[Asterisk-Users] 30 button vip 1 way audio

[Bill Coward]

It seems this is a redundant question (or at least problem) within
this group, but I'm unable to find a solution/combination.

I have 3 30 button VIP phones running behind 3 different firewalls/servers (NAT)

 My asterisk server is running great with a public IP address (no NAT)

 The 30 button phone work fine to a point (load, register, date&time,
and process calls to and from) but

Only get one way audio, and my skinny debug shows the internal ip
address (10.x.x.x) rtp session packets leaving (attempting,
firewalled) to leave my server.  Which geek nob do I need to turn to
get my server to initiate the rtp on the public address...

I've tried the nat=yes (and no...) statement in the skinny.conf tried
host=dynamic host= (ip address)
 

[billvip]
device=SEP00D0BA8477CC
version=P00203010003
context=default
line => 1313
nat=yes
host=24.159.111.193
callerid="Bill VIP30" <1313>
mailbox=1313

tried nat in the sip.conf?


[general]
context=default     ; Default context for incoming calls
port=5060   ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0    ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes     ; Enable DNS SRV lookups on outbound calls
nat=yes

 
Using: chan_skinny.so that came with the Asterisk 1.0.7 dist.

 PS x-lite clients work fine 

 Anyone who can help a thousand thank you's in advance !!!!!!

 
Bill Coward, MSCE, CCNP, CISSP, CCSE, SCSA
Sr. Network Engineer
Graphic Packaging International