[Asterisk-Users] Surge Protector for T1/PRI ?
Just recently a client of mine took a lightning hit, which in turn blew out their Digium TE411P board. This just so happened to be their main office where their call center was located. We had a backup card on hand, but this still meant downtime for the client until we got out there to replace the card. I was thinking - what if we put a surge protector device between the PRI card and the circuit itself? That way, the client themselves could replace surge protector units (if it got hit again) and protect our expensive telco equipment from getting damaged. Has anyone else experience surges on a T1/PRI circuit? What did you do to prevent further issues? Anyone from Digium - do you see a surge protector device causing interferrence or a problem with the equipment? Example device I'm looking at: http://www.apc.com/resource/include/techspec_index.cfm?base_sku=PDIGITEL Thanks!! ___ --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] Voicemail volume adjustment
It's not in the right syntax. Debugging the console should display that. It probably comes from my original message having the 'u' in the front, sorry about that - was in a hurry typing. For #1: - usg(2)[EMAIL PROTECTED] should be: [EMAIL PROTECTED]|usg(2) For #2: - [EMAIL PROTECTED]|g(2) should be: [EMAIL PROTECTED]|usg(2) That's weird that is causes asterisk to crash for #2 - what version of Asterisk are you running? Worse case you should just get a message saying that entry 'us1006' doesn't exist. Cullin J. Wible wrote: Because: usg(2)[EMAIL PROTECTED] causes the app to exit with a non-zero status, Because: [EMAIL PROTECTED]|g(2) causes asterisk to hard crash. And trying to use g2 in either case doesn't work either. Cullin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dustin Wildes Sent: Wednesday, June 28, 2006 1:03 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Voicemail volume adjustment Why use an application like sox - when you can make the voicemail application do it natively: exten = s,1,Dial(SIP/100,10) exten = s,2,Voicemail([EMAIL PROTECTED]|ug(10)) The key is the g(10) parameter: From the 'show application voicemail': g(#) - Use the specified amount of gain when recording the voicemail message. The units are whole-number decibels (dB). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Technical Support Sent: Tuesday, June 27, 2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail volume adjustment I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD ___ --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 ___ --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 ___ --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] Voicemail volume adjustment
Okay, that would make sense if you wanted 2 different volume levels for the messages. Just typically if the email attachment has low volume, usually the message on the phone is low too. In any case - you have 2 options now for adjusting volume. :-) Aaron Daniel wrote: The other problem is that if you add the gain to the original message, it seems to me the volume on the phone will be too loud as compared to the volume of the emailed message. Just a thought. ___ --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] Voicemail volume adjustment
Why use an application like sox - when you can make the voicemail application do it natively: exten = s,1,Dial(SIP/100,10) exten = s,2,Voicemail([EMAIL PROTECTED]|ug(10)) The key is the g(10) parameter: From the 'show application voicemail': g(#) - Use the specified amount of gain when recording the voicemail message. The units are whole-number decibels (dB). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Technical Support Sent: Tuesday, June 27, 2006 3:51 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Voicemail volume adjustment I frequently find voice messages are emailed to users with insufficient volume - barely audible. I would like to have asterisk run a sox command to adjust the volume of each message before emailing (perhaps once the message has been left). Has anyone done this? Care to share the steps? Thanks, MD ___ --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] GXP-2000 and Shared Line Appearances
Daniel Salama wrote: Dustin, any updates on this? Thanks, Daniel Hey Daniel! Yes - just posted the link. I appologize for the delay. Here's the link to the forum as well, if anyone is interested. This should compile and run on Asterisk-1.2.4 and higher. http://www.vecsector.com/phonecall/valet/ Enjoy! Dustin Wildes VecSector, LLC 1.912.422.7082 x101 ___ --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] GXP-2000 and Shared Line Appearances
shadowym wrote: That feature is called Bridged (or Shared) line appearance. That is one of the things Asterisk cannot do and nobody seems very interested in making it do that because it is apparently not easy. There has been some talk about implementing it but so far there does not seem to be any progress. http://forums.digium.com/viewtopic.php?p=23974#23974 I will be posting the code later today. --Dustin Wildes ___ --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] Asterisk queue log solution?
