SIP Trunk between Asterisk and VoIP Provider like Anveo

SIP Trunk between Asterisk and VoIP Provider like Anveo

Are you looking for SIP Trunk from Asterisk to your VoIP Provider to route your incoming and outbound calls via VoIP Provider? If yes, follow the below post which will help you to create SIP Trunks with VoIP Providers.  I have tested it with Anveo and couple of others VoIP Providers, it works great and i assume that same configuration will work for most of the VoIP Providers, you might need to just tweak the configuration.
Assuming you know how to create Inbound Routes and Outbound Routes in Asterisk, i am just putting the Trunk Outbound Settings and Incoming Settings.

Outgoing Settings

username=123456 (The account number provided by your Service Provider)
fromuser=123456 (The account number provided by your Service Provider)
type=friend
secret=mN2P42*1.9 (Password provided by your Service Provider)
port=5060
insecure=port,invite
host=sip.anveo.com (Hostname of your VoIP Provider)
disallow=all
allow=ulaw
context=from-trunk
nat=nonat
qualify=no
canreinvite=no
keepalive=yes

Incoming Settings

username=123456
fromuser=123456
type=friend
secret=mN2P42*1.9
port=5060
insecure=port,invite
host=dynamic
disallow=all
allow=ulaw
nat=nonat
qualify=no
canreinvite=no
keepalive=yes
context=from-anveo

Enter SIP Registration String as following:  

123456:mN2P42*[email protected]:5010

Save the configuration and test your incoming and outgoing calls.

Note: 123456 is the Account Number provided by Anveo and mN2P42*1.9 is the secret provided by Anveo. The same can be replaced by Extension and Password if your VoIP Provider authenticates with Extension and Password.

Hope this helps!!
Cheers!!

1 thought on “SIP Trunk between Asterisk and VoIP Provider like Anveo”

  1. Do you have directions for chan_pjsip? The above is for soon to be deprecated chan_sip which I have never been able to figure out.

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