PBXs are gradually gaining their popularity. They are cheap, require no maintenance, and provide enterprise customers with decent functionality. Similar Services offered by many carriers, including mobile operator Beeline. What is " Virtual PBX» from Beeline and what can she do? Let's try to consider its functionality in our detailed review.
Possibilities of the "Virtual PBX" service
The service "Virtual PBX" from Beeline is focused on corporate clients. It provides state-of-the-art digital telephony without requiring the purchase of expensive equipment. Previously, office telephony was carried out with the help of institutional telephone exchanges, to which external telephone lines This made it difficult to move. Today, all you need to create telephony in the office is the Internet.
"Virtual PBX" works on the side of the operator, where all the equipment is located. VOIP phones are installed on the client side or softphones- no other equipment is needed here. After connecting and activating the service, you only need to configure the telephones installed in the office. Office employees will be able to communicate with each other, receive external calls, as well as make long-distance calls and around the world.
The equipment for connecting the office to the Internet is purchased by the client independently. Also, it is responsible for choosing a provider - Beeline provides only SIP-telephony.
To organize call centers and customer support services it is possible to connect numbers with the prefix 8-800. Customers will call these numbers absolutely free of charge. In addition, SIP from Beeline provides the following possibilities and features:
- Manual transfer of calls between employees;
- Automatic distribution of incoming calls;
- Providing short numbers for internal dialing;
- Integration mobile phones into a single corporate telephone network;
- Preferential billing of calls within the network;
- Inexpensive tariffs for domestic and international communications;
- Lack of geo-referencing;
- Call forwarding;
- Call holding and waiting;
- Connection of external numbers;
- Set automatic voice greeting.
SIP-telephony from Beeline and the Virtual PBX service is the whole world modern possibilities connections. Using the services of Beeline, each client receives at his disposal a reliable and stable connection through the Internet. Virtual PBX allows you to quickly complete office telephony and distribute phone numbers between employees. Also, on its basis, call-centers are organized to support customers and provide assistance to them. The service is managed via the Internet, using a special web-interface.
When moving the office, all phone numbers are saved - just connect phones and computers to the Internet. In the case of landline telephony, moving without losing numbers would be extremely difficult.
How to activate the "Virtual PBX" service
Are you interested in the "Virtual PBX" service from Beeline, and want to take advantage of this offer? In this case, you need to apply for connection - this is done by the manager responsible for working with telephony. Sending an application from the Beeline website is free of charge. Also, it is possible to order the service at the operator's offices. It will also be possible to specify the tariffs for communication.
The "Virtual PBX" service is available for large customers with more than 50 employees and jobs. Otherwise, the service connection will be denied.
It all started with the fact that another competition for communication for our hotline 8-800 won Beeline. At first, the service was provided simply by redirecting the 8-800 number to one of our external numbers that we receive from another operator. And according to the terms of the contract, we have prescribed the service either via SIP through a dedicated channel, or by redirecting to one of our numbers.
And at some point, Beeline decided that "it's enough for us to serve the service by redirect, let's run the optics to the subscriber and submit it via SIP." In general, we successfully brought in optics, installed a media converter (moreover, it’s not rack-mounted, and without a shelf - we had to adapt it in a rack). A week later, we roughly set up the channel, raised the SIP trunk from our asterisk to their server, and made sure that the telephony was working. Switched 8-800 to this trunk. Everything worked successfully for 2 (jwa!) days.
After 2 days, it turned out that registration on the Beeline SIP server fell off on asterisk. It is not surprising, because. at this time, there was no ping from our SIP server to the Beeline SIP server. Naturally, they immediately created a request for Beeline technical support, before the situation was clarified, they switched 8-800 to the old scheme.
Interestingly, in the course of the proceedings, it turned out that if we ping any IP-shnik in the subnet of the beeline from our SIP server (with the exception of the IP SIP server), then the connection with the SIP beeline is magically restored - pings pass, asterisk registers. It all works for half an hour, then the connection disappears again.
I'll tell you a little about the network configuration. From the media converter of the beeline, the patch cord goes to our switch to the port that is configured in access mode and belongs to the 21st vlan. Further, this 21st vlan gets to the server with telephony (already tagged) and then to the openvz container with asterisk. In this container, there are thus 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 in the beeline, they said that they would conduct checks on their network. In general, we called back and forth for 2 weeks, the beeline tested its network several times, did not find any problems and answered in the spirit of "yes, you have it there local problem, repair your equipment". It was almost impossible to prove anything. In general, I managed to go on vacation and get out of it, but the problem remained the same as it was.
I'll tell you how the communication with technical support was typically built. My colleague Igor drank this case to the fullest while I was on vacation. A call to technical support, we hang on the line for 10 to 60 minutes, we get to a polite employee of the 1st line, who does not understand anything in the technical part, can only create / close an application, add a comment. After explaining who we are, naming the application number, as a rule, a transfer to another specialist, already a technical one, follows. Only now the chance that he will pick up the phone is not at all 100%. Yes, yes, you can hang on the line for an hour and never get on a specialist.
And now about the qualifications of network specialists of the 2nd line. To say she's weak is an understatement. The "specialists" from Beeline have problems with a basic understanding of how IP networks work. I ended up with such a specialist, described the problem to him, and this interesting trick with ping, after which everything suddenly starts working. In itself, this did not interest him, but he wanted to check our settings. Is our IP like this? Is that a mask? What is the SIP server address? Do you have such and such a gateway? This is where I say - the gateway is not indicated 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 the Beeline one are wrapped in our router. It was here that the Beeline "specialist" got really excited. Yes, how is it, why are you, how is it that you did not register our gateway, because of this, of course, everything does not 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. an arp request left us, from desired host the answer came, the IP packet went where it needed to, no packets are sent to the gateway anyway. To hell with it, the comrade doesn’t know how the network works in general - he continues to insist that the gateway is simply necessary for such interaction! He doesn’t want to hear anything, let alone work with his brain and sort out the problem a little.
Well, to hell with you, you will have a gateway. We take a laptop with linux, directly connect it to the media converter, set up the interface as it should, with a gateway. What would you think? Naturally, the problem is reproduced here as here. Half an hour works - then everything, pings disappear.
After that, I contacted the manager who oversees us, explained the situation and the "competence" of their specialists to him, he promised to do something about it.
Indeed, a few days later their specialist came to us with a VoIP gateway and a hardphone. Connected, waited half an hour - the connection disappeared, as expected. At the same time, the specialist kept in touch with someone "from the other side", who could poke equipment, see something there, etc. In the process of these tinkering, it was suggested that it is possible that the IP-shnik that was issued to us is already used somewhere in the Beeline network. He suggested that they look at some of their equipment, what kind of mac they see before and after.
Well And so it turned out - the IP-Schnik issued to us had already been issued to some client before. From here all network problems also occurred. The specialist who came was shocked that they had such a mess. Immediately, as this guess was confirmed, I was just the spitting image of forman.jpg.
