similar to: Install Adhearsion on Debian [SOLVED]

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

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,

Not strictly an Asterisk question. I've tried to install adhearsion on TWO relatively fresh CentOS 5.x systems, and I get this... [root at localhost rubygems-0.9.5]# gem install adhearsion Bulk updating Gem source index for: http://gems.rubyforge.org ERROR: While executing gem ... (Errno::ENOENT) No such file or directory - /usr/lib/ruby/gems/1.8/gems/adhearsion-0.7.7/bin/ahn The

Hi, I'm giving Queue-Tip a try, following installation instructions in http://queue-tip.rubyforge.org/install.html. My setup is : ruby 1.8.7 rubygems 1.3.7 rails 3.1.3 Adhearsion 1.2.3 I'm struck in step 7 in the above installation procedure : # rake --trace db:create (in /usr/local/src/queue-tip) rake aborted! no such file to load -- initializer

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

2010/10/24 Olivier <oza_4h07 at yahoo.fr> > > > 2010/10/14 Danny Nicholas <danny at debsinc.com> > >> ------------------------------ >> >> *From:* asterisk-users-bounces at lists.digium.com [mailto: >> asterisk-users-bounces at lists.digium.com] *On Behalf Of *Olivier >> >> *Sent:* Thursday, October 14, 2010 3:34 PM >> *To:*

2012/1/10, Olivier <oza_4h07 at yahoo.fr>: > Hi, > > 1. This patch didn't correct the issue but I'm far from certain that I > correctly applied the patch. I was right to suspect I was wrong : now, after correctly applying the DAHLIN-275 patch, it's working OK (with the EchoCan module plugged-in). Thanks for your lighting fast correction !! > 2. I took the

2012/1/11, Jos? Pablo M?ndez Soto <auxcri at gmail.com>: > Im using the one that comes with Ubuntu Server 10.10 (0.0.6~pre12-1): > > http://packages.ubuntu.com/search?keywords=libspandsp&searchon=names&suite=maverick&section=all > > And having a sweet time with T.38 gateway. Oneiric already offers latest > pre18. T.38/T.30 gatewaying can tricky enough to

2010/8/24 Olivier <oza_4h07 at yahoo.fr> > Hi, > > On lenny, when I'm adding a "blacklist hfc4s8s_l1" statement in a > /etc/modprobe.d/myfile.conf file, this change seems ineffective until I > reboot : > # dahdi_genconf -v system > Default parameters from /etc/dahdi/genconf_parameters > Empty configuration -- no spans > Generating

2013/9/19 Matthew Jordan <mjordan at digium.com> > > On Thu, Sep 19, 2013 at 9:02 AM, Olivier <oza_4h07 at yahoo.fr> wrote: > >> Hi, >> >> Asterisk 11 doc says CDR(src) value is read-only (see >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR). >> >> For various reasons, I would appreciate to change its value so that it

Hello, we are using Asterisk with Adhearsion as our application server, with another Asterisk box acting as the office PBX, where all office phones are registered. A REFER to transfer calls within the office results in the Adhearsion application call being dropped, because the leg between the PBX and the app server is terminated by the PBX following the REFER. Is there a way to configure

-- Luca Pradovera luca.pradovera at gmail.com Hello, sorry, I managed to lose the reply amidst the traffic. What we have here is our application server APP with leg A in AsyncAGI in an Adhearsion application, which after some magic dials leg B on the office PBX through a configured peer. Leg B then decides that user C knows more about the subject, and initiates a blind transfer to C?s phone

Hello, as a variation of our issues with Adhearsion calls dropping when an INVITE comes in for a bridged call, I now have a new issue to contend with. Our call is in an AsyncAGI application, and has been bridged to another channel. The provider that supplies the DID sends a polling reINVITE every 15 minutes (it's a documented Metaswitch behavior amongst others). The reINVITE is seen as a new

AUC is Friday at 12:30 PM EDT. See http://x2z.eu Hi, One of our guests this week will be Jay Phillips to tell us about Adhearsion. Haven't heard about the open-source Adhearsion? Look here: http://www.linuxjournal.com/article/9519 Be with us to ask Jay questions. If you can't be there, download the recorded version (including last week's chat with Mark Spencer) here:

Hi , Thanks danny nicholas. Finally we get the things done with following. If i specify busypatten=500,500 then asterisk does not recognize hang up signal. After removing it only all are working fine. I choosed 2nd option as per your suggestions. working chan-dahdi.conf: ==================== signalling = fxs_ks busycount = 3 busydetect = yes callprogress = yes progzone=in usecallerid=yes

Hello everyone, Recently we've been focusing quite heavily on making Adhearsion[0] faster. To do that, we needed a convenient way to test our Asterisk voice apps. The obvious tool in the Open Source world is SIPp[1]. SIPp is great! Though it's a little clumsy to use sometimes, especially if you're trying to use it to drive interactive calls like an IVR. So to make our own lives

Hi, Is there a way to execute next priority in the dialplan if you have called agi:async? I want to play warning message if adhearsion is down. Currently I wasn't able to make it work. The dialplan execution ends after the first priority. [incomming] exten => _X.,1,AGI(agi:async) exten => _X.,2,Answer exten => _X.,3,Playback(some-message) exten => _X.,4,Hangup Regards, Pavel

Hello, I've tried to use adhearsion, but i think it's used for stress testing, not for funcionality testing. Somebody knows somehow that i can test my dialplan? Best Regards, Luiz Gustavo Chiaretto -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I have tried installing app_swift on both mac os x and ubuntu now and am getting the same error. I must be missing something, as I have tried multiple versions and everytime do sudo make install i get: if ! [ -f /etc/asterisk/swift.conf ]; then \ install -m 644 swift.conf.sample /etc/asterisk/swift.conf ; \ fi if [ -f app_swift.so ]; then \ install -m 755 app_swift.so

Hi, 2013/4/19 Olivier <oza_4h07 at yahoo.fr> > Hello, > I've just realized that several phones display both caller name and number > while ringing but when on call, caller name is not displayed anymore. > Could you recommend a sip phone that still displays caller name during > phone call ? > Regards > I've been testing Aastra 6757i SIP phone and it appears