Does sipxbridge have to be behind nat in order for remote users or trunking to work? For trunking I can get inbound calls and make calls, whether or not I use public address for call setup, but have consistent audio issues.
I've been trialling a standby system that I could use at a central location and spin it up in the event it was ever needed and cannot get past the audio problems. Before worrying about call traces and the like, I'd like to get input as to whether anyone has ever done thios before or knows any reason this will not work. Here are a few of the things I've seen: Calls from the telco to an inside user (via DID) get disconnected on the telco side upon pickup, but the internal user goes off hook and the call stays connected indefinitely.Calls from internal user to telco sometime have 2 way audio, other times no audio at all Calls to the AA from a remote user work fine, but calls to the VM system send no audio back. Is there an example where this is known to work or is it technically impractical (gnoring the security implications). I'm wondering if the media relay is not able to work like this, and any input from soemone who know or has tried would be appreciated. Thanks, Tony
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