I have two questions:
1)Does anyone know how to perform a hook flash on non-zap channel or at least, know how to send a media event 16 down a channel using RFC-2833?
2)I want to change the message that states “The number you have dialed is not in service” for calls that don’t match the inbound routes. How do I change it?
Answer to #2…
Use an ANY/ANY inbound to route the calls wherever you like.
I asked question #1 a long time ago and never got an answer. I did extensive investigation and never found a thread that ever addressed it, and so I reached two conclusions:
There probably isn’t a way, becuase -
There doesn’t seem to be any demand for it besides you and I.
I eventually gave up and switched to SIP trunks exclusively and am much happier (except when they don’t work!).
If you google “asterisk switch hook flash” there are a number of entries addressing it.
I think AdHominem is on the right track. I think from the beginning, FreePbx envisioned the use of SIP or PRI as trunks rather than using POTS. Asterisk does have the Flash() command in its repertoire, but I don’t believe I’ve seen a way to implement it from the GUI.
I’ve been looking into question 1 for a long time. I guess its time to give up on it . The tandem transfer (activated via hook flash, hook switch, etc ) would have freed my PSTN line allowing me more flexibility.
I actually read Ad Hominem’s posts while building my system (especially the obi110 setup) but was hoping someone came up with a solution to the hook flash problem, at least this year.
Btw, is it just me or are PSTN lines are nearly equal in price to VOIP Business Lines. Each of my Business grade PSTN lines cost about $30, unlimited nationwide. Most VOIP Business Lines I’ve seen cost around $24 for a limited set of minutes. Google Voice is the exception but I don’t know whether to count google voice as business reliable.
P.S.: Thanks Bill for question’s 2 solution