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how to configure TATA PJSIP trunk in asterisk vicidial freepbx

    how to configure TATA PJSIP trunk in asterisk vicidial freepbx

    Step by step guide to configure the TATA PJSIP trunk in asterisk based dialers like vicidialgoautodial,Freepbx,elastix,issabel. If you have purchased the TATA VOIP trunk which supports PJSIP protocol and want to configure the same in your asterisk PBX then this Tutorial is for you. In the article I have provide PJSIP settings required to configure the TATA VOIP in asterisk and vicidial.

    Tata PJSIP Network Settings:

        Tata PJSIP trunk is provided with a dedicated network from TATA tele service, that is you will be provided with a router with dedicated subnet, either you need to have two ethernet interface in your dialer to connect to TATA network and also connect to your existing network, or you need to have a router which can support two networks with proper routing.

    How to enable PJSIP:

    PJSIP is a free and open source multimedia communication library written in C language implementing standard based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE.PJSIP is both compact and feature rich. It supports audio, video, presence, and instant messaging

    Tata PJSIP Carrier Details

    Once you have purchased the TATA PJSIP trunk, you will be provided with the below details,

    DID Range and Pilot Number
    Username and password
    TATA network subnet range
    SIP gateway and Media IP

    Asterisk – Vicidial PJSIP Settings for TATA

        If you are using Plain asterisk or configuring the trunks in command line, then add the below PJSIP settings in pjsip.conf which is located in the /etc/asterisk/ folder.
    If you are using vicidial then add all the pjsip settings mentioned here under the same carrier Account settings.
    Note: you no need to add any registration string for the pjsip trunks.
    Replace the username ,password, IP addressmy username:00918069123456, pass=1234, IP=10.60.41.32

    [tatasiptrunk]
    type=registration
    retry_interval=20
    max_retries=10
    contact_user=00918069123456
    expiration=600
    transport=0.0.0.0-udp
    outbound_auth=tatasiptrunk
    client_uri=sip:00918069123456@10.60.41.32:5060
    server_uri=sip:10.60.41.32:5060

    [tatasiptrunk]
    type=auth
    auth_type=userpass
    password=1234
    username=00918069123456

    [tatasiptrunk]
    type=aor
    qualify_frequency=60
    contact=sip:00918069123456@10.60.41.32:5060
    default_expiration=600

    [tatasiptrunk]
    type=identify
    endpoint=tatasiptrunk
    match=10.60.41.32
    [tatasiptrunk]
    type=endpoint
    transport=0.0.0.0-udp
    context=from-trunk
    dtmf_mode=rfc4733
    disallow=all
    allow=alaw
    allow=ulaw
    allow=g729
    direct_media=no
    rtp_symmetric=yes
    trust_id_inbound=yes
    send_rpid=yes
    ;from_domain=10.60.41.32
    inband_progress=yes
    rewrite_contact=yes
    ;force_rport=yes
    aors=tatasiptrunk

    Asterisk PJSIP Dialplan

    Use the below dialplan to dial over the tata trunk with PJSIP dialplan application.
    If you are using the Asterisk and manging over command line, then add this dialplan in extensions.conf under your preferred outbound context,
    For vicidial users add this dialplan in same carriers settings under Dialplan Entry

    dial plan for Vicidial Users

    exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log)
    exten => _9X.,n,Dial(PJSIP/${EXTEN:1}@tatasiptrunk,30,Tto)
    exten => _9X.,n,Hangup()

    dialpal for plain asterisk users

    exten => _9X.,1,Dial(PJSIP/${EXTEN:1}@tatasiptrunk,30,Tto)
    exten => _9X.,n,Hangup()

    Conclusion:

    Hope this article is helpful for you, if you like this post kindly share and follow.
    Still you are facing issue or need a professional support reach out to be me on skype or telegram – id: striker24x7

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