search for: adhearsion

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 multitud...

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/g...

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...

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/adhears...

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 -------------- next part -------------...

...e 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 wondered if I forgot something (before or after gem > install adhearsion). > > To be more specific, I followed instructions from > https://github.com/adhearsion/adhearsion/wiki/Getting-Started, using > System Ruby option and skipping > > gem update --system > > > Is it standard to add a path for each gem installed ? > > > Replying to...

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 /usr/lib/ruby/1.8/rubygems/custom_require.rb:31:in `gem_original_require' /usr/lib/ruby/1.8/rubygems/custom_require.rb:...

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 f...

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 behavio...

-- 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 using the desk phone that sends a REFER request. When leg B is hung up as part of the REFER going th...

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: http://www.talkshoe.com/talkshoe/web/tscmd...

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: <http://lists.digium.com/pipermail/ast...

...l 31 15:26:24] WARNING[5216]: chan_dahdi.c:7121 ss_thread: CallerID returned with error on channel 'DAHDI/1-1' -- Executing [s at from-pstn:1] AGI("DAHDI/1-1", "agi://localhost") in new stack -- AGI Script Executing Application: (BACKGROUND) Options: (/home/guest/adhearsion/songs_play/voices/welcome) -- <DAHDI/1-1> Playing '/home/guest/adhearsion/songs_play/voices/welcome' (language 'en') hope guidance from you, Ashik On Sat, Jul 31, 2010 at 1:50 PM, asteriskguru asteriskguru < beaasteriskguru at gmail.com> wrote: > hi, > &g...

...ion for parameter ?size_t? but no such parameter app_swift.c:451: error: expected ?{? at end of input make: *** [app_swift.o] Error 1 Im not sure whats going on here, i have setup asterisk and gotten it configured with the x-lite soft phone, so i know that is working. I am ultimately trying to use adhearsion to integrate with my rails app. I have also installed cepstral voices and these work in the terminal so i am confident that is also installed correctly. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachmen...

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 o...

...duct. I think we're starting to get some great info from the user community and I would like to see more service providers show themselves. I think it's great to hear what people like JerJer of Nufone to say. We've had some bleeding edge stuff on past conferences as well, with Jay from Adhearsion and I'm waiting for some video content from him as well. Digium has been great about following the evolution of these community projects and providing time for people like Russell to join the conference and take users' questions. If you have any video to contribute to this 24/7 Internet T...

...2.134.192,60,D(22622#${YOUR_PIN}#)) If you would like to talk about services or products your company provides and answer users' questions, contact me off list. Anyone is welcome to be a guest and answer users' questions. Previous guests have been Teliax, Lumenvox, Digium (duh!), Trixbox, Adhearsion Listen to the archived recordings here: http://x2z.eu/astusers.htm The Asterisk Users Conference is independently run and has nothing to do contractually or financially with Digium who owns the Asterisk trademark. -------------- next part -------------- An HTML attachment was scrubbed... URL: h...

...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 for old system disallow=all allow=g729 canreinvite=yes qualify=no Old box: root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf siptrunk.conf: [box2] ; All box2 extensions; see extensions.conf type=peer context=local_SIP host=172.17.145.145 ; IP for new...