[asterisk-users] Testing a sip call through Asterisk?

[Jeff LaCoursiere]

On Fri, 16 Apr 2010, Nathan Clemons wrote:

> I'm looking to find a test tool that will register with our Asterisk
> (Trixbox) server here at work and place an outgoing call via our main SIP
> trunk (BroadVoice) to confirm that things are working. I've looked around
> but I can't seem to find any tools that will do what I'm looking for.
>
> I can't just monitor the status of the trunk inside Asterisk, as this is the
> normal status:
>

[snip]

just add "qualify=yes" to your context and it will monitor the RT latency.

j