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?
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The short answer is that you don't.... That isn't entirely true while at
the same time it kind of is, but for the most part you don't configure
the softkeys. You enable or disable them via TCL. Here is the long
answer. Be sure to read the whole thing or e...
Topology: IP Phone > Switches > Microsoft NPS setup to forward 802.1x
proxy to > ISE 2.1 patch 3 Authentication: EAP-TLS using Cisco MIC SANs
Phone Models 802.1X support? 802.1x flavor Addtl Comment EAP-MD5 EAP-TLS
Cisco 3905 Y Y N Cisco 6911 Y Y N Cisco ...