similar to: Adhearsion Install Fails.

Hi, I'm giving Adhearsion a try on a Debian Squeeze. I read here (https://github.com/adhearsion/adhearsion/wiki/Getting-Started) that the command "sudo gem install adhearsion" should "automatically add the ahn command to your system". On mine I can't run ahn without specifying full path (/var/lib/gems/1.8/bin/ahn). Did I miss something ? Regards -------------- next

Is it possible to set an outboundproxy for REGISTER from Asterisk? register => xxx:yyy at sip99.foobar.com [foobar] type=peer host=sip99.foobar.com disallow=all allow=g729 canreinvite=no secret=yyy fromuser=xxx port=5099 outboundproxy=xxx.42.149.69 However, SIP REGISTER still goes directly to sip99.foobar.com, not xxx.42.149.69. Thanks, Doug.

Thanks to the hard work of many people in the Adhearsion community, I am pleased to be able to announce the immediate availability of Adhearsion version 1.0. Since Jay Phillips first began work on the project in 2006 Adhearsion has changed the way developers think about telephony applications. Now with several years of operating experience and multitudes of applications deployed to production,

So... I'm trying to access CDR(duration) and CDR(billsec) inside h... I keep getting 0. Can I access the CDR function inside a hangup extensions? Asterisk 1.4.13 Thanks, Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

The Adhearsion team announces the release of Adhearsion version 1.0.1. Adhearsion is an open source Ruby-language framework for creating telephony applications. This update primarily addresses compatibility with newer versions of other software but also adds native support for Bundler to newly created Adhearsion applications. Here are some highlights from the changelog: Handling of new Asterisk

We're doing callback here. Asterisk dials a number, waits for an answer, plays a prompt, dials a second number, and bridges the channels together. Calls are initiated from the AMI. No problems there. Easy stuff. However, I'd like to know if it's possible to have Asterisk dial the same two numbers simultaneously, play the prompt to the first one that answers, dial the second one and

2011/11/25 Olivier <oza_4h07 at yahoo.fr> > > > 2011/11/25 John Knight <john at classiccitytelco.com> > >> Was your PATH variable modified to add /var/lib/gems/1.8/bin perhaps? >> > > No I didn't. > I would have thought that rubygems installation should car of this (adding > installed gems into users paths). > As I'm new to Ruby, I

Hi All; Is there an GUI for Asterisk that can help in showing the call flow (who is in progress, who is connected, called number, ...)? I was think in AsteriskNow does this? Any advise? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

Hi All; I tried Firefly softphone with IAX and it gave very poor quality. Any one advise a good strong softphone that can work with IAX fine? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

Hi Jeff. I read that you use PCI Passthrough and works well, i was wonder if you can tell me how you do that ''cause i can''t find how to do it :( and i really need to. So if you can give some advice about how you do it would be really helpful. I appreciate your help Jeff. Regards Ivan ____________________________________________________________________________________

Hi All; Is there any limitation on the number of users for MeetMe Conference? In other words, how many parties can join to the room and become a member of the room? Is there any limitation? Regards Bilal ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now.

I have a game that is not working under wine (it just seem to hang up with a black screen, though when I try to close the program, confirmation dialog for quitting appears). Output solely says it's executing an interrupt with unimplemented parameters in wine: err:int21:INT21_Ioctl_Block int21: unknown/not implemented parameters: int21: AX 440d, BX 001a, CX 0848, DX fbac, SI fc10, DI fbec, DS

Hi All; I need to buy one IAXy device, but I discovered that it supports only g711 and ADPCM codec, so I was wonder that it does not support g729 or GSM?! Anyway, what is that ADPCM and how much it consumes bandwitdh? Also, asterisk support such codec? What its name in the configuration? Any advise? Regards Bilal

Hi, I'm doing an upgrade to the latest samba 3.0.28, and something's bugging me: on the old server I had this working OK: root preexec = /usr/local/bin/ntlogon.py --user=%U --os=%m --dir=/home/samba/netlogon/ && /usr/local/bin/pdc-log.sh %U %m %a %d On 3.0.28, I can only get the first script to run, everything after && is ignored. I have tried to put just the second

Thank you all for the voicemail cards you sent. If you have the following in PDF or laying around (scan): * AT&T/Cingular flow voicemail card * Verizon flow voicemail card * Sprint flow voicemail card * TMobile flow voicemail card * Alltel flow voicemail card * Avaya Nortel Octel flow voicemail card * Comedian Mail (Asterisk) -- I have the flow, need a card if someone has one I will work on

Hi Jean-Marc I am using speex in my voip app (wideband mode). I have noticed that when VAD kicks in, CPU considerably increases (bitrate decreases to 4 kbps). It takes about 7 ms to encode one frame versus 0.5 ms (for speech). This happens regardless preprocessor is used. Although the issue seems to be driver dependent, it occurs on XP, Vista and Mac OS X. Besides complexity and quality, this

Are shorewall-shell and shorewall-common required at compile time even if one only wishes to use shorewall-perl (4.0.9)? ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

Hi. I am trying to pass a variable from one Asterisk PBX to another. I'm using DUNDi with IAX2. Is there a way to do it? I tried the following but it fails. On peer1: [dundi-outgoing] switch => DUNDI/priv exten => s,1,Set(CDR(userfield)=test) exten => s,2,Set(DUNDIVAR=${ARG1}#TEST) exten => s,3,NoOp(Passing ${DUNDIVAR} to DUNDi peer.) exten => s,4,Goto(${DUNDIVAR},1) On

I tried to encode two minutes from an AVI DV file 720x480: ffmpeg latest release produces a FLV 320x240 25fps 360k video 40k audio in about 45 sec. ffmpeg latest release produces an OGG 320x240 25fps 360k video 40k audio in about 1:20 sec. ffmpeg2theora the same in about 2 minutes or more. The FLV quality it's quite good, some dropped frames maybe. I was wondering to know if ffmpeg team

Here is a function I wrote. It runs no problem, but generate empty pdf files. I can't find what is the problem. create.pdf<- function(x, dir) { dir.create(dir, showWarnings = FALSE) plist<- c("a", "b" , "c", "d") for(j in plist) { filedir<- paste(dir, "/", j, ".pdf",