Asterisk to SIP:#@domain - directly - bypassing proxy

Thank you ahead of time for any help!

We are looking to call a sip application directly and bypass the proxy(s).

As it is, we dial #@domain through our xlite client(logged into asterisk). The call then goes through our trunk, and eventually ends up receiving an “all circuits are busy” message when it tries to hit the destination.

If I call the voxeo sip application directly - with Blink(SIP/Softphone/Utilizes Sip2Sip) I hit the application without a problem.

So in short, my pressing question is if can I get asterisk to call [email protected] DIRECTLY?

Secondly, if anyone can lend some insight into why asterisk to vitelity to sip application doesn’t not work, but Blink to sip application does, I would sincerely appreciate it!!!


yes, create a custom trunk. In the custom dial field, type:

SIP/[email protected]

He means a custom extension !