Learn how to setup SIP trunk accounts on Vicidial to start making and receiving calls using Switch2VoIP provider, verify that your Asterisk or VICIdial server is configured following these instructions.
VICIDIAL is one of the most used Open-Source Dialers worldwide for call centers using VoIP to make calls all over the globe. Vicidial can handle anywhere between 2 to 300 or more agents at a time making thousands of calls per day with great capacity.
VICIBox Server is the authorized installation CD for the VICIdial Call Center Suite. It’s based on OpenSuSE server and will correctly install the VICIdial Call Center Suite very easily. It can be downloaded from here: http://vicibox.org/server/index.html
It is recommended to have basic operating system skills when installing VICIBox but it is not necessary. If you follow the basic Vicidial Installation Manual you will then have a fully functioning VICIdial system.
How to Configure/Setup Carrier Settings on VICIDIAL
We authenticate IP-PBX SIP Trunking traffic by the two common methods:
1- Digest Authentication (Username and SIP password) and 2- IP Authentication (IP address)
After you decide which dialing platform to use (Vicidial, Goautodial) you will need to establish a SIP trunk with our US proxy server 176.9.138.209 and input your IP address into our portal or register your switch with us.
Configure your dialer to allow traffic from Switch2VoIP IP 176.9.138.209
Video instructions: How to configure SIP trunk on Vicidial carrier settings
In this video you will learn how to configure SIP trunk on Vicidial carrier settings, we made this video based on the instructions explained this article: https://youtu.be/_iZUVKeFnlQ
Setup SIP trunk with Digest Authentication Settings (Username and SIP Password)
With the digest authentication method, you will be able to setup SIP trunking on Vicidial using the username and password provided at the moment of the signup.
1- Login to your Vicidial from the administration panel, on the left side go under the Admin section and click on Carriers, then click on Add A New Carrier on the top side.
2- Fill in the Registration String text box with the username and password from the VoIP provider.
register=>username:password@144.76.245.228:5060/pass
3- Go to Account Entry to enter the parameters for this new carrier, make changes according to your username and password.
[switch2voip] disallow=all type=friend username=yourusername secret=yourpassword host=176.9.138.209 dtmfmode=rfc2833 context=trunkinbound qualify=yes nat=yes insecure=port,invite allow=ulaw allow=alaw
4- Globals String: The carrier name has been defined as [switch2voip], so the Globals String should be defined as [switch2voip] under the SIP protocol.
switch2voip=SIP/switch2voip
5- Dialplan Entry: Example. Fill in the dial plan entry to dial with 1 for calls to the USA.
exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@switch2voip,,tTo)
exten => _91XXXXXXXXXX,3,Hangup
Setup SIP Trunk option #2, Digest Authentication Instructions
If the configurations settings above did not work to setup SIP trunk, it may be because you are using a different version of Vicidial or Goautodial, in that case, you can try the second setting option:
Here you will also use your username and password to setup SIP trunk on Vicidial. As you can see in the picture below the only difference from the above settings is that we have added the line “fromuser=”, it will work depending on your dialer version.
1- Login to your Vicidial from the administration panel, on the left side go under the Admin section and click on Carriers, then click on Add A New Carrier on the top side.
2- Fill in the Registration String text box with the username and password provided by the VoIP company.
Registration String: register=>username:password@176.9.138.209:5060/pass
3- Go to Account Entry to enter the parameters for this new carrier, make changes according to your username and password.
[switch2voip] disallow=all type=friend fromuser=yourusername username=yourusername secret=yourpassword host=176.9.138.209 dtmfmode=rfc2833 context=trunkinbound qualify=yes nat=yes insecure=port,invite allow=ulaw allow=alaw
4- Globals String: The Vicidial account entry has been defined as [switch2voip1], so the Globals String should be defined as [switch2voip1] under the SIP protocol.
Globals String: switch2voip=SIP/switch2voip
5- Dial Plan: This field allows you to setup the options to make outbound or inbound calls.
exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@switch2voip,,tTo)
exten => _91XXXXXXXXXX,3,Hangup
IP Authentication (IP Address) Vicidial SIP Trunk Configuration
To setup SIP trunk on Vicidial with IP authentication method is normally simpler to provision and should be used only when you have a static IP Address. It is also somewhat more secure since your SIP trunk can only be used from the IP Address you provide.
