Multiphone from Megaphone - use an alternative SIP client

    I do not want to discuss the appropriateness of the service as such - this has already been discussed in other topics ( # 1 , # 2 ). In any case, the service may be useful to the end user, for which I express my gratitude to Megafon.

    I also want to admire the employees of Megafon who were preparing the product for release - because it seems that this is a completely closed product ... but even a superficial study allows us to understand that we can use the basic functionality of “Multifon” using any SIP client and a regular WEB browser.

    I’ll make a reservation right away - the service is interesting only for Megafon-Moscow subscribers, I went to the nearest service office for testing and bought a SIM card there.

    So, to use the service we have to go through the following steps:
    1. Registration
    2. Connection of the service (for using the SIP-In and SIP-Out services)

    Well, during operation we will need:
    3. Making / receiving voice calls
    4. Switching the operation mode (only GSM / only PC / PC + GSM)

    Of course, all these operations can be performed with the help of a native client , but this is not always possible. Immediately I propose an alternative way.

    1. Registration


    WEB registration is available at: sm.megafon.ru/sm/register
    I advise you to create an account like <your phone number> @ multifon.ru when registering.

    These links may also be useful:
    Password recovery page: sm.megafon.ru/sm/ recover
    Your profile: sm.megafon.ru/sm/profile?login= YOUR_LOGIN@multifon.ru&password=YOUR_PASSWORD

    2. Service connection


    To activate the service you need to dial * 137 # <send call> on your phone (already registered in "Multifon") and then follow the instructions in the menu.

    A small addition for those who, like me, will also buy a new Megafon SIM card - the service can be used 2-6 hours after making the first call from your SIM card. Immediately after purchase, the card is in non-activated mode and Megafon software complexes require some time to activate the SIM card

    3. Making / receiving voice calls


    Some time after the activation of the service (not immediately!), The platform will update your profile and allow you to make outgoing calls or configure the mode of work with incoming calls.

    Now it is time to configure your SIP client to work with "Multifon".
    I used SJPhone and I will give examples based on it:
    1. Go to Menu => Options => Profiles and click on the “New ...” button:
    Image
    Specify the profile name, set “Profile Type” = “Calls through SIP Proxy” and click “OK”

    2. Select the “SIP Proxy” tab:
    Image
    And fill in the fields:
    Domain / realm: multifon.ru
    Use outbound proxy: yes
    Proxy URI: sip: sbc.megafon.ru: 5060; transport = TCP
    Proxy usage mode: Strict outbound proxy

    3. Select the “SIP Registration” tab: We
    Image
    expose (I’m not sure about this field, maybe everything will work correctly without it):
    Transport of registered contact: TCP

    4. Save the account (“OK” button) and select only that the created account click "Initialize ...": We
    Image
    write:
    Account: your identifier (without a domain name!). If during registration in the service you took advantage of my offer and indicated your phone number, then indicate your number in this field; if registered by nickname - indicate your nickname.
    Password: password for the service

    5. Click “OK” in the “Options” window
    Image
    and after that you should see this screen:
    Image
    The inscription “Multifon Ready to call” means that your SIP client has successfully registered for the SIP Proxy of Megaphone and is ready to work.

    If you have already activated your client, then from this moment outgoing calls are available, you need to dial the number in the format 7XXXYYYYYYY, i.e. no plus at the beginning of the number

    4. Switching the operating mode


    Of course, it is much more convenient to do this from the native Megafon client. But what if you do not have the opportunity (or desire) to use it? Here you will be saved by an ordinary ... WEB browser.
    Suppose your phone number is 79261234567 and your password is aaaBBB

    Get Current Forwarding Status

    We follow the link of the form:
    sm.megafon.ru/sm/client/routing?login=79261234567@multifon.ru&password=aaaBBB
    In response we get an XML page:
    Image
    An important parameter for us:
    1

    it just indicates the forwarding mode of incoming calls to your phone:
    0 = Only GSM
    1 = Only PC
    2 = PC and GSM

    Set new call forwarding status

    sm.megafon.ru/sm/client/routing/set?login=79261234567@multifon.ru&password=aaaBBB&routing=XXX
    at the same time instead of “XXX” - specify the redirection mode you need.
    You can get either the following answer:
    Image
    i.e. “Operation completed successfully”

    Or this:
    Image
    I got this answer when the Multifon service was not connected on the SIM card.

    If someone suddenly wondered where all this information came from ... I suggest installing a megaphone client and carefully looking at the contents of the files:
    * defaults.xml
    * customize / vendor.xml
    from the directory in which Multifon was installed.

    ps:
    * I have nothing to do with the megaphone
    * I had never before used a mobile connection from Megafon, I bought a SIM card specifically for testing the "Multifon"

    UPD (08/04/2009): Connection to Asterisk


    To receive incoming calls to asterisk, you need to add the following lines to sip.conf:
    1. In the [general] section,
    register => tcp://ваш_sip_id_с_доменом:ваш_пароль:ваш_sip_id_без_домена@sbc.megafon.ru
    example: with
    register => tcp://79251234567@multifon.ru:password:79251234567@sbc.megafon.ru
    this line we ask asterisk to register on the Megaphone SIP proxy server

    2. Add a section: it was possible without this section , but there is one caveat - DTMF codes are transmitted not in signaling (as is done by many SIP providers and is considered a good tone rule), but inside the voice stream, so Asterisk has to decode the DTMF on its own.
    [multifon-in]
    type=peer
    host=sbc.megafon.ru
    dtmfmode=inband


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