Hi,
I have question about translating H.323 signals to SIP signals, vice versa. How does it get done in the code ? Which file or section code is doing the translation. For example, converting H.323 setup to SIP invite.
I am a newbie to asterisk. I appeciate for all your help.
This is very complex process and all conversion usually done within Asterisk internals, and not related to H.323/SIP channels. For example, every incoming call will go through dialplan (extensions.conf) where any modifications into call flow and call parameters can be made.