[Asterisk-Users] Nat...again....

[Mark Woods]

Leif Madsen wrote:

>On Mon, 26 Jul 2004 23:07:46 -0400, Mark Woods <asteriskadmin at fuse.net> wrote:
>  
>
>>I believe so, but I'll check it again.  I did see that in the docs, and
>>I do have both inbound and outbound calls to/from the outside SIP
>>channels working.  This is where I'm baffled...it seems to me that if I
>>can place an inbound call through the server to, say, the Zap/2 channel,
>>and an outbound call to a SIP extension at the same time, from, say, my
>>Zap/1 channel, then I've accomplished the basics and the server should
>>be able to bridge two outside SIP channels (incoming SIP -> server ->
>>outgoing SIP).
>>    
>>
>
>
>Maybe this isn't a NAT issue and is a dialplan issue?
>
>  
>
Not a bad thought, but all of the sip extensions are in the same context:

[extensions]

exten => 5001,1,Dial(Sip/5001,60)     
exten => 5001,2,Voicemail(u9491)
exten => 5001,102,Voicemail(b9491)

exten => 5002,1,Dial(Sip/5002,60)     
exten => 5002,2,Voicemail(u5002)
exten => 5002,102,Voicemail(b5002)

exten => 5003,1,Dial(Sip/5003,60)      
exten => 5003,2,Voicemail(u5003)
exten => 5003,102,Voicemail(b5003)

exten => 5004,1,Dial(Sip/5004,60)     
exten => 5004,2,Voicemail(u5004)
exten => 5004,102,Voicemail(b5004)

....removed for brevity

[sip-extensions]
ignorepat=8
include => beginning
include => extensions

It's beginning to look like maybe I didn't just absentmindedly miss 
something simple, and that I may just have to end up waiting until I can 
test with my users and get a network trace....but on the other hand, 
maybe at lest I *didn't* miss something stupid...  :)