[asterisk-dev] SIP, NAT and duplex audio when running Background()
tim.ringenbach at gmail.com
Fri Mar 16 18:55:11 MST 2007
This sounds like a -users question.
On 3/16/07, Lee Azzarello <lee.azzarello at voicedynamix.com> wrote:
> I'm trying to debug this problem and I keep getting stuck and retracing my
> steps. Here's what I have so far:
> Asterisk A ---sip---> SER ----sip----> Asterisk B
> Calling from Asterisk A to Asterisk B gives one way audio ONLY when the
> Background() application runs.
In what sense do you expect full duplex audio while Background() runs?
(You do understand that background plays a sound and waits for an extension
and does NOT go on to the next step until the sound finishes or an extension
is entered, right? There's been confusion on that point in the past)
Unless you're using Monitor or MixMonitor() to record the call, I don't
understand how you can even tell if you have full duplex audio.
-------------- next part --------------
An HTML attachment was scrubbed...
More information about the asterisk-dev