How to setup your Trunk / Carrier / Provider in ViciDial - VicidialNow - Goautodial - Vicibox - 4.2 out of
Asterisk 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 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 18.104.22.168
Configure the switch to allow for traffic from Switch2Voip For our US Server allow: 22.214.171.124 and 126.96.36.199 (for the CLI to work you have to setup both IP's or use sipusa.switch2voip.us) For our UK Server allow: 188.8.131.52 For our HK Server allow: 184.108.40.206 For our BR Server allow: 220.127.116.11 and 18.104.22.168 For our PE Server allow: 22.214.171.124 and 126.96.36.199 For our AR Server allow: 188.8.131.52 and 184.108.40.206
Digest Authentication Settings (account and SIP password)
User Detail username=<account> user=<> type=user port=5060 context=from-pstn canreinvite=no allow=g729&ulaw&alaw
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:
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=220.127.116.11 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 - 18.104.22.168 for UK - 22.214.171.124 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: