similar to: Call classification with Asterisk

Hi all, I was wondering if there is any documentation of the Asterisk C source code. I have downloaded the code, but I haven't seen any references to documentation neither in the Asterisk home page nor in Asterisk wiki. Does anyone know if this documentation is freely available? We are specially interested on the application API that would allow us to implement new applications for Asterisk.

hello all i'm interested in setting up a small pbx using asterisk and the primary goal is keeping the cost down. the general layout of the net is as folows: * 4 phone lines (2x isdn+ 2x analog) [or 2x isdn + 1x analog, as one might be put aside for a traditional phone/fax with no fancy stuff] * a server box * several client hosts (all linux with x) currently, the isdn lines are unused (to

Hi all, I am trying to setup a three party conference using the Asterisk Manager API. I am using the Redirect action over an established two party call. The procedure I am using is to try to redirect the two existing channels to a third party. I would expect this to connect both channels to the third party. However, one of the two parties gets disconnected. Is this the expected behavior? Is there

Hello, for a long time I've been having problems with analog modems and faxes communicationg over ISDN BRI lines. Now I began to suspect that this is due to * being out of sync with the PSTN. I have a quadBRI card with first two ports connected to PSTN, and defined as follows: span=1,1,0,ccs,ami span=2,2,0,ccs,ami This should mean that spans 1 and 2 are used as primary and secondary

Kelly Lee Myers: > Depending on what you are broadcasting, for 28.8 I would go 16kbps and be > really safe... well, you could consider not broadcasting at all, that would be 0kbps, and would fit my 9,6kbps gsm connection ;) scnr uno > ----- Original Message ----- > From: "John Griffiths" <john@capmon.com> > To: <icecast@xiph.org> > Cc:

ok so 24kbps for 56k modems... can i go any lower and get the 28 k modems? (still a lot of them about) or will 24 be good enough fo that? At 04:57 PM 4/4/2001 -0700, Jack Moffitt wrote: >> > >> > the bitrate on the mp3's is 128 (great for listening off the HD) >> > >> > whats a good bitrate to encode for streaming audio? >> >> I think this

Depending on what you are broadcasting, for 28.8 I would go 16kbps and be really safe... Lithium ----- Original Message ----- From: "John Griffiths" <john@capmon.com> To: <icecast@xiph.org> Cc: <icecast@xiph.org> Sent: Thursday, April 05, 2001 3:12 PM Subject: [icecast] bitrate for slow modems > ok so 24kbps for 56k modems... > > can i go any lower and get

Hi all, this is a rather "newbie-oriented" question, so please bear with me... The system running Asterisk has been provided with an AVM FRITZ!Card PnP. SuSE Linux 9.0 recognizes it right after booting the system and it seems to be configured (MSN) correctly... The hwinfo looks like this: --- pbx:/etc/asterisk # hwinfo --isapnp 11: ISA(PnP) 01.0: 10300 ISDN Adapter [Created at

Hi all, I'm currently searching for a PCI modem that will be used to receive faxes. I've tried out a few modems but they all use conexant chipsets, which need out-of-tree kernel drivers and currently doesn't work here (kernel oops when the installation script modprobes the driver). Does anyone know of a PCI modem that works out of the box with in-tree kernel drivers? Thanks,

Hello all, I'm a newbee and i'd like to test some analog modems i have before purchasing any new hardware. I'm using: - Pentium 500 MHz - Several 56K and 33.6K analog modems (internal and external). - ISDN BRI 1 port card. - A LAN i've successfully tested some SIP sofphones. - An ordinary telephon line (and an ordinary phone :) ). I only modified

At the same site, DTMF recognition is functioning badly, sometimes duplicating digits and sometimes totally missing others. We have checked already /proc/interrups, there is no interrupt being shared. Our zaptel has .. span=1,1,0,ccs,hdb3 On zapata we have ... switchtype=euroisdn signalling=pri_cpe relaxdtmf=no (yes doesn't seem to help) -----Original Message----- From:

Hi all, I've been lurking here and reading the Wiki for a month or so now, getting information on the suitability of Asterisk for my installation. I'm responsible for the technical stuff at a mostly-inbound call centre in Melbourne, Australia. Due to our rapid expansion, it's getting close to the time to put our old beloved Fujitsu 9600 PABX out to pasture. I'm evaluating

Hello there. What I'm trying to make is - have an asterisk server with sccp/mgcp/skip/h.323 support to handle calls between various company locations. Let's say the company has five different locations, internet connections in each one of them and would like to use it via asterisk to do telephone calls. Hardware-wise I have a couple of dual Opterons and some classic i386 servers

Hi, I am thinking of starting a project at Work to pilot to use of Astrisk and VOIP, and would like to tie in several projects together. Currently we are looking at purchasing an ISDN and H.323 based video conferencing system from Polycom. It suggests the use of 3-4 BRI lines (ie 6-8 B channels, but in the form of individual BRIs). We need ISDN for the video conferencing to communicate with

> > > > the bitrate on the mp3's is 128 (great for listening off the HD) > > > > whats a good bitrate to encode for streaming audio? > > I think this might be the problem 128kbps isn't going to work over a modem. > > I think 24kbps is probably the pratical limit for 56k modems. 56k modems might > be able to get a 32kbps stream but only under

Hello all, I am using the e1071 SVM with the tune options for classification, which work pretty well, given the examples of using tune.svm function for classification. But I have not found any example to tune the SVM novelty detection (one-classification) parameters (gamma, cost, nu), for example this are some of the options I have tried with no success: obj<-tune(svm, x,y, type

Hi, I'm looking for a mean to send digital data over an E1 line, just like isdn4linux or Capi via AVM's FritzCard is able to do it with BRI lines (e.g. for PPP or ISDN raw connections). I'm not looking for modulated audio data representing digital data, like fax or the analogue modems of former times. I want an interface to the ISDN raw data, with an outgoing call marked as

configuration ISDN BRI card : ISDN Olitec PCI 128 (hisax gazel) internet connection by ISDN 64kb/s dynamic IP nom de domaine registered to : dyndns.org avec ddclient to register IP par ddclient asterisk (on internet gateway) configuration pour ISDN BRI par virtual modems /dev/ttyI* (modem.conf) logical telephone SIP "SJPHONE" on 2 local stations "windows" (i don't succeed

Hi All I Have an ISDN 30 circuit passing through an asterisk box to a legacy pbx, all is working well but I have had a problem that modems do not work, I thought of turning off echo cancelation but I cann t seem to find the ial switch do do it, could someone point me in the right direction to enable /disbale ec on a zap channel per call? Thanks Robb

Hi all, I am doing binary classification and want to improve the classification results on imbalanced response data. Currently the performance is poor. Are there ways I could improve the performance? I could either try different classification methodologies, or try exploring the data more, and throwing away noisy data, and manipulate the data more before sending into the classifiers. I was