Up to now I have been using one remote server for both incoming and outgoing. The SIP entry looks like this:

[combined]
disallow=all
allow=ulaw
allow=gsm
allow=ilbc
dtmfmode=rfc2833
host=206.380.260.100
defaultuser=6477957868
secret=xxxxxxxxxxxxxx
insecure=invite,port
type=friend
context=unauthenticated

Now I am switching to a provider that uses one server for origination and two different ones for termination. This is my new config.

[incoming01]
disallow=all
allow=ulaw
allow=gsm
allow=ilbc
dtmfmode=rfc2833
host=205.265.36.100
insecure=invite,port
type=peer
context=unauthenticated

[incoming02]
disallow=all
allow=ulaw
allow=gsm
allow=ilbc
dtmfmode=rfc2833
host=205.265.133.100
insecure=invite,port
type=peer
context=unauthenticated

[outgoing]
allow=ulaw
allow=gsm
allow=ilbc
dtmfmode=rfc2833
host=205.265.22.100
type=peer

Does this seem correct? Is there any way to combine the two incoming ones? Do I have more than I need? Am I missing anything?

TIA.

--
D'Arcy J.M. Cain
Vybe Networks Inc.
http://www.VybeNetworks.com/
IM:da...@vex.net VoIP: sip:da...@vybenetworks.com

--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

Check out the new Asterisk community forum at: https://community.asterisk.org/

New to Asterisk? Start here:
     https://wiki.asterisk.org/wiki/display/AST/Getting+Started

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
  http://lists.digium.com/mailman/listinfo/asterisk-users

Reply via email to