No one ever responded to this inquiry but I figured out what the issue
was. I thought I would respond with the solution just in case someone
runs into the same issue in the future.
Firstly, when setting up trunking between servers the username = field
is not optional. :) Also, I had a lot
I've got an issue where I am trying to route calls between Asterisk
Servers. I can route calls inbound to a server but seem to have an
authentication issue going out over the same sip account. It appears
that my server isn't sending the second invite after proxy
authentication request. I
Try www.asterisk2billing.org
On 1/11/07, Pablo Bullian [EMAIL PROTECTED] wrote:
Hi,
I have an issue with the authentication for the outgoing calls.
What I want is to give every user a different password, that they must
enter everytime they make an outgoing call.
What are my possibilities?
You may use astdb for this.
Just set an entry on AstDB with user password and then for every outgoing
call prompt an audio to introduce password and then check if it exists on
AstDB.
User may be the caller ID and the pass is introduced by DTMF.
Then you may use a GOTOIF to allow or not
Hi,
I have an issue with the authentication for the outgoing calls.
What I want is to give every user a different password, that they must
enter everytime they make an outgoing call.
What are my possibilities? and can u show me an example please?
Thanks a lot.
--
'May the source be with you'
