Re: [asterisk-users] Emulate and script emulation of users calling in/receiving calls, transferring calls etc
Thanks, looks like: sipp.sourceforge.net supports scripting... Are there sample University Curricula for teaching VOIP with Asterisk or FreeSwitch? On Sun, Oct 16, 2011 at 5:26 AM, Daniel Tryba dan...@tryba.nl wrote: On Sat, Oct 15, 2011 at 08:12:33PM +1100, Alec Taylor wrote: If asterisk or freeswitch would be taught in a classroom environment, is there someway to emulate and script emulation of users calling in/receiving calls, transferring calls etc? The Asterisk Manager Interface (AMI) and callfiles to/from Echo(), voicemail or IVRs! But much easier is to have multiple (soft)phones available to 1 student. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Emulate and script emulation of users calling in/receiving calls, transferring calls etc
Good evening, If asterisk or freeswitch would be taught in a classroom environment, is there someway to emulate and script emulation of users calling in/receiving calls, transferring calls etc? The plan is to have each student setup there own Asterisk or FreeSwitch box, and measure handling efficiency, and communicate between the servers (by transferring calls from server to server). Thanks for all suggestions, Alec Taylor -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Who is the creative mind behind changing Asterisk commands at CLI?
We need explicit namespaces with asterisk CLI commands On Mon, Sep 26, 2011 at 3:22 AM, Paul Belanger pabelan...@digium.com wrote: On 11-09-25 01:01 PM, Bruce B wrote: Paul, These trolls are the people who put your kid to school and put food on your table by giving valuable input and testing the open source software. Are you sure Digium endorses this stand of yours? Does everyone at Digium think the users who gives feedback that is not exactly what you like is a troll? WOW! I thought only rogue users try to censor this list but congratulations to Digium's own employees. Антон, Thanks. I will explore the option. If you had bothered to search or even look at the CHANGES file, located in the source directory of asterisk, you would have seen the following: * Cleanup another bunch of CLI commands. Now all modules follow the same schema. (Done by lmadsen, junky and mvanbaak during the devcon 2008) Additionally, you could have taken the time to actually find the commit that made the change, since this is open source software everything is listed online [1]. Which was done by mvanbaak, an asterisk community member, not a Digium employee. [1] http://svnview.digium.com/svn/asterisk?view=revisionrevision=145121 -- Paul Belanger Digium, Inc. | Software Developer twitter: pabelanger | IRC: pabelanger (Freenode) Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk on Android?
My old phone could run Asterisk (as a PBX server). Battery died pretty quickly though... On Fri, Sep 9, 2011 at 9:03 PM, amit anand onewaytoconn...@gmail.com wrote: Hey can you share something on this On Thu, Sep 8, 2011 at 23:49, Cobra 2 cob...@linuxbasement.com wrote: I've chrooted debian onto a Motorola Droid running Cyanogenmod 7 and I've gotten asterisk to run on that just fine. On Sat, Sep 3, 2011 at 9:45 AM, Daniel Tryba dan...@tryba.nl wrote: On Sat, Sep 03, 2011 at 01:53:54PM +0200, Gilles wrote: Do you want to run the entire PBX on the Android client or are you just looking for a IAX programm to be installed for receiving calls?! The entire PBX so I can have an IVR in the phone. I don't think you can access the radio of the phone (RIL) at this moment. So if you want to use the GSM itself you are out of luck. -- Daniel Tryba -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Amit Anand +91 9818559898 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Conference calls through web-interface with moderation using Asterisk?
