[asterisk-dev] SIP, NAT and duplex audio when running Background()

[Lee Azzarello]

>----- "Tim Ringenbach" <tim.ringenbach at gmail.com> wrote:>
>>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.

I only realized how ridiculous that sounded when you pointed it out. What I meant to say is that an RTP stream is opened up (network traffic is passing through Asterisk A's NIC) but there is no audio on the UA attached to Asterisk A from the recording being played by Asterisk B. This is not a user error on Asterisk B since it worked in the exact same configuration before upgrading to 1.4.

I'm curious if anyone else is interested in this as a bug, since I can reproduce it reliably every time. Should I go directly to bugs.digium.com?

-- 
____________________
Lee Azzarello
Senior Network Engineer
Voice Dynamix
+1 646 367 0732