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

[Lee Azzarello]

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. Dialing a DID that rings a SIP UA directly has duplex audio. Asterisk A is behind NAT, Asterisk B is not but Asterisk B's UAs are behind NAT. Both Asterisk A and B were upgraded from 1.2 release tarball to SVN-branch-1.4-r58168 on March 12th. This problem started at the same time.

SIP nat parameters on Asterisk A:

[global]
externip=xxx.xxx.xxx.xxx
localnet=192.168.0.0/255.255.0.0
externrefresh=10
canreinvite=no

[user agents]
nat=no
canreinvite=no

SIP nat parameters on Asterisk B:
[global]
nat=no
canreinvite=no

[user agents]
nat=yes
canreinvite=no

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