Good Morning, I have been researching this for a while, basically I'd like to have a website with the following functionality: • One-click call-in to show (after setting username, best-case scenario: sign-in through Drupal, use that name for conference-call) • Web-interface only (Flash/Flex, Javascript/JQuery or Java), without any additional software/addons/plugins to install • Moderation: host of conference call can quieten/mute or even kick people from the conference call if they're being rowdy So far I have setup an IceCAST server, broadcasting through edcast in an mp3 stream. Viewers of my website can now listen-in on the /radio/ sub-page. How do I setup the aforementioned [3] features using Asterisk? — Do I need [Free, Open-Source] products other than Asterisk to get this done, if so, which? Thanks for all suggestions, Alec Taylor -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Join and listen to conference call through web-interface
Thanks, looks really helpful for managing connected users (half my problem). On the web-interface question, how do I create a website with a [call-in] button? I'm using Drupal, so will make it a members only page. Basically they click the [call-in] button, and straight away they're in the conversation. It needs to grab input from mic, so I'm thinking Java or Flash. Do you know of a solution which implements this? Thanks for all suggestions, Alec Taylor On Tue, May 3, 2011 at 4:13 PM, Andraž atle...@gmail.com wrote: This will help you start: http://www.757.org/~joat/wiki/index.php?n=Main.HomebrewAsteriskConferenceManager On Sun, May 1, 2011 at 12:41 PM, Alec Taylor alec.tayl...@gmail.com wrote: Good Afternoon, I'm working on an audio conferencing web-frontend. It'd be helpful if I could know: • Who's connected to the conference • Number of people listening to the stream I also need to be able to manage/screen/kick participants. One way I can think of is having acting as proxy between conference call and guest, and if I approve them, connect them through to the conference call. Features required: • Web frontend to listen to live audio stream of conference call • Web frontend to join conference call (1 click call-in, grab mic input) Can I do this with asterisk? - If so, how? Otherwise, can you recommend a different FOSS project to use for this? Thanks, Alec Taylor -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Using Asterisk + other FOSS projects to facilitate a call-in Internet Radio web-frontend
Hmm... it's just that I've seen implementation of this done already, in Google Voice, BlindSide, BigBlueButton and others, however none provide a simple interface for voice-only broadcast from the browser. I'm sure there's a way to do it using Asterisk, I just don't know of it! Please suggest ways of hosting mic-stream-input using a ready-made project for my website. Thanks, Alec Taylor On Wed, May 4, 2011 at 3:20 AM, Leif Madsen leif.mad...@asteriskdocs.org wrote: On 11-04-30 03:10 PM, Alec Taylor wrote: Good Evening, I'm setting up an Internet Radio website with call-in functionality, and need to know the kinds of FOSS tools I should install to get the job done. Here's an example of what I'm looking for: http://i56.tinypic.com/aafz4k.png Call protocol: [Producer calls in] [Host calls in] [Guest calls in]-[Screened by Producer, if accepted, conferenced into host] On the website they'll need to be able to call in (mic input grabbed), and listen in (without calling in). I've been suggested many things, including Skype, IceCAST and [currently the most promising] Asterisk+Red5+Red5Phone. Are there any better ways of doing this, and if not, how do I setup asterisk for the above task? Thanks for all suggestions, While it isn't a free option, using chan_skype (license purchased from Digium) may be the easiest solution here. A lot of people know what Skype is, and may already have it installed, and instead of requiring your users to install a softphone, they could just click the Call Me link you provide. Alternatively, you could get a DID and allow people to call in the old fashioned way. You'd have to pay for the interconnectivity, but if you use a SIP provider the costs should not be unreasonable (1-2 cents per minute). You could always look into using Google Talk as well :) Leif. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Using Asterisk + other FOSS projects to facilitate a call-in Internet Radio web-frontend
Good Evening, I'm setting up an Internet Radio website with call-in functionality, and need to know the kinds of FOSS tools I should install to get the job done. Here's an example of what I'm looking for: http://i56.tinypic.com/aafz4k.png Call protocol: [Producer calls in] [Host calls in] [Guest calls in]-[Screened by Producer, if accepted, conferenced into host] On the website they'll need to be able to call in (mic input grabbed), and listen in (without calling in). I've been suggested many things, including Skype, IceCAST and [currently the most promising] Asterisk+Red5+Red5Phone. Are there any better ways of doing this, and if not, how do I setup asterisk for the above task? Thanks for all suggestions, Alec Taylor -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