With an open-source application (such as Asterisk), you can setup your SIP trunk with IP Authentication as follows:
Copy the values from this template, please note the Registration String is not needed:
[switch2voip] disallow=all type=friend host=176.9.138.209 dtmfmode=rfc2833 context=trunkinbound qualify=yes nat=yes insecure=port,invite allow=ulaw allow=alaw
Dial plan for Vicidial SIP trunk setup with IP authentication
exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log) exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@switch2voip,,tTo) exten => _91XXXXXXXXXX,3,HangupAsterisk based dialers: VICIdial, GoAutodial, Vicibox, Vicidialnow Outgoing Configuration Parameters
For web-based Asterisk PBX (like FreePBX) the IP authentication setup is slightly different:
In “Outgoing Settings”, name the section “out-1”
Then, in “Peer Detail”, enter the following:
type=peer port=5060 nat=auto insecure=invite ignoresdpversion=yes host= 176.9.138.209 dtmfmode=rfc2833 context=from-trunk canreinvite=no allow=ulaw allow=alaw allow=g729
In “Incoming Settings”, name the section “in-1” in “User Context”. Then, in “User Detail, enter the following:
disallow=all type=peer port=5060 nat=auto insecure=invite host=176.9.138.209 dtmfmode=rfc2833 context=from-trunk canreinvite=no allow=ulaw allow=alaw allow=g729
After this has been completed, you will have to create a separate trunk. For the second trunk, name the outgoing “out-2” and again enter the following information:
type=peer port=5060 nat=auto insecure=invite ignoresdpversion=yes host=176.9.138.209 dtmfmode=rfc2833 context=from-trunk canreinvite=no allow=ulaw allow=alaw allow=g729
Then, for the second trunk, name the incoming “in-2” and again enter the following information:
disallow=all type=peer port=5060 nat=auto insecure=invite host=176.9.138.209 dtmfmode=rfc2833 context=from-trunk canreinvite=no allow=ulaw allow=alaw allow=g729
No registration string is required for IP Authentication.
Please make sure to configure your router/firewall to allow traffic from:
176.9.138.209 for US176.9.138.209 for UK
176.9.138.209 for Hong Kong
In addition, please allow all RTP traffic from any IP Address ports 20000-24000 UDP.
The dial plan should look like this
Dialplan Entry:
US dialplan:
exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@switch2voip,,tTo)
exten => _91XXXXXXXXXX,3,HangupUK dialplan:
exten => _944.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _944.,2,Dial(SIP/${EXTEN:1}@switch2voip,,tTo)
exten => _944.,3,HangupAustralia dialplan:
exten => _961.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _961.,2,Dial(SIP/${EXTEN:1}@switch2voip,,tTo)
exten => _961.,3,HangupUniversal Dialplan:
exten => _847.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _847.,2,Dial(SIP/${EXTEN:3}@switch2voip,,tTo)
exten => _847.,3,HangupDial plan for United Kingdom and/or Australia
If you are also dialing to the United Kingdom or Australia and you want to use both USA and UK dialplans then your dialplan for UK and USA should look like this:
Make sure you change the prefix on your UK and USA campaign to 9. Copy everything below this line and paste it on your dialer trunk configuration.
exten => _91XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91XXXXXXXXXX,2,Dial(SIP/${EXTEN:1}@switch2voip,,tTo)
exten => _91XXXXXXXXXX,3,Hangup
exten => _944.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _944.,2,Dial(SIP/${EXTEN:1}@switch2voip,,tTo)
exten => _944.,3,HangupFeel free to contact our Live Chat VoIP Tech Support if after following this tutorial you are still having issues with your Vicidial Carrier registration or your calls are not connecting the way they should.
Additional links and information about setup SIP trunks
- Setting CallerID in Vicidial Asterisk Based Predictive Dialer
- Configure your softphone, X-Lite, X-Ten or Eyebeam to start making calls with Switch2VoIP.us
- How to select Codec G729 on Eyebeam to reduce bandwidth
- Video how to setup Carrier Settings on Vicidial and Goautodial
- Asterisk SIP trunking configuration
- Download Vicidial