I would really prefer to use an open-source solution, or a commercial solution that offers me the code and the flexibility to manipulate the code on a need to basis. Questions: Does a solution exist that I am overlooking that may provide the functionality I am after? Is there a developer out there who we could contract to assist in creating this application? Thanks for your time and advice. /Chris Hey Chris! I'm in the process of writing a GPL compliant Queue Analyzer into PhoneCALL - but can also run outside of it, independantly. My clients are also asking for more information detailed reports, and couldn't find an open-source solution to help fill the need. I'm waiting to get this done in about 2-3 weeks. I'd love to have more testers gain input on the effort. Give me a email and/or call if interested! I'll be coding this anyway for my clients (payment is already taken care of) - so all you would have to bring to the table is criticism! ;-) *JK* Dustin Wildes VecSector, LLC email: [EMAIL PROTECTED] 1.912.422.7082 x101 ___ --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
[Asterisk-Users] Need to Hire: PHP Programmer for PhoneCALL
Hello all! It's come time where I need to add another programmer to our team. You should have at least 3 years of work experience with PHP/MySQL. Please send me your resume and a few code samples if you can. If you can only work part-time or full-time, please include that in your response. Along with your salary requirements. You'll be working with PhoneCALL, so be sure to look over the code first before applying. http://www.vecsector.com/phonecall Thanks everyone! --- Dustin Wildes President VecSector, LLC 1.912.422.7082 x101 email: [EMAIL PROTECTED] web: http://www.vecsector.com ___ --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] asterisk management interface
start shameless plug I invite you to look at our interface PhoneCALL. I designed it from the ground up, all 100% php. If anything, just to learn how it's done. http://www.vecsector.com/phonecall end shameless plug Thanks! --Dustin moona ather wrote: As I know only php and no other langugae like perl or any other... most of the links to such applications i have seen on voip.org site made in php are removed or are inactive. Can you tell me of any such application that i can use or make my own using that made only in php and serving my pupose? thanx! From: Kerry Garrison [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] asterisk management interface Date: Sun, 7 May 2006 23:45:58 -0700 Why make a brand new? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of moona ather Sent: Sunday, May 07, 2006 11:36 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] asterisk management interface Hi, I have to make a web-based management interface of configuring asterisk i wanted to know if it is as simple as reading the .conf files and searching for the required section in the file and adding users etc. or there are other steps involved too?? As I have seen many such built codes on this site and found lots of code... kindly tell me how complex it is and how many other steps are involved in making this interface as i am new in this. Emmo. _ Don't just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/ ___ --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 ___ --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 _ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/ ___ --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 ___ --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] asterisk management interface
yet another shameless plug Sorry, but this made me think of something else. PhoneCALL is 100% capable of being rebranded and resold as your own if you get a developer license from us. Furthermore, we can also function as the programming resource for the business, so all you have to worry about is building the PBX servers, load up 'your' version of PhoneCALL (including logos, names, etc..) and be on your way. Developer licensing also comes with direct support options. Our intentions for doing this was to consolidate efforts. It takes a lot of energy, time, planning (and troubleshooting) to build an interface - and is why alot of projects on here come/go. We want to work together with other businesses so we ALL make money! Thanks, and sorry for the marketing lecture. ;-) Dustin Wildes Kerry Garrison wrote: Those are reasons for WANTING to create your own, he specifically said "I HAVE to make my own" and I wanted to know why he HAS TO create his own when there are fantastics products already available. There is a huge difference in saying "I would like to create my own" and "I have to create my own". I totally understand the 'want', I "want" something that is different and don't the way I want but I don't "need" to right now. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Senad Jordanovic Sent: Monday, May 08, 2006 2:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] asterisk management interface [EMAIL PROTECTED] wrote: http://www.freepbx.org Why would you need to create your own? Many reasons: 1. not relying on already busy open source developers 2. creating something that you can possibly offer as your own commercial offering 3. have it designed exactly they way you want it from ground up 4. have a lot fun with it (and headaches :) ) etc... It is a long road though. We started PBXware in 2003 and there are still many features we wish to implement. Senad ___ --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 ___ --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 ___ --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
[Asterisk-Users] Need to Hire PHP Programmer(s)
