search for: interdigit

Hello list, does anyone know how to change the "interdigit timeout" when using Cisco IP Phone 7940/7960 with SIP-Firmware and Asterisk? it's default value is 15 sec. but i have nothing found to set this in tftp-config file etc. Thanks in advance, Roland

...dial progress than what I'm seeing with the SPA-3000 - How do the TDM cards, the Mediatrix 1204, or other PSTN interfaces do in terms of dial progress? I've also noticed that a dial plan for line 1 (FXS) on the SPA-3000 of only: (*5) still does not connect the call until after the interdigit short timer has expired. Especially as the *5 is the only thing in the SPA-3000's dial plan, there are no other dial candidates to match. If I do the same thing with a dial plan of (0) the call is connected right away. The SPA-3000 manual says this about interdigit timeouts: "The inter...

Hi. I have a SPA3102 updated with with Software Version: 5.1.7(GW). I have this settings on Voice/Regional: Interdigit Long Timer: 10 Interdigit Short Timer: 3 Anyway, when hooking up (without dialing anything), the timeout starts after 3 seconds. It's like the Long Timer is unused. After dialing, the Short Timer is also used to timeout. Is that normal? Am I missing something? Thanks. -- Rodolfo Alcazar Re...

Hello List Inhabitants: I don?t know what this operation is called or if there is a function that does it automatically, hence I seek your help! If I divide a large data collection tasks between two students, and I have a master list of samples, but one student records some of the values, and the other student the remainder, I need to get the two sets of student measurements into one column.

I have a Sipura 2000 working fine, but whenever I dial any extension there is a delay of 5-10 seconds before it starts ringing. However, if I dial the extension and hit the pound key after the number, it goes through right away. Is there any way around this?

Dear all with image I can plot only one set of values in one plot. Do somebody have any insight how to put those 2 matrices into one picture so that in one cell in image picture are both values from mat[1,1] and mat2[1,1]. mat<-matrix(1:4, 2,2) mat2<-matrix(4:1,2,2) x <-1:2 y <-1:2 image(x, y, mat) image(x, y, mat2) The only way I found is to mix x or y for both matrices let

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Hi, Which is the correct syntax to call using IAX? I have two Asterisk boxes behind a NAT and one of them use the default port 5036 for IAX, the second one use 5038. To call an extension of the first one, the line in extensions.conf is: exten => _9XXX,1,Dial(IAX/user:pass@195.3.32.191/${EXTEN:1}) and for the second one: exten => _8XXX,1,Dial(IAX/user:pass@195.3.32.191:5038/${EXTEN:1})

...that. My current problem is 'blending' of digits. For example, if I receive pulses for 8 0 1 or 8 0 8 they are correctly recognised almost all of the time. If however I receive 8 8 8 they are almost never correctly translated. Essentially, it seems to be something to do with timing for the interdigit gap. I have tried terminating the lines on a Norstar MICS DID card w/Immediate Pulse and all digits are correctly received all the time. Can anyone suggest the appropriate timing parameter to twiddle to adjust the interdigit detection? Switching to Wink or Delay Start to get DTMF is not an option...

...#define ZT_PULSETIMEOUT ((ZT_MAXPULSETIME / 8) + 50) And the pulse detecion loop that consumes these parameters begins at line 4866 of zaptel.c The intermittant loss of pulses from second and/or subsequent digits appeared to be caused by a slightly too long blanking period that occurs after the interdigit timeout occurs. Lowering MAXPULSETIMEOUT, recompiling and reloading the zaptel.o modules appears to have fixed the problem. Likely this is happening because they are CO generated DID pulses and hence tuned for as fast a signalling rate as possible. For the interested (or bored) far too much inform...

Hei! Does anyone know how to configure this phone to autodial the number after interdigit timeout has passed? Rennes

Possibly OT? I've hooked up a Linksys PAP2 to my Asterisk 1.6 fairly painlessly. The only issue I can't beat with it is the dial delay when calling internal or external numbers. No matter what it seems to take 10 -15 seconds to actually dial. I've altered the device removing all *xx combos and unnecessary waffle and cut the dialplan string to (x.S0) but the problem persists. Anyone

My question: How do I configure AAH via AMP to make a connection through our legacy PBX to the PSTN? Details: We're trying out Asterisk through Asterisk @ Home. Our legacy PBX has a modem type dial tone port that we hooked a Digium FXO to. Now I can dial from the XTEN client on my computer to any legacy PBX extension. If I connect a regular phone to the modem dial tone port, I can dial

...! interface FastEthernet0 ip address 192.168.0.3 255.255.255.0 no ip route-cache no ip mroute-cache duplex full speed auto ! ip classless ip route 0.0.0.0 0.0.0.0 192.168.0.1 no ip http server ! ! ! call rsvp-sync ! voice-port 0:0 echo-cancel coverage 16 compand-type a-law timeouts interdigit 2 ! voice-port 3:D compand-type a-law ! voice-port 1:D input gain -6 compand-type a-law cptone ZA timeouts interdigit 4 ! ! mgcp profile default ! dial-peer voice 1 pots destination-pattern .T port 0:0 forward-digits all ! gateway resource threshold high 100 low 95 ! sip-ua sip-...

Hi jpm miao, You can get CSV files that can be imported into Excel like this: library(prettyR) sink("excel_table1.csv") delim.table(table(df[,c("y","z")])) sink() sink("excel_table2.csv") delim.table(as.data.frame(table(df[,c("y","z")])),label="") sink() sink("excel_table3.csv")

Hi, How could we print the frequency table (produced by "table") to an Excel file? Is there an easy way to do so? Thanks, Miao > df <- data.frame(x = 1:3, y = 3:1, z = letters[1:3]) > table(df[,c("y","z")]) z y a b c 1 0 0 1 2 0 1 0 3 1 0 0 > test<-table(df[,c("y","z")]) > as.data.frame(test) y z Freq 1 1 a

Hi list, What paremeter can I change to control interdigit timing? Because my PSTN provider aren't receiving all the digits I dialed on Zap/g1. My Zap/g1 are an E1 (E400P) using E&M immediate sigalling. thanks in advance Eduardo

I would like to know if it is possible to slow down the dialing process in asterisk. I have 4 of my 8 phone lines that are VoDSL. When we try and dial out these 4 VoDSL Lines, the number is often miss dialed, or incomplete. I added a wait before Asterisk tries to dial the whole number, but that has not solved my problem. If I use a regular phone and dial out these lines, they work fine. My

Hi all, I have a problem with the dialing tone in PAP2: When making a call, I can hear the calling tone 5 times and then it stops. The called party still hears the call but not the calling party. I've playing around with different parameters on the PAP2 web config with no success until now. Anyone has seen the same probelm? Thanks, Jose

Hi, Am running 1.4.18 at the moment, and am trying to implement inline blind transfer. I have : [featuremap] blindxfer => *6 ; Blind transfer in features.conf And in extensions .conf under [globals] : DYNAMIC_FEATURES=automon#blindxfr So what am I missing ?? Have read through http://www.voip-info.org/wiki/view/Asterisk+config+features.conf Thanks,