Re: [asterisk-users] Preferred language for Asterisk AGIs development ?
Drop Asterisk, move to Freeswitch. Much easier to interact with external code bases, and it has more than one language interpreter built in (javascript, lua, etc.). If you're intent on staying on Asterisk, I would suggest skipping AGI, and write a client that monitors the state of asterisk via the manager interface. AGIs are messy in my opinion. There isn't really any "best" language for AGIs, as AGIs just communicate with Asterisk via a pipe. So really, the best language for AGIs are the language you like the best and/or best fits the application domain/requirements for your project. On Tue, 2009-05-05 at 11:52 +0500, Kashif Naeem wrote: > Hello, > > We are going to start development for a product based over Asterisk. > According to you, which is the preferred language for AGIs / IVRs > development in Asterisk. I got opinions that Perl is going to > be replaced by PHP for all future developments. > > > > -- > Kashif Naeem > Business Development Manager > Hadi Telecom > www.haditelecom.com > > Cell: +92 (0)345 4226006 > Office: +92 (0)42 5692766 > > Email: kas...@haditelecom.com > MSN: kashif__na...@hotmail.com > Gmail: meet.kas...@gmail.com > Skype: kashif.naeem > > 302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan. > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Kenneth Shaw ExpiTrans, Inc. 129 W. Wilson St., Suite 204 Costa Mesa, CA 92627 tel: 949.650.4600 fax: 949.642.6044 k...@expitrans.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom SoundPoint 501 + Asterisk
Guys, I recently purchased some new SoundPoint 501 phones. The new ones that I purchased had a newer firmware than the 501s we had deployed at the time. I updated the firmware on the older 501s, and now these phones have some very odd echoes with Asterisk, etc. I assume this is due to changes in the config file somehow (I love Polycom!), but was wondering if anyone else had experienced odd audio problems with the 501s after upgrading to the latest bootrom and sip images. Any info would be much appreciated! Thanks! -- Kenneth Shaw Director of Technology ExpiTrans, Inc. 129 W. Wilson St., Suite 204 Costa Mesa, CA 92627 tel: 949 278 7288 fax: 949 642 6044 [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Automon / wW options ?
On Thu, 2005-11-17 at 07:10 +1300, Matt Riddell wrote: > Kenneth Shaw wrote: > > exten => 123,1,SetVar(DYNAMIC_FEATURES=automon) ; enable One-touch > > exten => 123,2,Dial(SIP/phone100,,wW) ; wW allow one-touch recording > > Heh, I wasn't even close. > Actually, you don't have to do a SetVar to enable it, which would require it to be enabled before every single Dial statement. Instead, to enable automon globally, you can in extensions.conf put: [globals] DYNAMIC_FEATURES=>automon Which will enable it for any Dial() or Queue() call with the options w or W set. -- Kenneth Shaw Director of Technology ExpiTrans, Inc. 129 W. Wilson St., Suite 204 Costa Mesa, CA 92627 tel: 949 278 7288 fax: 866 494 5043 [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Automon / wW options ?
On Wed, 2005-11-16 at 08:50 -0800, Kenneth Shaw wrote: > On Tue, 2005-11-15 at 19:37 -0600, Kevin P. Fleming wrote: > > Kenneth Shaw wrote: > > > Currently I have automon enabled, but I have absolutely no idea how to > > > get it to work. I have the latest CVS HEAD release. > > > > This was just talked about in Mantis... set DYNAMIC_FEATURES in the > > dialplan to include the features that you want enabled. > > > > Ok, fair enough. But I have a couple more questions. What would the > Set(DYNAMIC_FEATURES= ... ) look like? > > Also, where do the files get recorded to? I've looked in Mantis, > however, I couldn't find the information that I needed. > Ok -- I worked it out. In features.conf put: [featuremap] automon => *1 ; *1 to activate One-touch record And then in your extensions.conf, before a Dial() or Queue() command, do this: exten => 123,1,SetVar(DYNAMIC_FEATURES=automon) ; enable One-touch exten => 123,2,Dial(SIP/phone100,,wW) ; wW allow one-touch recording IN and OUT audio will be split into two files, and will be available on your asterisk server in the following directory: /var/spool/asterisk/monitor And now it's cost effective to comply with wire-tapping laws! Go Asterisk! -- Kenneth Shaw Director of Technology ExpiTrans, Inc. 129 W. Wilson St., Suite 204 Costa Mesa, CA 92627 tel: 949 278 7288 fax: 866 494 5043 [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Automon / wW options ?
