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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,

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

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

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, 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

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, 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

Hi, Does anybody know if RAMI (Ruby Ami) is still functional? And is this still compatible with asterisk 1.8 Best Regards, Arjan Kroon Mobillion BV

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

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

-- 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

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:

I caught Mark Spencer, Kevin Fleming, John Todd, Russell Bryant, "the other Mark" in a truly Digium moment in Rostock, Germany on their way to listen to the sea shanties. http://tr.im/rawhide - be afraid, be very afraid (Adhearsions' Jason Goecke is also in the picture somewhere) /r

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 , 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

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

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

In conjunction with Mark Spencer's visit to our Paris office, we'll be kicking off Asterisk TV (http://asterisktv.com) live during the weekls Asterisk Users Conference. I believe someone from Lumenvox will be back with us on the conference, now that I've had a chance to play with their speech recognition product. I think we're starting to get some great info from the user

You can listen or join the Asterisk Users Conference Fridays at 12:30 PM EDT Today's subject suggestions: FAX capabilities, what's your solution? Multiple asterisk server implimentation: ENUM, DUNDI or even two servers connected Your subjects? Share your ideas, ask your questions! See http://x2z.eu for instructions on how to join or listen

Hi! I'm trying to set up a SIP trunk so that I can test calls, etc., between a new Asterisk box, and an old 1.4 box. --------------------------------------------------------------------------- New box: root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf siptrunk.conf: [box1] ; All box1 extensions; see extensions.conf type=peer context=adhearsion host=172.17.0.17 ; IP