How to Configure/Setup Carrier on VICIDIAL with Switch2Voip.us

Original article: http://ow.ly/zfig302hJgz
VICIDIAL is one of the most used Open-Source Dialers world wide 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 VICIdial Call Center Suite very easily.
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 than have a fully functioning VICIdial system.

Setting the Trunk in VICIDIAL or the Carrier, can sometimes be tricky or time consuming for most customers. We have set up a detailed informative configuration of the Vicidial predictive dialer to help you start making calls.

Click here for a full guide on How to setup your Trunk in VICIDIAL / GOAUTODIAL:
http://www.switch2voip.us/contact-support/how-to-setup-your-trunk-carrier-provider-in-vicidial-vicidialnow-goautodial-vicibox

Asterisk VoIP based dialers: Vicidial, GoAutodial, Vicibox, Vicidialnow Outgoing Configuration Parameters

1) How do I setup my SIP trunk for inbound/outbound calling?

To start making and receiving calls using Switch2Voip please verify that your Asterisk  VoIP server is configured as follows.

We authenticate IP-PBX SIP Trunking traffic by:
IP Authentication (IP address) or Digest Authentication (account and SIP password)

After you decide which switch platform to use, you will need to establish a SIP trunk with our US proxy server sipusa.switch2voip.us and input your IP address into our portal or register your switch with us. Alternatively, if your switch is not in Central or North America, you can use one of our international POPs to reduce transit delays, for Europe and Asia point to 111.235.152.141

Our IP’s

Configure the switch to allow for traffic from Switch2Voip
For our US Server allow: 169.132.196.33 and 206.20.196.19 (for the CLI to work you have to setup both IP’s or use sipusa.switch2voip.us)
For our UK Server allow: 213.166.103.6
For our HK Server allow: 111.235.152.141
For our BR Server allow: 206.20.196.31 and 177.53.194.33
For our PE Server allow: 206.20.196.33 and 169.132.196.43
For our AR Server allow: 206.20.196.34 and 169.132.196.44

Digest Authentication Settings (account and SIP password)

[Switch2Voip]
Peer Detail
username=
type=peer
secret=
progressinband=never
port=5060
nat=auto
insecure=very
ignoresdpversion=yes
host= sipusa.switch2voip.us
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=g729&g711

User Detail
username=
user=<>
type=user
port=5060
context=from-pstn
canreinvite=no
allow=g729&ulaw&alaw

Registration
register=>account:password@sipusa.switch2voip.us

IP Authentication (IP Address)

The 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 applications (such as Asterisk), you can setup your SIP trunk with IP Authentication as follows:

Outgoing Settings:

[out-1]
type=peer
port=5060
nat=auto
insecure=invite
ignoresdpversion=yes
host= sipusa.switch2voip.us
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw
allow=g729

Incoming Settings:

[in-1]
disallow=all
type=peer
port=5060
nat=auto
insecure=invite
host=169.132.196.33
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw
allow=g729

[in-2]
disallow=all
type=peer
port=5060
nat=auto
insecure=invite
host=206.20.196.19
dtmfmode=rfc2833
context=from-trunk
canreinvite=no
allow=ulaw
allow=alaw
allow=g729

If you are using a web-based Asterisk PBX (like FreePBX), 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= sipusa.switch2voip.us
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=169.132.196.33
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= sipusa.switch2voip.us
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=206.20.196.19
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:

– sipusa.switch2voip.us for US
– 213.166.103.6 for UK
– 111.235.152.141 for Hong Kong

In addition, please allow all RTP traffic from any IP Address ports 20000-24000 UDP.

Your USA dialplan should look something like this:

exten => _91.,1,Set(callerid(num)=+1XXXXXXXXXX)
exten => _91.,2,Set(callerid(ani)=Phone number)
exten => _91.,3,AGI(agi://127.0.0.1:4577/call_log)
exten => _91.,4,Dial(sip/${EXTEN:1}@Switch2Voip,55,o)
exten => _91.,5,Hangup

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 campaign to 8 and leave 9 for USA. Copy everything below this line and paste it on your dialer trunk configuration.

exten => _91.,1,Set(callerid(num)=+1XXXXXXXXXX)
exten => _91.,2,Set(callerid(ani)=Phone number)
exten => _91.,3,AGI(agi://127.0.0.1:4577/call_log)
exten => _91.,4,Dial(sip/${EXTEN:1}@Switch2Voip,55,o)
exten => _91.,5,Hangup

exten => _8011.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _8011.,2,Dial(sip/${EXTEN:1}@Switch2Voip,55,o)
exten => _8011.,3,Hangup