Tony, If it comes to it, I'll just instruct the client to dial a prefix on the fax machine and handle it from there.
Not elegant, but hey -- they are coming from an Asterisk setup so thus far they are head over heels over the fact that everything just plain works. They were on a FreePBX system and were suffering the typical Asterisk maladies... Dropped calls, one-way-audio, echo. They're on all of the same hardware now (server, network components, and handsets) and have no problems with anything at all. With SipX, the issues just vanish... (Incidentally, my first experience with VoIP was with Asterisk and it was a complete disaster. I had a third-party vendor whose business was Asterisk build a setup for a 50-person shop. Same problems as above. Dropped calls, one-way audio, echo, near hourly core dumps... It was a complete nightmare!) -- Robert On 2/1/2012 10:01 AM, Tony Graziano wrote: > We do this with the Patton's via a dialplan entry but I don't mess > with the Cisco/Linksys stuff so I can't say how flexible that is. They > do have manuals and forums for this kind of thing and you will > probably find your answer pretty quickly in their forums. > > Another route I've tried with mixed results... put the gateway and the > ATA/User in a different branch and try to get the branch to handle the > gateway permissions. It looks like this is "broken" or not finished > yet though, and there are JIRA's on it. > > _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
