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?
Is there a way in UCCE with using an admin script maybe using the email application in Call Studio to send out an email? I know in UCCX you can send out an email if the script were to fail but is there a way to do it in UCCE? Any help would be great!
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