On Tue, 2005-11-15 at 19:37 -0600, Kevin P. Fleming wrote: > Kenneth Shaw wrote: > > Currently I have automon enabled, but I have absolutely no idea how to > > get it to work. I have the latest CVS HEAD release. > > This was just talked about in Mantis... set DYNAMIC_FEATURES in the > dialplan to include the features that you want enabled. > Ok, fair enough. But I have a couple more questions. What would the Set(DYNAMIC_FEATURES= ... ) look like? Also, where do the files get recorded to? I've looked in Mantis, however, I couldn't find the information that I needed. -- Kenneth Shaw Director of Technology ExpiTrans, Inc. 129 W. Wilson St., Suite 204 Costa Mesa, CA 92627 tel: 949 278 7288 fax: 866 494 5043 [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Automon / wW options ?
Currently I have automon enabled, but I have absolutely no idea how to get it to work. I have the latest CVS HEAD release. in features.conf: [featuremap] ;blindxfer => #1; Blind transfer ;disconnect => *0 ; Disconnect automon => *1 ; One Touch Record ;atxfer => *2 ; Attended transfer in extensions.conf: [testinterior] exten => 1000,1,Dial(SIP/192.168.2.2,,wW) include => cleanup Any help on this matter would be much appreciated. -- Kenneth Shaw Director of Technology ExpiTrans, Inc. 129 W. Wilson St., Suite 204 Costa Mesa, CA 92627 tel: 949 278 7288 fax: 866 494 5043 [EMAIL PROTECTED] ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Polycom Phones
Much Appreciated! Thanks! On Wed, 2005-08-03 at 15:15 -0600, dbruce wrote: > Information for configuring idle images on the polycom phones is now > available on the wiki at > http://www.voip-info.org/tiki-index.php?page=Polycom+Phones > > Regards, > Derek > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users -- Kenneth Shaw Director of Technology ExpiTrans, Inc. 2428 Newport Blvd #8 Costa Mesa, CA 92627 tel: 949 278 7288 fax: 866 494 5043 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Phones
Slightly off topic as this doesn't pertain directly to Asterisk, but with the Polycom 500/501 phones, does anyone know how to correctly put a custom logo for the idle screen on the device? I've read the Admin Guide through and through and the information there is not enough to implement it. Any information you may have is much appreciated. Thanks! -- Kenneth Shaw Director of Technology ExpiTrans, Inc. 2428 Newport Blvd #8 Costa Mesa, CA 92627 tel: 949 278 7288 fax: 866 494 5043 [EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Bandwidth Load Balancing / Dundi
Just wondering what other kinds of solutions people have considered/implemented for load balancing bandwidth and IAX connections over the net. Ideas? Results? Suggestions? Experience? -- Kenneth Shaw <[EMAIL PROTECTED]> ExpiTrans, Inc. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 released
I would never submit a story to Slashdot -- on principle that they're not a real news source & are more evil than Microsoft. -Ken Shaw... On Thu, 2004-09-23 at 09:28, Lex Lethol wrote: > Kenneth, > > Did you submit to slashdot and are you on Astricon?? > > Mark has just stated he will give out a price to the person who > submitted to slashdot.. My submition got rejected :( > > You beat me to the minute. ;) > > Congrats if you did! > > Lethol > > On Thu, 23 Sep 2004 07:58:31 -0700, Kenneth Shaw <[EMAIL PROTECTED]> wrote: > > To be Slashdotted within 30 minutes. > > > > -Ken Shaw... > > > > On Thu, 2004-09-23 at 07:28, Lex Lethol wrote: > > > Hi, > > > > > > Reporting from Astricon, Mark uploaded the 1.0 release while giving > > > his speech a few mintues ago.. > > > > > > Bring out the champagne :) > > > > > > Lethol > > > ___ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > To UNSUBSCRIBE or update options visit: > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.0 released
To be Slashdotted within 30 minutes. -Ken Shaw... On Thu, 2004-09-23 at 07:28, Lex Lethol wrote: > Hi, > > Reporting from Astricon, Mark uploaded the 1.0 release while giving > his speech a few mintues ago.. > > Bring out the champagne :) > > Lethol > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Using queue app with external members/destinations
What you want to do is connect the remote phone number to an internal extension. You can do this in a couple of ways, using the Manager interface and the "Connect" command. Alternatively, you can create a call file in Asterisk's call spool (usually /var/spool/asterisk or whatever) which has the same makeup as the Connect command. All you do then is specify in your extensions something like the following: [outboundqueue] exten => s,1,Answer exten => s,2,Wait(1) exten => s,3,Queue(...) All other queue stuff should work from there. -Ken Shaw... On Sun, 2004-09-19 at 14:42, Paul Crick wrote: > Hi guys > > I've got a need to do some call queueing, with the slightly unusual caveat > that the destination for the calls is not a phone or group of phones > connected to my local asterisk box, but an "external" PSTN number. > > Can I setup a queue in asterisk and make the queue "member" an external > address like SIP/[EMAIL PROTECTED] > > There will be a smaller number of PSTN lines available at the far end > destination than there are inbound calls queueing, so after X number of > calls, attempts to call that "agent" will receive a busy response back until > a call in progress is finished and a line becomes available to take the next > queued call (does that make sense?) > > It sounds simple enough, and doable too, I just wanted to check if anyone > else had experience and/or thoughts on this kind of setup? > > Thanks in advance > Paul > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as an outbound call machine?
