I have a main site with 10 SIP trunks. I am trying to have long distance calls from the remote site to go across the VPN and call out the SIP trunks at the main site. I started by just running a simple test to call a 7 digit phone number from the remote site through the main site. Here is my configuration with example phone number:
Post "debug ccsip messages" and provide called and calling number.
Is there a reason you have H323 between the sites and SIP to SIP trunk, why not configure SIP between sites as well?
Can do you hard code everything to one codec? As your outbour leg from remote site has no codec class or codec specified which means it will default to G729, yet your SIP trunk dial peer has codec class (which you did not post).
You mention VPN connection, Is there any NATing taking place? Are all needed ports permitted?
The effort to search the best and simplest explanation of complex topic is so harder and passionate. You test, you analyze and you imagine the best schema to explain it, even the size of you drawing is taken into consideration.
This is what I did f...
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