If anyone would be interested in a 1099, work from anywhere job with PHP/MySQL programming - please contact me. Most jobs would be centered around Asterisk and PhoneCALL GUI, so in-depth knowledge in both is desired. Send Resumes and pay requirements to: [EMAIL PROTECTED] Thanks! Dustin Wildes VecSector, LLC ___ --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] 5,000 concurrent calls system rollout question
Dinesh Nair wrote: On 02/01/06 09:29 Damon Estep said the following: Ok, now lets go for 5000 of them. 160kbps*5000=80kbps or 800mbps - full duplex. Have you ever seen a NIC or switch that can run GigE full duplex at 80% utilization and not at least start to fall apart? additionally, 5000 simultaneous SIP calls at 20ms intervals will send, 5,000 * 50 * 2 = 500,000 packets per second (full duplex). not too many boxes can handle such packet load, in spite of the relatively small packet sizes. Why not bond multiple NICs together to do a load balance output? Would provide redundancy as well. ___ --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] Re: Web interface
Tzafrir Cohen wrote: On Tue, Jan 31, 2006 at 11:24:54AM +0100, Vikram Rangnekar wrote: +++ Strain Jer [30/01/06 01:29 +]: I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks Check out www.voiceroute.net DRUID is much better than AMP or any of the other interfaces out there. Also its under active development so expect a lot from it. And unlike AMP, it is non-free. BTW: there is also DeStar: http://destar.berlios.de/ . Version 0.1.1 was recently released. Nice and clean. Generally runs its own daemon, though can run under apache. I guess while we're on the subject - check out ours: PhoneCALL(tm) at http://www.vecsector.com/phonecall I think you'll agree that it's by far the most flexible and versatile GPL GUI out there. We're almost done with the Tenant User portals as well, to make setup even easier for the junior admin for the regular user to administer their phone services. I'd love any feedback/suggestions you'd have on it! Dustin Wildes VecSector, LLC ___ --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] Re: Web interface
Steve Totaro wrote: I don't see how any of these are better than AMP or [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] . What features do any of these have that [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] doesn't? The only problem with [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] is the name. [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] may have been a better selection. Thanks, Steve Totaro Well, with PhoneCALL - it's designed to be just a GUI for Asterisk, not a self-contained installation of Asterisk. This is most beneficial for developers, integrators and engineers who want to build their own servers/applications/services and have a versatile GUI to handle the front-end. This creates opportunity for the guys who want to build on any platform that supports Apache/PHP(BSD, Mac, Windows, Linux), and still have a way of professionally configuring their box. For features: Probably the biggest is PhoneCALL is Multi-Tenant capable and multilingual capable. It has the same 'modular' design as Asterisk where you can create any number of scripts, IVR menus, phone templates, realtime status monitoring for Calls and Queues, CDR reporting, and soon billing - all builtin. Since all these components are modular - it's a snap to upload a new 'app_x.so' and create a script/macro for it, then start using it. And, we are adding a '[EMAIL PROTECTED]' installation script for next release, in the event you do want to run PhoneCALL on [EMAIL PROTECTED] Hope that helps explain a little more. Dustin ___ --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] PhoneCALL version 2.7-RC1 Released!
Doug Lytle wrote: They must have fixed it, because I just logged in. Looks nice, will have to give it a try this long holiday weekend. Doug Hey Doug - yes, it was fixed this morning - we'd purged all the old demo data forgot to re-create the demo account. We've already gotten quite a few feature requests (like real-time status events for accounts, fax monitor, an interface to the backend logging security) that we're getting ready to put in place. Just keep in mind it's an RC1, so there maybe a few remaining bugs/issues which we're hoping to gain alot of feedback in the next week or so as we prepare for a -stable release. We'd love to hear your input as you try it out! :-) Fixes are usually very quick as the codebase is rather easy to understand and follow since it's all in PHP/Smarty - all of the core DB functions should be (there are few sections that still do DB function directly) in the libs/accounts.php class. If you want to use Dreamweaver to edit the templates, we posted the SMARTY extension we use for Dreamweaver. It works with both MX 2004 that we've tried. You can find it in the '3rd party' section of the downloads. --Dustin ___ --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] PhoneCALL version 2.7-RC1 Released!
