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

[Nathan Clemons]

Turning on qualify=yes, or qualify=60, seems to break the BroadVoice
connection (it goes from UNKNOWN to UNREACHABLE and calls fail).

I'm wondering if they don't support OPTIONS probing or something.

-- Nathan Clemons


On Fri, Apr 16, 2010 at 3:22 PM, Jeff LaCoursiere <jeff at jeff.net> wrote:

>
>
> 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
>
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