search for: flashtim

...have been trying for some time now to make the hook flash work on the FXS port. I am using Asterisk 1.4.10.1 with zaptel 1.4.4. When I "manually" flash the hook I can manage to find the duration to put a call on hold. However when pushing the flash button it never works. The phone's flashtime seems to be too short. I tried to set a shorter flashtime in the zapata.conf file, but it seems to be ignored. I have flash=100 configured in the zapata.conf en when reloading it, this is what is reported on the asterisk console: == Parsing '/etc/asterisk/zapata.conf': == Parsing ...

...nk: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ; rxflash: Receiver flashtime (default 1250ms) ; debounce: Debounce timing (default 600ms)

...be reused and if the time interval is right, the telco will interpret this as a hook-flash. Is there a way to set a minimum 'on-hook' interval on a (zap) channel? And while I'm on zapata questions, what is the definition of debounce and is the assumption that flash is the (intentional) flashtime sent by asterisk & rxflash the approximate time understood to be a flash (for instance from a phone)? Thanks, John This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivirus.

Hello I need Asterisk (1.4.39.2) to simulate a flash hook (ie. hitting the "R" key on European handsets) so I can put a call on hold, dial a second number, and set up a conference call. By default, linux/include/dahdi/kernel.h sets the flashtime to 750ms, which appears to be too long for European telcos, as they seem to expect a line cut of about 100ms. After editing the DAHDI_DEFAULT_FLASHTIME accordingly, I recompiled/upgraded Dahdi, and ran the script, but Flash() still hangs up the call: =========== [from_fxo] exten => s,1,Wait(2...

We are having line noise issues in our Asterisk to legacy PBX integration. All SIP calls originating from IP phones sound crystal clear. All calls that originate from the legacy PBX (Isoetec 228) and route through the Asterisk and out SIP have a lot of line noise. I believe I have it pinned down to these Blue Alarm errors that I can see on the legacy PBX side. zttool shows no alarm but when I

...nk: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ; rxflash: Receiver flashtime (default 1250ms) ; debounce: Debounce timing (default 600ms) ; rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes...

...nk: Pre-wink time (default 50ms) ; preflash: Pre-flash time (default 50ms) ; wink: Wink time (default 150ms) ; flash: Flash time (default 750ms) ; start: Start time (default 1500ms) ; rxwink: Receiver wink time (default 300ms) ; rxflash: Receiver flashtime (default 1250ms) ; debounce: Debounce timing (default 600ms) ; rxwink=300 ; Atlas seems to use long (250ms) winks ; ; Whether or not to do distinctive ring detection on FXO lines ; ;usedistinctiveringdetection=yes ; ; Whether or not to use caller ID ; usecallerid=yes cidsignalling=v23 cids...

Help. Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'? Why is it that when I call my asterisk phone number, I get a fast busy?