search for: rxflash

...ike that until the phone is manually picked up and hung up again (or asterisk is stopped/started). I guess this is a function of an unclean hangup (being read as a flash instead of a hangup?). A 'soft hangup zap/27-1' will not do anything (though it makes an attempt). Does shortening the rxflash time sound like it may help this? (Does anyone have a good explanation, or link to one, of the prewink, wink, preflash, flash, start, rxwink, rxflash, debounce timing functions?) Thanks, as always... Steve

...I press the "R" key nothing happens. The other side of the communication hears a "click" but everything stays the same. I have found by trial and error (and in some docs after that) that the standard flash hook timing for my country (Spain) phone systems is 100ms. So, if I put rxflash=100 in Zap/1 channel configuration in zapata.conf (actually any value below 104) Asterisk detects when I press the "R" key, but instead of interpreting it as a flash hook it's interpreted as if I hung up the phone. I get a new dialtone but the other side gets disconnected, not on hold...

...erisk are: ; prewink: 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)

Is there anyway to change the "flash" time on a TDM400 phone port (a user has a phone that seems to generate a short flash which isn't being picked up). Steve -- NetTek Ltd UK mob +44-(0)7775 755503 UK +44-(0)20 79932612 / US +1-(310)8577715 / Fax +44-(0)20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/Mac stevekennedyuk / MSN steve@gbnet.net Euro Tech News Blog

...nnel =>94-108,110-124 context=ddd echocancel=256 echocancelwhenbridged=no usecallerid=yes cidsignalling=bell cidstart=ring threewaycalling=yes callwaiting=no transfer=yes relaxdtmf=no txgain=0.0 rxgain=0.0 language=pt_BR cancallforward=yes signalling=fxo_ks callgroup=1 pickupgroup=1 ;flash=500 ;rxflash=500 ;Cadencias padrao cadence=125,125,2000,-4000 cadence=250,250,500,1000,250,250,500,-4000 cadence=125,125,125,125,125,-4000 cadence=1000,500,2500,-5000 ;Chamada interna cadence=500,300,500,3500,500,300,500,-4000 ;Ring continuo cadence=10000,1,60000,1 callerid="" <7875> context...

...l 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.

...recognition in asterisk, but still have the flash button on the analog phone operational? Could I use these settings in zapata.conf to fix my problem? *prewink*: Sets the pre-wink timing. *preflash*: Sets the pre-flash timing. *wink*: 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." --

Hi, Is there any documentation on the fields prewink, preflash, wink, flash, rxwink, rxflash, start and debounce in zapata.conf? The "Recall" button on my phone doesn't seem to trigger a transfer via my shiny new TDM40B. However, tapping the hook does, but only if I tap it for long enough. Presumably the "Recall" button's timing is too short? Further, most use...

Hello - My single line extension users (connected via channel banks) need to be able to hang up faster. If they just flash the hook it doesn't disconnect right away. Any ideas on how to resolve this? Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

...des? Here are my settings [zapata.conf] [channels] context=from-internal-receiver ; Points to the default context of your extensions.conf language=en faxdetect=none usecallerid=yes signalling=featd ; FXS for ringing phones callerid=asreceived group=0 threewaycalling=yes transfer=yes flash=350 ;rxflash=350 rxwink=300 prewink=20~~ echocancel=no ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=no immediate=no channel => 1-24 signalling=pri_cpe ; pri_cpe = PRI slave ; pri_net = PRI master switchtype=national pridialplan=unknown echocancel=no ; You can set this t...

...lling=featd ; FXS for ringing phones echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived group=1 threewaycalling=yes transfer=yes rxwink=300 rxflash=700 busydetect=yes busycount=10 context=from-internal ; Points to the default context of your extensions.conf channel => 1-24 ztcfg -vv Zaptel Configuration ====================== SPAN 1: ESF/B8ZS Build-out: 0 db (CSU)/0-133 feet (DSX-1) Channel map: Channel 01: E & M (Default) (Slave...

...f ------------- [trunk_1] trunkname = pstn ; GUI metadata busydetect = yes busycount = 3 busypattern = 480,620 ringtimeout = 8000 answeronpolarityswitch = no hanguponpolarityswitch = no callprogress = no progzone = in usecallerid = yes cidstart = ring pulsedial = no cidsignalling = v23 flash = 750 rxflash = 1250 mailbox = callerid = asreceived dahdichan = 1 context = DID_trunk_1 group = 1 hasexten = no hasiax = no hassip = no registeriax = no registersip = no trunkstyle = analog disallow = all allow = all gui_volume = 2 ; GUI metadata signalling = fxs_ks gui_fxooffset = 0 ; GUI metadata rxgain = 0...

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

...Including: ; prewink: 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...

...viously, this is a big problem for us... Below are my zapata.conf, zaptel.conf and extensions.conf: -------------------------- zapata.conf -------------------------- [channels] usecallerid=yes hidecallerid=no echocancel=yes musiconhold=service busydetect=yes ;callprogress=yes busycount=3 flash=20 rxflash=40 transfer=yes threewaycalling=yes ;rxgain=100% ;txgain=1.0 ;relaxdtmf=yes ;------------------------------ context=int_soft ;------------------------------ group=1 callgroup=1 pickupgroup=1,6,7 threewaycalling=yes transfer=yes ;useincomingcalleridonzaptransfer=yes callwaitingcallerid=yes ;echoc...

...Including: ; prewink: 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...

...hidecallerid=no restrictcid=no usecallingpres=no [Audio Quality Options] relaxdtmf=no echocancel=yes echocancelwhenbridged=yes echotraining=yes rxgain=0.0 txgain=0.0 [Call Logging Options] accountcode=slmindustries amaflags=documentation [Timing Parameters] ;prewink= ;preflash= ;wink= ;rxwink= ;rxflash= ;flash= ;start= ;debounce= [Other Features] mailbox=4000 ; channel => 1-23 group = 1 [mrottler] context=mrottler [Call Logging Options] accountcode=mrottler amaflags=documentation [Caller ID Options] callerid="SLMI - M. Rottler <317 123-4567>" callerid= callerid=asrecieved u...

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?