similar to: OT: Optimum voice problems.

I have one TDM2404E digium card on asterisk box, after configuring the zaptel and zapata configuration files, I am getting these errors when reloading asterisk: ast_unregister_indication_country: Removed default indication country 'us' setup_zap: Ignoring signalling setup_zap: Ignoring answeronpolarityswitch unable to recognize channel 13-5 what is the reason for that? Thanks,

I've got a dumb Western Electric payphone and some homebuilt hardware to control the coin relay which is accessible to Asterisk through the AGI interface. I'd like to be able to set the state of the coin relay to collect at the end of a call if a called party answers. [Hey, I admit this project is being persued just for the fun of it ] Looking through the documentation, there is a way to

Hello Astrites, I was just about to send out a long email about not being able to detect hang-up with my CAC II FXO module on my PSTN POTS lines. I had tried many configurations. I had attached a toner to the line to see if I was getting this "disconnect supervision" signal after hang-up. (toner has a line powered light in the off position) It seemed that I was getting the signal,

I would like to know if any body tried to connect gsm gateway with polarity reversal to fxo module at asterisk server ,and if the polarity reversal solve the problem of the answer and hangup supervison on calls .i appreciate any help.Thanks in advance;Wassim _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's

Good day all We got a Wildcard TE110P I installed linux,zaptel,libpti and asterisk I coped over my zaptel.conf and zapata.conf from a previous E100P config But when I try to start asterisk it gives error not bying able to load zap channles: == Parsing '/etc/asterisk/zapata.conf': Found Jan 10 08:17:18 WARNING[-1084595552]: chan_zap.c:9308 setup_zap: Ignoring switchtype Jan 10 08:17:18

OK, I''ve been fritzing around with something I noticed last night, thinking that I understood what was going on, but now it''s getting confusing again. A system that has been running for a couple of months had a hole in the probe ID list near the end in the middle of the fbt probes. And then a couple of syscall probes were stuck in the hole. It looked like this: ... 40311

Hi all, I am running samba-server-3.0.6-4.1.100mdk, openldap-servers-2.1.25-6mdk, lib64sasl2-plug-digestmd5-2.1.15-10.1.100mdk. I have searched through the lists and I am wondering if I am the only one doing this kind of set-up.. Anyway question is as follows: In my ldap server I have normal posix accounts with plain text password that are sorted out by a sasl-regex in the slapd.conf and

Hi all what is the difference between show channels sip show channles i see the difference in both show channels show me 30 channels sip show channels shows me 221 channels any description on this ram -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070909/6dd07e17/attachment.htm

n asterisk 1.2 branch SVN 51363 zaptel svn 1980 libpri svn 393 addons svn 332 Asterisk is connected via tdm400p to an avaya system to reach PSTN. When a pstn phone hangs-up asterisk seems unable to detect the busy tone and i keep hearing like 20 busy tones until the zap channel get closed. I'm using loopstart to connect the fxo to the avaya. Some suggestions for busydetection? Thanks, --

Hi all, haven't seen many posts about asterisk in Singapore... Getting a server going there and was wondering if TDM400Ps will be fine in FCC mode, and if there are indications / cadence values that I should be putting on there as in other international locations. Seen an unsettling post on voip-info about Singapore issues with Call Polarity/Hangup Detection -- crossing my fingers I

I'm learning how to use tapply. Now I'm having a go at the following code in which dati contains almost 600 lines, Pot - numeric - are the capacities of power plants and SGruppo - text - the corresponding six technologies ("CCC", "CIC","TGC", "CSC","CPC", "TE"). .....................................................

Tracing Fans, I know it''s been a long time in coming but the CPU Performance Counter (CPC) provider is almost here! The code is currently in for review and a proposed architecture document is attached here for review. Any and all feedback/questions on the proposed implementation is welcome. Thanks. Jon. -------------- next part -------------- An embedded and charset-unspecified text

Hi, Finally, the long promised CPU performance counter provider has arrived into Nevada build 109! For those that need an injection of excitement at the start of the week, you can check out some usage examples on my blog: http://blogs.sun.com/jonh/entry/finally_dtrace_meets_the_cpu Also, you can find a new section for the provider in the DTrace docs:

Hello, I want to use Asterisk PBX in front of my old, legacy PBX. The legacy PBX can be outfitted with caller-ID and is already able to handle Calling Party Control Signal Detection (this is a Panasonic KX-TD1232 Super Hybrid PBX. My question is how would one enable Asterisk to control the TDM400P/FXS port to provide to the /FXO CO port on the legacy PBX, support for proper answer supervision/CPC

Hi there, I am trying to fit a generalised linear model to some loan application and default data. The purpose of this is to eventually work out the probability an applicant will default. However, R seems to crash or die when I run "glm" on anything greater than a 5-way saturated model for my data. My first question: is the best way to fit a generalised linear model in R to fit the

Dear R community I have a question regarding the value of cost complexity parameter "k" used in "tree" package for pruning purpose. Any help in finding the optimum value of "k" is requested. Please give some suggestion in this regard. In the example below i used k=0 but i don't know why? But if i use k=NULL, then it will not plot the resultant tree.

Callerid issue 1) if you run ztmonitor on the fxo line & call in do you hear the fsk tone if yes then we beleive the CAC is passing fsk 2) in chan_zap->ss_thread around line 4154 (current cvs) if you get to the callerid_feed at least once then if you get to chan_zap->ss_thread->callerid_get around line 4163 (current cvs) does this parse fail

Hello, I have installed the latest version 12 that has been released (12.1.0.rc3). I have setup default dtmf mode (rfc47..) but when I am calling to a endpoint that doesn't support it (no telephony event in the rtpmap) the asterisk responds OK in the signalling but DTMF is not working. Is it a known issue? Below you can see the output of the asterisk monitor. <--- Received SIP request

Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : "TDM400P &amp; Not Detecting Hangups: Got a TDM400P installed and having problems with Asterisk not detecting

hi, I am required to take text/doc files uploaded to a share/directory in a static ip, do some processing on this queue, convert the file format and save the results in a directory for it to be downloaded (or I could upload the same to another share/folder in another static ip). volume of files is about 1-5 files/second of about 50KB to 100KB each and the converting and delivery has to be in