[asterisk-dev] Call routing between two Asterisk boxes using SIP not working ...

[Kevin P. Fleming]

Mauro Sergio Ferreira Brasil wrote:

> Has anyone had this problem ?
> Can anyone help me out on that ?

Please ask configuration/usage/etc. questions on the asterisk-dev
mailing list.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kpfleming at digium.com
Check us out at www.digium.com & www.asterisk.org