I finally dug this up and wanted to post it here in case someone is digging around for the rules:
1) Use the already included SIP protocol information
2) DEPENDING ON YOUR CONFIGURATION Please check \etc\asterisk\rtp.conf for the port range your setup is configured to use for the Asterisk RTP protocol. Most commonly it is
UDP 1:65535 -> 10000:20000
You may also have to configure your VOIP client to only use that range of ports. Good luck!
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