Hello Everyone! For all of you PhoneCALL users, we have a treat for you today as PhoneCALL 2.7-RC1 has been released! We've worked hard to make this release as close to as bug-free as possible, but in the event you find a bug - PLEASE report it to the bugtracker. It doesn't matter how small of a 'bug' or problem you think it is - all input helps and makes the program better for everyone. The bug tracker is at: http://bugs.vecsector.com Get your copy of PhoneCALL in the Downloads section at: http://www.vecsector.com/phonecall http://www.vecsector.com/phonecall/modules.php?name=Downloads Thanks!! Dustin Wildes INFO on 2.7-RC1 New System Features include: - -- Better script handling of Arguments -- New Queue Configuration -- New Conference(MeetMe) configuration -- Defaults configuration for SIP/IAX/Voicemail -- Easy to use Installation Wizard -- Better Multi-Tenant Support -- More Security Enhances for user groups -- New user-login methods from accounts -- DID Manager implemented -- New Provider/Trunk manager -- More Advanced configuration options for accounts -- Beginning of Wizard API -- New Context Manager (for creating custom contexts) BUG FIXES - ---0001--- Warning: closedir(): supplied argument is not a valid Directory ---0002--- some settings in configs/generalsettings.php appear to have no affect / redundant ---0003--- debug (echo) statements in systemPrefs.php ---0004--- Site Name in system prefs doesn't appear to be used anywhere ---0005--- Make the $path option a configurable option ---0007--- Phonecall reports asterisk as not running while in fact the service is running ---0008--- saveconfig does not write correctly the arguments ? ---0009--- slashes,subject and body for voicemail - general settings ---00011--- Text Message field in voicemail config screen too short ---00013--- phonecall.sql file not compatible with mysql 4 ---00014--- 2 variables in generalsettings.php that look the same ---00015--- AEL ---00020--- Default account preferences for NEW accounts ---00022--- Update script causing top bar not to display for slow WAN ---00029--- Script with Multiple arguments posts to DB with ARG# 1 off ---00030--- Macro Copy ---00031--- Adding a new Extension does not bring up a screen to fill in arguments ---00032--- Arguments are not being processed properly with dropdown accounts UPCOMING FEATURES - -- Realtime Asterisk Support -- Statistics Support -- Realtime Monitoring Status viewers -- More templates(template engine) -- More Wizards and Macro defaults -- Whatever else maybe entered into the bug tracker by the community(this means you! :) ) Thanks to everyone for their feedback, contributions support for getting us to 2.7-RC1! ___ --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] Asterisk GUI/web interfaces that don't change config files
Stay tuned for PhoneCALL's 2.7-RC1 release scheduled soon. We're adding a new Security Manager that allows you to set the levels of editing for your users/admins. Chris Bagnall wrote: Hello all, I'm trying to find an Asterisk web interface (or windows gui interface) to asterisk that won't allow users to go making changes to config files. I've trawled through the very extensive list in the wiki, but there doesn't seem to be a clear defining line between applications that are purely status viewers and ones that will allow config changes. I'm looking for the user to be able to do fairly simple things like see the last few people who called them, find out if other extensions are busy, add entries to the CLID directory and so on. Thanks in advance folks. Regards, Chris ___ --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] PhoneCALL v2.7 goes MultiLingual
Hello Everyone! PhoneCALL version 2.7 http://www.vecsector.com/phonecall is finally approaching, which will be a major improvement over the past releases thanks to everyone's input feature requests! One of the newest features to PhoneCALL is the ability for the entire interface to be translated to any language you want/need. We currently have guys working on a Spanish Russian language file. It works by auto-detecting your language settings of your browser, and by selecting your language. The language file is rather simple to edit (if you are bilingual *grin*), and I'm asking for help translating PhoneCALL to your language of choice. Here is a sample format: /// // GENERAL // ALL==All ADD==Add EDIT==Edit DEL==Delete FIELD==Field VALUE==Value NAME==Name DESCRIPTION==Description TENANT==Tenant If you are interested, please contact me I'll get you started on a language file. We'd love to get as many languages as possible! :-) Thank you for your time! Dustin Wildes VecSector, LLC [EMAIL PROTECTED] http://www.vecsector.com/phonecall ___ --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] From Database, PHP-Webinterface - TO flatfileconfiguration
Hey Arne! My project 'PhoneCALL' http://www.vecsector.com/phonecall does pretty much the same thing as you are describing - stores the configs in mysql then submits the changes to flat files reloads asterisk on completion. For me my clients - there hasn't been any noticeable difference, as we used mysql/php for CDR anyway. Also - the flat-files are loaded in memory once Asterisk starts, so it's not like it's constantly hammering the mysql database for info. Go for it! :-) --Dustin Arne Morten Johansen wrote: Hi. I've started working on a PHP-project that generates the configuration files i need based on what's in my MYSQL database. I can add, delete and edit users from the web. I can set up exactly the dialplan i need by arranging the users in a firms and groups if needed. I've also set up a java servlet so that i can get asterisk to reload by pushing a button from the web-interface. The php-scripts communicates with ip-sockets. So what's my question? I'm just wondering if this is a good idea. Any comments? I've looked into the mysql support in the addons but I find it hard to do and complicated. For me it's easier to write the config-files from a php-script. But what about performance? Any big difference here? What do you think is the pros and cons of a setup like this? Regards, Arne Morten ___ --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 ___ --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] Will a VIA Epia ME6000 with a 600MHz Eden fanless CPU be suffiecient for 8 extension system?
