| |
| |
 |
For
IAX2 Routing
|
 |
|
| |
|
|
IAX2
or Inter Asterisk Exchange is
designed for talking from asterisk
to asterisk, so since we use asterisk,
that is the best protocol to use
with didx.
But we do not provide registration
on our network; you have to forward
the calls to your asterisk server.
We
will provide a PUBLIC KEY Soon
that will make sure that the call
is coming from us and no hacking
would be possible on your network
this way.
If you have any suggestions, please
do contact us with them.
Now, to route the call to your
network.
1.
Buy a Number.
2.
Click on IAX
3.
Define the Ring to address of
your asterisk server.
IE guest:guest@didx.org/33170725902
Or if you have a particular user
you want to send the call to,
you can send it to that user.
IE 33170725902:w3s235s22@didx.org/33170725902
If you have any problems, click
on CONTACT US after logging into
your account. Please remember
that DIDX is for SERVICE PROVIDERS
and CARRIERS and if you need professional
support, we do provide that for
50$ an hr, or 25$ for every 20
mins.
|
|
| |
|
|
|
| |
 |
DiD
/ DDi routing for astbill
|
 |
|
| |
|
|
This
Code has been sent to us, by Robin
Dexter of www.dexwebtelecom.com
You may contact them directly
for support.
Tue, 2006-11-28 22:30
Following Skippy's comments on
submitting addon's and fixes,
here's my extension to Mohamed's
DDI code (I hope you don't mind),
no alterations to the database
necessary. the code will automatically
account for IAX2 or SIP.
Click
here for more detail
|
|
| |
|
|
|
| |
 |
Support
|
 |
|
| |
|
|
If
you have any problems, click on
CONTACT US after logging into
your account. Please remember
that DIDX is for SERVICE PROVIDERS
and CARRIERS and if you need professional
support, we do provide that for
50$ an hr, or 25$ for every 20
mins.
|
|
| |
|
|
|
| |
 |
VOIP
Switch.com
|
 |
|
| |
|
|
If
you are using VOIPSWITCH.com and
want to give did's to your customers
using didx.org or you wish to
buy numbers for your calling cards.
You need to set your switch to
accept the calls from our ip address,
and accept the call on g723 or
g729 what ever works best for
you.
You need to create a user in your
system and route the call to that
user ie 12126555763@yourip.com
This will be on SIP in our system.
|
|
| |
|
|
|
|
|