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I asked this on the list over the weekend, and likely missed a few people inboxes. I'm having a problem pulling data from RTPAUDIOQOS. For testing purposes I have asterisk sending QOS data to the console. It seems I get QOS data only if the caller hangs up, with the variable being empty if the callee (or asterisk) hangs up. Any idea why I would see this? exten =>

Hello, I have a question about best practice (or recommended practice) for allowing SIP registrations from the Internet. This is what I was thinking of implementing: 1. Use OpenSips for the SBC, enable SRTP and TLS 2. Allow limited access to the actual Asterisk PBX (behind firewall) via OpenSips Is there anything that I am missing that probably should be implemented? Thanks, Richard

I'm playing around in a lab, and I was wondering if its possible to have Asterisk act similar to that of a Avaya PBX, where we have media gateways do the heavy lifting. This is what I was thinking of trying. 1. One asterisk server will contain the logic of the phone system (ex: queues, extensions...etc). 2. The mains server will not handle RTP traffic, it will send the RTP traffic to

I'm having a problem pulling data from RTPAUDIOQOS. For testing purposes I have asterisk sending QOS data to the console. It seems I get QOS data only if the caller hangs up, with the variable being empty if the callee (or asterisk) hangs up. Any idea why I would see this? exten => h,1,NoOp(RTPAUDIOQOS: ${RTPAUDIOQOS}) Thanks, Richard Seguin -------------- next part