Angus Comber wrote: Hello I am using a VIA Epia ME6000 with a 600MHz Eden Fanless CPU. Is this likely to be enough power for a 8 extension system with 6 external pstn lines? How important is cpu? Is there some measure, eg xMHz CPU per extension or something benchmark? I have installed 512MB memory - again any benchmark for asterisk memory usage? Angus Hello Angus! We are using the MII6000 at several locations. Some with 4port FXO, others with T1. Users range from 3-15. They have been running fine, one location with only 4 users is running with 128meg ram because our 1gig chip was bad - and even they haven't had any trouble. ___ --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] Software only Asterisk PBX (commercial)
Hello Matthew! Give our product a spin! It's called PhoneCALL, and you can find it here: http://www.vecsector.com/phonecall Demo is at: http://www.vecsector.com/phonecall/demo username: demo password: demo We have a GPL version Commercial version. Very, very soon our Reseller OEM channels will be ready. Please feel free to contact me if you have any questions. Thanks!! Dustin Wildes VecSector, LLC Matthew Crocker wrote: Are there any switchvox/fonality type Asterisk based PBXs where I can buy just the software? I don't want to buy their 'bundles' that come with junky PC hardware. I just want their software/GUI to run on my hardware. Does Asterisk BE come with a GUI management console for managing phones, queues, VM and the like? -Matt -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com ___ --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 ___ --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] Phonecall or something as robust
Joshua Abbott wrote: Has anyone every heard of Phonecall? : www.vecsector.com/phonecall/ Feedback? Is there something as good as it or better ? Recommendations? I've heard of it! ;-) Currently, the biggest trouble with it is the hardware configuration. I'm working on a new Hardware Manager to help resolve these issues. We've had a few bugs that have been worked out - thanks to everyone who submitted them on http://bugs.vecsector.com I've had a few programmers offer to help write a plugin to FOP that will autogenerate the configurations (op_buttons.cfg), realtime support, meetme control a queue manager for the next version 2.7. I'm sure it's not perfect, but I think with more feedback like you are requesting, we can really iron this out to be a nice complimentary GUI for Asterisk. Feedback is most welcome, either onlist or offlist. Thanks! --Dustin Wildes ___ --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] Asterisk overheating on VIA Epia M Seriesmotherboard
Angus - I have several mini-itx systems based on the Epia MII6000 (fanless) system. They all run great, and I have no problems. I also run 'mpg123' with several mp3s. I run it in an embedded configuration (in house). However, I do remember one board that I got where the heatsink on the CPU was loose which caused the thermal compound to be detached from the CPU. I removed the heatsink and put a silver compound in the place of the other compound, and we were okay again. My systems usually run around 45C-50C under load. Angus Comber wrote: But the systems are sold in this configuration. There is a fan option. I chose the fanless option. Angus - Original Message - From: C F [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, September 06, 2005 1:28 AM Subject: Re: [Asterisk-Users] Asterisk overheating on VIA Epia M Seriesmotherboard As you suspected, the problem is the fact that you don't have a fan. Since a machine that runs just a file server does not require much CPU power, the CPU doesn't get too hot. However Asterisk does use lots of CPU, therefore the CPU is hot, and yes the problem of stopping to work is because of the CPU being overheated, you are lucky that the computer booted after that, in most cases the overheating of a CPU means that the CPU expanded too much, when you shut it down it cools off, and shrinks, which could result in cracking the CPU. You should never run a CPU without it's fan if it's meant to run with a fan. Even if running it just as a file server. The fact that you are lucky doesn't mean that you don't need a fan. On 9/5/05, Angus Comber [EMAIL PROTECTED] wrote: Hello I am running Asterisk on SUSE Linux Professional 9.3 on a VIA Epia M Series motherboard - CPU runs at 1GHz. There is no fan - just a large heatsink. Currently system is running off standard IDE hard drive - because I couldn't get astlinux to run with my Digium TDM04B card (only PCI card in system). Strangely I also have the same system also running SUSE Linux running as a file server and that does not run so hot and does not overheat? Why the difference? Just booting up both systems for 15 minutes you can tell the Asterisk box is quite a bit hotter. Also the Asterisk box overheated (well think that was the problem) and stopped operating as PBX at one stage. Anyone any experience of this sort of thing? any ideas how to fix - ideally I don't want to have to fit a fan. Is SUSE not the best distro to use for this sort of thing? Should it be something to take up with VIA? Angus __ ___ --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] PhoneCALL version 1.0 Administrative Manual - Released
