Step by step guide to configure the TATA PJSIP trunk in asterisk based dialers like vicidial, goautodial,Freepbx,elastix,issabel.
Tata PJSIP
If
you are looking for TATA PJSIP trunk configuration in asterisk or
vicidial then this blog post is for you. In this tutorial i have
provided asterisk pjsip configuration for TATA PJSIP trunk.
Tata Tele
Business Services, belonging to the prestigious Tata Group of
Companies, is the country’s leading enabler of connectivity and
communication solutions for businesses. Tata Providing PRI trunks over
VOIP as SIP or PJSIP to indian customers.
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
Check out the article how to enable PJSIP in vicidial.
Tata PJSIP Carrier Details
Once you have purchased the TATA PJSIP trunk, you will be provided with the below details,
DID Range and Pilot NumberUsername 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 address
[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
id:manish.kadiya
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