search for: hookswitch

Ok So when i get into a context and dial a number that does not exist the 'i' exten works fine. I have a test exten that runs my menu system if i dial that then dial an exten that does not exist ' i ' works. BUT when i dial a number directly that does not exist the 'i' exten does not get called i did try exten => ${INVALID_EXTEN}, Do summit ( no good ) What i need

...ink*: Sets the wink timing. *rxwink*: Sets the receive wink timing. *rxflash*: Sets the receive flash timing. *flash*: Sets the flash timing. *start*: Sets the start timing. *debounce*: Sets the debounce timing. "The debounce settings in the Asterisk configuration affects how Asterisk handles hookswitch transitions on its FXO/FXS interfaces." --

On Tue, 2005-01-18 at 09:49 -0600, asterisk-dev-request@lists.digium.com wrote: > > Hi, > > I am Arnaud F?vrier, I teach in a technical university in Marseille. > > I'd like to know if is is possible to connect a very old phone to > asterisk and dial pulses with it? > > Are digium cards pulse dial compatible? > > Is there any specific configuration

I can't seem to make the "Hold" button function on the GS BudgeTone-100. I'm trying a procedure like this: 1) On a call 2) Press "Hold" button 3) Hang up phone What I expect is for the call to go on hold until I pick up the receiver again. (Like my SPA-2000's, except it's a flash, but I can hang up the phone and the call waits there for me to pick it up

Hi All... I have installed a single X100P card in my PC and am playing with Asterisk. The wire I plugged into the X100P has two POTS lines on it, wired on the RJ45 in the normal way. I am getting odd behavior. It seems when I dial out that the X100P dials both lines at the same time. I have two questions. First, I see that the X100P is only a single channel. Does this mean that I can

Hi: I am looking into getting incoming IAX DID channels for our office. I've found a provider. What I want, though, is an incoming hunt group -- that is, say we have three lines: 555 1212 555 1213 555 1214 Calls coming in on 555 1212 may end up on any one of the three. If 555 1212 is busy, the call forwards to 555 1213, and so on. I was under the impression that this has to be done by the

Hey gang, I'm looking at escaping from a Nortel Meridian CISC system to Asterisk/Digium/SIP phones. I'm currently in the testing and proof of concept phase. I'm going to need a SIP phone and don't want to re-purchase and have "orphans" around. We currently run Nortel 7310 phones and they work great. I'm sort of overwhelmed by all of the different IP phones. I