Sipgatge trunk with Asterisk/FreePBX
I’ve used Sipgate for the past few years with my Asterisk box and have been pretty impressed.
For anyone else looking to use Sipgate with Asterisk/FreePBX here is my trunk setup
Trunk Name: Sipgate
PEER Details:
username=1234567
type=peer
secret=XXXXXXXX
qualify=yes
nat=never
insecure=very
host=sipgate.co.uk
fromuser=1234567
fromdomain=mydomain.com
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=yes
authuser=1234567
allow=ulaw
Register String:
1234567:XXXXXXXX@sipgate.co.uk/1234567
PEER Details:
username=1234567
type=peer
secret=XXXXXXXX
qualify=yes
nat=never
insecure=very
host=sipgate.co.uk
fromuser=1234567
fromdomain=mydomain.com
dtmfmode=rfc2833
disallow=all
context=from-trunk
canreinvite=yes
authuser=1234567
allow=ulaw
Register String:
1234567:XXXXXXXX@sipgate.co.uk/1234567
If you are using NAT between your Asterisk box and Sipgate you will need “canreinvite=no” and “nat=yes” or you will probably get one way audio only on your calls
Edit: context changed to “from-trunk” to make the post clearer
Related posts:
- CallCentric trunk setup with Asterisk/FreePBX
- Getting the DID number from a CallCentric SIP trunk for FreePBX
- CallCentric with FreePBX - free trunk setup
wiseoldowl:
I copied this post over to http://www.freepbx.org/support/documentation/howtos/howto-setting-up-voip-provider-trunks/sipgate-u-k - hope you don’t mind. I will note that FreePBX users should change the context statement to context=from-trunk (if by some chance you then have trouble creating a working inbound route, please see the page at http://www.freepbx.org/support/documentation/howtos/how-to-get-the-did-of-a-sip-trunk - the registration string format given above SHOULD should avoid this problem, but just in case…)
9 November 2008, 5:55 pmmatt:
Copying the post is certainly OK. Many thanks for your comment and the clarification on the context setting.
Matt
9 November 2008, 6:42 pm