similar to: Digium Analog card and Asterisk

hello list i have an IVR menu in asterisk 1.4 like below exten => 600,1,Ringing() exten => 600,n,Wait(2) exten => 600,n,Goto(home,s,1) [home] exten => s,1,SetGlobalVar(sounds_path=/var/lib/asterisk/sounds/) exten => s,n,Background(${sounds_path}music1) exten => s,n,Background(${sounds_path}music2) exten => s,n,Background(${sounds_path}music3) exten =>

Hello, I have found a "bug", I think in the way TDM400P cards handle FXO interface disconnect/re-connect problems. Normally I do keep all the wires connected from my CO / PABX quite securely, but I had a need to re-route the cable from one side of the desk to another, and I simply disconnected the RJ-45 connector and plugged it back in. THIS PROMPTLY RESULTED IN VERY VERY SCRATCHY AUDIO

Hi, I was wondering if anyone knew of an implementation of a function similar to "eigs" in Matlab (full description here: http://www.mathworks.com/access/helpdesk/help/techdoc/ref/eigs.html). This function differs from the standard "eigen" in that it computes a *few* eigenvectors for cases in which your matrix is very large and/or you don't need all the eigenvectors.

Hi, I'm having an issue with busy detection, the busy is not being detected. Asterisk: 1.6.2.13 DAHDI: 2.4.0 Chandahdi: busydetect=yes, busycount=2 Indications zone = us, with the modifications for my country for busy: 425Hz Pattern(0.2ms on, 0.2ms off, 0.2ms on, 0.6ms off) I compiled with BUSY DETECT DEBUG. I can see: [Feb 10 15:48:06] DEBUG[26968]: dsp.c:1276

Hi, just wondering if there is anyone that can help me configure my quintum box to operate with asterisk. i have tried and made numerous attemtps configuring the tenor to work with asterisk@home but have been unlucky. anyone out there has a cheat sheet to configure this device. thanks.. for some reason i cannot get it to work. your help is appreciated.

Hi, I followed the how-to from http://www.alcatelunleashed.com/viewtopic.php?f=44&t=840 All works fine except for Asterisk->Alcatel calls. Actually, calls from Asterisk to analog extensions on the Alcatel work. However, calls from Aserisk to digital extensions on the Alcatel 4400 do NOT work. I get this error in the Asterisk log: -- Executing Dial("SIP/4053-0823dd48",

Anyone have experience with T.38 passthru in Asterisk 1.4 to a MAX TNT Media Gateway? We're experiencing sporadic results... Topology is described below... Thanks in advance.. -Joe Traditional Fax <-Analog Line-> ATA <-SIP-> Ast1.4T.38Passthru <-SIP-> MAX TNT <-PRI-> PSTN -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I have a problem with dtmf recognition an analog lines connected to Sangoma A200. The digits (in most cases the first one) are doubled and so my IVR is useless. I tried to adjust the rxgain, toneduration and relxing the dtmf but nothing worked. I also noticed one thing it only happens during the background application: exten => s,1,Background(soundfile) exten => 111,1,Dial(SIP/111)

I have had to install a TDM800 in a site, as the telco has held off installing ISDN indefinitely.. It's all fine except for the fact that it takes ages to hang up the line (6 or more rings), and sometimes doesn't even bother. This is only on incoming calls - outgoing calls work perfectly. Is there any good tricks for a fast and accurate hangup detect in this situation?

Is it possible to pass the Message Waiting indicator from an analog line to a DAHDI port and into asterisk (and to an extension)? Thanks, Brian

Hello! I am new to R. I could not find a function analogous to matlab's function buffer, which is used in signal processing. Is there such a function in R? What I need to do is as follows. If I apply the function to the vector c(1:5) for example with a window length 3 and overlapping 2, I need to get a matrix like this: 1 2 3 2 3 4 3 4 5 In matlab this is achieved with the function buffer. Is

Hi, folks! Sangoma has an informal user survey up on their home page (at http://www.sangoma.com) asking if people would use their A500 card for North American BRI, if it were supported. I encourage anyone with an interest in voice BRI in North America to vote; this information will be used for deciding whether to make the investment in developing a National ISDN driver layer for it. Benefits to

Hi. Am new to this concept but have been requested to add VOIP capability to a small office phone system. They currently have 4 standard analog lines running into a PBX feeding 16 phones, with all the usual features, call transfer call hold internal calls etc. would the following seem reasonable ? asterisk server:- ( what specs ) cat5 > broadband (VOIP) 4 FXO's for incoming PSTN

Asterisk gurus, To have analog call progress (which as far as I know asterisk does not have right now) does it not come down to 4 states to detect - and I hope that some asterisk gurus can implement those 4 states easily. I see this as 4 states: Busy - a 50% duty cycle Ringing - a % on duty and MORE of a bigger % off duty cycle - or basically not BUSY and not TALK detect Talk - something

I know it is a strange arrangement but due to contracts, it is what it is, no PRI for now. I wonder if anyone on the list has run a server with both types of cards installed? Results? I have never touched a BRI except in concept and Cisco lab. Not sure what the BRI stuffed package may or may not do to anything else that might relate to zaptel or Sangoma. Thanks, Steve Totaro

For use with rgl, I'm looking for a function to draw a plane in an rgl scene that would function sort of like abline(a, b) does in base graphics, where abline(0, 1) draws a line of unit slope through the origin. Analogously, I'd like to have a plane3d function, so that plane3d(0, 1, 1) draws a plane through the origin with unit slopes in x & y and plane3d(3, 0, 0) draws a

Hi, I'm wondering how I could notify to a dumb analog phone that a voicemail message is waiting. My goal would be to change the tone that is heard just before user starts to dial. Any idea on that ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090706/6a2c88d5/attachment.htm

Hi there, I am stuck with the location issues. It would be easy if you have DID for each extension so that outgoing caller id would be DID of the respective extension and also physical address. Now if you are not able to get DID's for some reason. I am thinking of some situations and appreciate your thoughts. 1. If I have a PRI and map physical number (original numbers in hunt group not the

Hi all. I have a DTMF detection problem by my new analog card (ATCOM 2 FXO port). When i`m playing a voice with 'GET DATA' AGI command, sometimes asterisk do not receive DTMF from caller while the voice is playing. But if user waits to the end of playing voice, there is no problem. I`m using Asterisk 10.3.1, dahdi-2.6.1 on CentOS.6.4. Could you please help me? Here is my configs:

I have the following dialplan. Everything seems good except for one thing. If the background message is playing and the user hangs up and does not press a digit how do I run an agi on that event. I tried an exten => h,1,agi(smvoice,-digium_failed) but that was never called. I am using 1.4.10 thanks, Jerry --------------------------- [smvoice-analog] exten => s,1,Wait(1) exten =>