Greetings Everyone! The version 1.0 of the PhoneCALL Administrative Manual has been released. It is more of an outline of the features and interface, and we'll be adding lots of more detailed information in the manual over the next few days/weeks. Of course, we'd love to get your input on the manual and areas we need to clarify or even some new sections in the manual that would help explain PhoneCALL and how it works. You can find the PDF version of the manual in the Downloads, or you can view the HTML version here: http://www.vecsector.com/phonecall/demo/manual Enjoy! Dustin Wildes VecSector, LLC ___ --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] PhoneCALL v2.6.1 - Released
Hello All! Just a notice that our PHP/Smarty-based GPL version of PhoneCALL version 2.6.1 has been released, and is the current stable release. http://www.vecsector.com/phonecall We're always looking for feedback/testers to help us enhance it and make it even easier for everyone to use. The current version is designed around the advanced Asterisk user, and we are working on a more 'restrictive' model for different types of users in the system, for example: 1) User-based logins so users can control their phone options (like DND, Call CellPhone, Text Message) or update their name, email 2) Admin-based logins that control the general 'call flow' - but not administer any of the scripts/macros and can only see the information for the tenant they are assigned. 3) Site-Admin has full access to all accounts/scripts, etc... like root account (current setup) We're taking feature requests, and all feedback is welcome. Thanks! Dustin Wildes VecSector, LLC ___ 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] PhoneCALL v2.6.1 - Released
Thanks Mark! You're right - this version is intended for the 'advanced' admin, one who is very knowledgable with Asterisk, but we are working on simplifying the interface in the next revisions that will make administration easier for most user types. Basically - think of it like this: The developer/integrator would use the 'admin' interface as-is now to configure/program the PBX. After loading the applications and setting up the accounts/extensions - they could create a 'local admin' account that would allow an office manager to add an extension, reset voicemail passwords, view reports, etc... And a user-account that would allow average-joe's (no offense to anyone named 'Joe' :) ) to easily configure their extension, review call logs - etc.. The great thing is, a system configuration can be created, exported - and ready to be loaded onto the next server. This templating can make deployment very easy and fast for Asterisk-based servers, and make life alot easier on distributors. I have the beginnings of an Administrator manual about 60% finished. It should be posted later this week or next week. --Dustin Mark Phillips wrote: I like it! Not quite as simple as AMP but it does seem to be more powerfull. Keep up the good work and write a manual! Mark Dustin Wildes wrote: Hello All! Just a notice that our PHP/Smarty-based GPL version of PhoneCALL version 2.6.1 has been released, and is the current stable release. http://www.vecsector.com/phonecall We're always looking for feedback/testers to help us enhance it and make it even easier for everyone to use. The current version is designed around the advanced Asterisk user, and we are working on a more 'restrictive' model for different types of users in the system, for example: 1) User-based logins so users can control their phone options (like DND, Call CellPhone, Text Message) or update their name, email 2) Admin-based logins that control the general 'call flow' - but not administer any of the scripts/macros and can only see the information for the tenant they are assigned. 3) Site-Admin has full access to all accounts/scripts, etc... like root account (current setup) We're taking feature requests, and all feedback is welcome. Thanks! Dustin Wildes VecSector, LLC ___ ___ 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] PhoneCALL v2.6.1 - Released
Michiel van Baak wrote: On 11:21, Tue 16 Aug 05, Dustin Wildes wrote: Thanks Mark! You're right - this version is intended for the 'advanced' admin, one who is very knowledgable with Asterisk, but we are working on simplifying the interface in the next revisions that will make administration easier for most user types. Basically - think of it like this: The developer/integrator would use the 'admin' interface as-is now to configure/program the PBX. After loading the applications and setting up the accounts/extensions - they could create a 'local admin' account that would allow an office manager to add an extension, reset voicemail passwords, view reports, etc... And a user-account that would allow average-joe's (no offense to anyone named 'Joe' :) ) to easily configure their extension, review call logs - etc.. The great thing is, a system configuration can be created, exported - and ready to be loaded onto the next server. This templating can make deployment very easy and fast for Asterisk-based servers, and make life alot easier on distributors. I have the beginnings of an Administrator manual about 60% finished. It should be posted later this week or next week. Will it be possible to allow the 'local admin' to only edit specific contexts and not all. Think of this as: 1 PBX, several companies configured on it, 'local admin's per company (context) ? That would be a great feature and convince me to stop coding what I am coding now. As of right now - not currently, but it is being worked on for the next release (v2.7). We'll love to have you on board! :-) ___ 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] SNOM Hint for MeetMe
This would be absolutely perfect! I found the app_devstate.so in the 'bristuff' package. Has anyone ported over the app_devstate.c to work with HEAD? Or do you have to use this with bristuff's patched version of asterisk? Klaus-Peter Junghanns wrote: Hi, take a look at app_devstate. It lets you control SNOM LEDs from the dialplan, e.g.: exten = 1234,hint,DS/1234 exten = 1234,1,DevState(1234,2) ; == solid , or 1234,6 for blinking exten = 1234,2,Meetme(1234) exten = 1234,3,Hangup exten = h,1,DevState(1234,0) ; LED off The confiugre one SNOM funtion key as a destination to 1234. have fun, Klaus -- Klaus-Peter Junghanns On Mon, 2005-08-08 at 21:55 -0400, Dustin Wildes wrote: Has anyone written a php/perl or a hack to the 'hint' function in Asterisk that will let you monitor a MeetMe conference? So if anyone was in a conference, I could have a button light up on my Snom 360? ___ 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] SNOM Hint for MeetMe
I had noticed the 'devicestate.c' in HEAD and was looking over both the custom-bristuff version and the HEAD to see how involved it would be. Not to be pushy or anything, but do you have an ETA of the new version? I have a client that I can get off my back if I make some of their buttons light-up! (not extensions - but settings related to astdb) *hahah* I'll be more than happy to test it out. Thanks for your help!! --Dustin Klaus-Peter Junghanns wrote: There is a bristuff for CVS HEAD (quite old though...), but a newer version is on its way. On Tue, 2005-08-09 at 08:16 -0400, Dustin Wildes wrote: This would be absolutely perfect! I found the app_devstate.so in the 'bristuff' package. Has anyone ported over the app_devstate.c to work with HEAD? Or do you have to use this with bristuff's patched version of asterisk? Klaus-Peter Junghanns wrote: Hi, take a look at app_devstate. It lets you control SNOM LEDs from the dialplan, e.g.: exten = 1234,hint,DS/1234 exten = 1234,1,DevState(1234,2) ; == solid , or 1234,6 for blinking exten = 1234,2,Meetme(1234) exten = 1234,3,Hangup exten = h,1,DevState(1234,0) ; LED off The confiugre one SNOM funtion key as a destination to 1234. have fun, Klaus -- Klaus-Peter Junghanns On Mon, 2005-08-08 at 21:55 -0400, Dustin Wildes wrote: Has anyone written a php/perl or a hack to the 'hint' function in Asterisk that will let you monitor a MeetMe conference? So if anyone was in a conference, I could have a button light up on my Snom 360? ___ 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 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 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] SNOM Hint for MeetMe
Actually, this was a bit tougher in HEAD due to the new ast_channel_tech
register methods.
I've re-written/patched the app_devstate.c file to allow the toggle of
any parameter in order to make the light go on/off with the SNOM phones.
This works as of HEAD today (08/09/2005):
Thanks for the info Klaus!
--Dustin
Klaus-Peter Junghanns wrote:
hmm..extracting it from:
http://www.junghanns.net/asterisk/downloads/bristuff-0.2.0-RC8f-CVS.tar.gz
shouldnt be rocket science. ;-)
good luck,
Klaus
On Tue, 2005-08-09 at 09:36 -0400, Dustin Wildes wrote:
I had noticed the 'devicestate.c' in HEAD and was looking over both the
custom-bristuff version and the HEAD to see how involved it would be.
Not to be pushy or anything, but do you have an ETA of the new version?
I have a client that I can get off my back if I make some of their
buttons light-up! (not extensions - but settings related to astdb) *hahah*
I'll be more than happy to test it out.
Thanks for your help!!
--Dustin
/*
* Devstate application
*
* Since we like the snom leds so much, a little app to
* light the lights on the snom on demand
*
* Copyright (C) 2005, Druid Software
*
* This program is free software, distributed under the terms of
* the GNU General Public License
*/
#include asterisk/lock.h
#include asterisk/file.h
#include asterisk/logger.h
#include asterisk/channel.h
#include asterisk/pbx.h
#include asterisk/module.h
#include asterisk/astdb.h
#include asterisk/utils.h
#include asterisk/cli.h
#include asterisk/manager.h
#include stdlib.h
#include unistd.h
#include string.h
#include stdlib.h
static int ds_devicestate(void *data);
static char *type = DS;
static char *tdesc = Application for sending device state messages;
static char *app = Devstate;
static char *synopsis = Generate a device state change event given the input parameters;
static char *descrip =
Devstate(device|state): Generate a device state change event given the input parameters. Returns 0. State values match the asterisk device states. They are 0 = unknown, 1 = not inuse, 2 = inuse, 3 = busy, 4 = invalid, 5 = unavailable, 6 = ringing\n;
static char devstate_cli_usage[] =
Usage: devstate device state\n