Jim,
What you are probably looking for is a superdialer mechanism, as it is
tricky to get Asterisk to do predictive dialing.
A superdialer (if you don't know what it is) is basically a forward call
succession plan. What happens is that you connect to one phone number
after another in successive order, with your agent sitting and listening
on one end.
Accomplishing this in Asterisk is fairly easy, and I approached a
similar problem with an AGI script (written in PHP to interface with our
database).
All you would need to do is have an agent connect to a specific
extension and then launch the AGI script. Here's something off the top
of my head:
extensions.conf
exten => 5000,1,Wait(1)
exten => 5000,2,Answer
exten => 5000,3,AGI(superdialer.php,${INSERT_SOME_PARAM_HERE})
exten => 5000,4,Hangup
superdialer.php (using PHP AGI)
conlog("SuperDialing: $number");
// dial with a 30 second timeout (approx. 5 or 6 rings)
$result = $agi->agi_exec("EXEC Dial
IAX2/[EMAIL PROTECTED]/$number|30");
if ($result['code'] != 200) {
// error here
}
$result = $agi->agi_exec('channel status');
if (!is_array($result) || $result['code'] != 200) {
// asterisk terminated on us, so exit out
break;
}
}
?>
Something like the above should allow you to accomplish what you're
looking to accomplish.
-Ken Shaw...
On Sat, 2004-09-18 at 10:56, Jim Archer wrote:
> Hi All...
>
> I have a need to phone a large number of people and collect information
> from them. I know Asterisk has a nice IVR system, but can it be used to
> initiate a call to people listed in a database or text file?
>
> Don't worry, this is not an annoying marketing thing.
>
> Thanks...
>
>
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[Asterisk-Users] Predictive Dialer, Web & Inbound Phone System
Currently I have a predictive dialer that is web-enabled, as well as a superdialer mechanism. I was wondering what kind of success people have had with their dialers (lessons learned, etc.) and how I need some direction on how to seamlessly integrate an inbound system with the predictive dialer. My thoughts are that inbound phone calls will be routed to a receptionist first, who will then switchboard them (either to a sales, support or other queue). Is it possible for a predictive dialer to take in these inbound calls? Any lessons learned or any that you care to share? What other kinds of setups are possible? I'm relatively new to Asterisk development and to PBXs in general, but am an accomplished developer and can code up pretty much anything. -Ken Shaw... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Legacy Toshiba Phones
I found some postings from Google (notably from Mark Spencer) about successful integration of a legacy Toshiba Strata system and Asterisk. I am also facing that current dilemma. The general legacy solutions that I can come up with is very easy -- either making Asterisk a "proxy" (or frontdoor) to the Toshiba system, or have it operate behind the Toshiba via regular extensions. I'm looking to improve the VoiceMail system (and conferencing, etc. etc.). All in all, what I'd most prefer is completely dropping the Toshiba system and just replacing the PBX itself and keeping the current physical phones. Has anyone had any success in accomplishing this, or could refer me to better information on the web? All that I really need to figure out how to do is to get the phone specific (Toshiba DKT2020-SD and DKT2010-H phones) features such as Paging and the Digital Displays to work. -Ken Shaw... ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
