I am installing asterisk 1.4 with 2 ISP and i have one card Diguim TE210
with 2 port E1.

now i bought another card Diguim TE410 and I want to add it

the current configuration : connection (WIMAX) from the first ISP and connection (fiber optic) from the secend ISP.

the desired configuration : connection (WIMAX) and connection (radio beam)
from the first ISP.from the second ISP no change (still have the fibre optic)

my question how to active the round-robin in asterisk 1.4 in order to active the 3 technology (WIMAX-radio beam and fibre optic)
any help please

What are best practices for allowing connection by remote SIP
extensions over the internet? I’m thinking of putting the SIP inside a VPN connection.

Kind Regards, Chris

Hello,

I am currently trying to set up CDR logging. I got all the ODBC stuff for my mysql server set up, also checked there’s a connection using “odbc show”
in the Asterisk console:

Name: asterisk


I also created a table for the CDR logs:


This table is also referenced in /etc/asterisk/cdr_adaptive_odbc.conf.

Now, what is the next step?
1) Do I have to call any CDR function in my dialplan for the CDR logging to become really active?
2) Do I have to create more fields in my asterisk_cdr table manually? Guess it should all be done automatically? Would I have needed the ID field at all?

Thanks for any hint


Best regards Stefan

I have a process that runs on a server and does a simple ‘asterisk -rx
“sup show peers’ > /tmp/peers”
and then looks for any “(Unspecified)” items and reports them as having
lost connection.
My server is running 1.4.43 and the two boxes I am monitoring are also
running 1.4.43.
Once in a great while 1 of my boxes reports “(Unspecified)”. I am trying
to find out why.

How can I make the remote boxes have a shorter heart beat to checking
more frequently
with the server so as not to go “(Unspecified)”. By the time I log in
and check its already
back connected again.

Any other thoughts?

Thanks,

Jerry

I am attempting to get an asterisk server to step out of the media
path, but am running into a brick wall. Can someone assist? Here’s my
setup..

Ultimate SIP Provider —> LCR Trunk (Asterisk 1.6) —-> PBX (Asterisk 1.8).

I am attempting to get the trunk to step out of the media stream.
There is no NAT involved, all machines have a public IP.

In the trunk’s sip.conf I have:

directmedia=yes
directrtpsetup=yes

And on the connection to the pbx I have canreinvite=yes

On the pbx I have the trunk connection set to canreinvite=yes.

In the CLI on the LCR trunk I see: