How to configure ringcentral sip in asterisk vicidial

    Step by step guide to configure the Ringcentral SIP/VOIP account in Asterisk based dialers like Vicidial , Goautodial, Freepbx.RingCentral A leading provider of unified communication and collaboration solutions for businesses of all sizes provides VOIP trunk to dial out and Dial in ,which can be configure in any softphones or hardphones. Using there other Phone options we are going to configure the Ringcentral sip trunk in asterisk

how to configure ringcentral sip in asterisk
ringcentral asterisk

Who is RingCentral?

    RingCentral A leading provider of unified communication and collaboration solutions for businesses of all sizes
RingCentral's cloud-based communication and collaboration platform offers a comprehensive set of capabilities that unify voice, business messaging, team collaboration, video conferencing, and online meetings.
RingCentral revolutionizes the power of the cloud to help companies across the globe work smarter, radically improving the way businesses partner with customers and co-workers.

Pre-Requisites for Ringcentral configuration

1. Ringcentral Portal login credentials.

2. Asterisk pbx console(SSH) access

3. Any GUI configuration utility like vicidial ,goautodial , Freepbx

Ringcentral SIP Trunk

Note: if you have already have the SIP Details from Ringcentral ,you can skip this step 1 and proceed with step 2

STEP 1: Getting the SIP settings for manual Provision

Get the SIP Settings For Manual Provisioning of your RingCentral SIP account.

1. Log in as an Administrator to the RingCentral Online account.

2. Go to Phone System > Phones & Devices. 

3. Under User Phones, look for the Existing phone that you wish to assign to your asterisk.

    Click the existing phone and then press Setup & Provision  or  under action select Setup & Provision. As shown below.

Picture 1:

ringcentral

Picture 2:
ringcentral-provision

4. Go to Other Phones. Under Existing Phone click Select.

ringcentral other phones

5. In next section Disable TLS(secure voice transport to NO) and Select your nearest outbound Proxy , followed that you will get your SIP settings, which will be used in asterisk for registration.

ringcental-SIP

STEP 2: Asterisk SIP Trunk Configuration

1. Go to sip.conf   and  add below sip settings for your ringcentral sip settings.

    you can follow same for vicidial/goautodial under Carrier settings, and in freepbx trunk settings in GUI.

Register string Sample:

register => USERNAME@sip.ringcentral.com:PASSWORD:AUTHORIZATIONID@<outboundproxyDomain:5090>/USERNAME

Example :

register => 1800000000@sip.ringcentral.com:ddde566fus:898989898@sip10.ringcentral.com:5090/1800000000

SIP SETTINGS 
[username]
type=friend
host=sip.ringcentral.com
transport=udp
outboundproxy=sip10.ringcentral.com:5090
fromuser=username
defaultuser=AuthorizationID
username=AuthorizationID
fromdomain=sip.ringcentral.com
secret=PASSWORD
qualify=yes
comtext=trunkinbound
dtmfmode=rfc2833
disallow=all
allow=all
insecure=port,invite
nat=yes
srvlookup=no
usereqphone = yes
callcounter = yes

STEP 3: Asterisk Dialplan

    Add the below dialplan in extensions.conf under your preferred context the default context is [default],
For Vicidial /goautodial use the dialplan in carrier settings.
For Freepbx Use the GUI settings under outbound route.

Dialplan for vicidial /goautodial
exten => _9X.,1,AGI(agi://127.0.0.1:4577/call_log
exten => _9X.,n,Dial(SIP/trunkname/${EXTEN:1},,Tto)
exten => _9X.,n,Hangup()
Plain Asterisk dialplan
exten => _9X.,1,Dial(SIP/trunkname/${EXTEN:1})
exten => _9X.,n,Hangup()

For dialplan pattern matching check my Dialplan entry tutorial
note : Replace trunkname to the name used in Sip settings (username)

STEP 4: Asterisk Cli command to confirm registration

Use the below asterisk commands to check the status of SIP registration.

asterisk -vvvvvr
asterisk>sip show peers
asterisk>sip show registry

INBOUND Configuration

To receive the inbound calls, follow the below steps.
Under the assigned user details for that DID  
click on tab Call Handling & Forwarding and make sure that under the section - Then forward calls to -  you have your new phone line Existing Phone and its toggle 

Conclusion:

    Hope with the help of this article you are able to configure the ringcentral sip trunk in asterisk ,vicidial ,freepbx. For professional support reach me on skype or telegram : striker24x7

2 Comments
  • Ajit Kumar
    Ajit Kumar April 4, 2022 at 9:21 AM

    How to configure ring central SIP in asterisk vicidial

  • Anonymous
    Anonymous September 10, 2024 at 8:08 PM

    Thanks very much for these instructions, they're very helpful! I got through all of them except the last section on INBOUND Configuration. Is that in Ring Central? I can't find any mention of "Call Handling & Forwarding" there, maybe they have moved/renamed it? Also, what does "DID" stand for?

Add Comment
comment url