[Asterisk-Users] Urgent help with Sip <------> H323 on FREEBSD

[Krystian Filiks]

I need some help with getting the following to work
 
SipPhone <------>  Asterisk <------> H323 GK (quintum)
And
H323Phone <------>  Asterisk <------> H323 GK (quintum)
 
I have tried to run the Asterisk from the newest ports and could after
some digging around in the configs register the SipPone to Asterisk and
Asterisk to the H323 GK.
But when I try to make a call from The SipPhone I get Segmentation
faults when I use any type of dial statement
 
This is how I done the ext....config file
 
Exten => _.,1,Dial(H323/<GK IP>)
 
I tried
 
Exten => _.,1,Dial,H323/<GK IP>
Exten => _.,1,Dial(H323/<GK IP>|60|r)
And other sorts of dial statements none of them worked
 
 
Then as suggested on the Web I got the latest CSV Head sources for
Asterisk, Asterisk-oh323-0.6.3a, OpenH323-1.13.5, and pwlib v1.6.6
When I did gmake on OpenH323 I got errors, but hen I installed the
OpenH323 from ports, then after installing the CVS asterisk it was
missing all modules like SIP etc...etc...or config files so I newer
continued
 
Does anyone know how to get this to work or have a idiot-proof guide how
to set up asterisk on linux with voicemail, SIP, H323 and other useful
modules as I'm not that good at Linux
 
/Krystian
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