Generate a device state change event given the input parameters.\n Mainly used for lighting the LEDs on the snoms.\n;
static int devstate_cli(int fd, int argc, char *argv[]);
static struct ast_cli_entry cli_dev_state =
{ { devstate, NULL }, devstate_cli, Set the device state on one of the \pseudo devices\., devstate_cli_usage };
STANDARD_LOCAL_USER;
LOCAL_USER_DECL;
static struct ast_channel *ds_request(const char *type, int format, void *data, int *cause);
static const struct ast_channel_tech ds_tech = {
.type = DS,
.description = Devstate,
.requester = ds_request,
.devicestate = ds_devicestate,
};
static int devstate_cli(int fd, int argc, char *argv[])
{
char devName[128];
if (argc != 3)
return RESULT_SHOWUSAGE;
if (ast_db_put(DEVSTATES, argv[1], argv[2]))
{
ast_log(LOG_DEBUG, ast_db_put failed\n);
}
snprintf(devName, sizeof(devName), DS/%s, argv[1]);
ast_device_state_changed(devName);
return RESULT_SUCCESS;
}
static int devstate_exec(struct ast_channel *chan, void *data)
{
struct localuser *u;
char *device, *state, *info;
char devName[128];
if (!(info = ast_strdupa(data))) {
ast_log(LOG_WARNING, Unable to dupe data :(\n);
return -1;
}
LOCAL_USER_ADD(u);
device = info;
state = strchr(info, '|');
if (state) {
*state = '\0';
state++;
}
else
{
ast_log(LOG_DEBUG, No state argument supplied\n);
return -1;
}
if (ast_db_put(DEVSTATES, device, state))
{
ast_log(LOG_DEBUG, ast_db_put failed\n);
}
snprintf(devName, sizeof(devName), DS/%s, device);
ast_device_state_changed(devName);
LOCAL_USER_REMOVE(u);
return 0;
}
static int ds_devicestate(void *data)
{
char *dest = data;
char stateStr[16];
if (ast_db_get(DEVSTATES, dest, stateStr, sizeof(stateStr)))
{
ast_log(LOG_DEBUG, ds_devicestate couldnt get state in astdb\n);
return 0;
}
else
{
ast_log(LOG_DEBUG, ds_devicestate dev=%s returning state %d\n,
dest, atoi(stateStr));
return (atoi(stateStr));
}
}
static char mandescr_devstate[] =
Description: Put a value into astdb\n
Variables: \n
Family: ...\n
Key: ...\n
Value: ...\n;
static int action_devstate(struct mansession *s, struct message *m)
{
char *devstate = astman_get_header(m, Devstate);
char *value = astman_get_header(m, Value);
char *id = astman_get_header(m,ActionID);
char devName[128];
if (!strlen(devstate)) {
astman_send_error(s, m, No Devstate specified);
return 0;
}
if (!strlen(value)) {
astman_send_error(s, m, No Value specified);
return 0;
}
ast_mutex_lock(s-lock
Re: [Asterisk-Users] Snom 360 4.0 firmware issue
Colin E. McDonald wrote: The new update seems to have cured my issue with calls intersecting and Zap lines not being hung up after the user terminates the session but now I am having sound issues with all of my phones. The sounds seems to be very low on all of them and there is a definite change from the same set when it was at 3.6j. The speaker also generates what appears to be static but you can discern a scratchy sounding echo. This is also occuring on all phones after the upgrade. I have genereated a support ticket to Snom but I wanted to see if anyone on the list has run into the same behavior. Thanks Colin I have about 15 snom 360 phones loaded with 4.0 - and mine seem to be working great. I did update the memory manager as well, not sure if it helped with the issues you mention because I loaded it right after 4.0: http://snom.com/download/share/snom360-3.31-r.bin ___ 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] SNOM Hint for MeetMe
Has anyone written a php/perl or a hack to the 'hint' function in Asterisk that will let you monitor a MeetMe conference? So if anyone was in a conference, I could have a button light up on my Snom 360? ___ 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] Pass-through
Adam Vocks wrote: In an order to save money, I would like to use a PRI that we have going to one of our dial-up modem banks (We are an ISP.) During business hours these channels are idle and during our peak internet times, we are closed. Sounds too good to be true, but I thought I would throw it out there. These are modem calls that if they would call our modem bank number, they would be bridged to the outbound zap channels??? And of course, if they dial our business number we would send them to the appropriate sip channels. I didnt know if this could be done with two T1 cards and asterisk Here is a primitive sketch. If anyone has information, please share. Thank You Adam Vocks CTI I've done the exact same thing. We had a 23-channel PRI that a client was using for voice, but had a small IVR for their banking application that had direct analog lines pointed to it. I ordered an Adtran Total Access 750 and an additional T1 (T100P) card. The TA750 had 24 analog lines, with one T1 interface. The asterisk server had 2 T100Ps one card was for the PRI, the second was a cross-over to the Adtran 750. Works great, don't see why it wouldn't work for you in the same method you are talking about for a modem pool. Drawing: inline: astModem.jpg___ 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] Tools for effectively manage Asterisk
[EMAIL PROTECTED] wrote: Hallo, we have started playing with asterisk about one month ago, and we do like very much what we are experiencing. Now we would like to take some step further towards standardizing installed modules, functionalities, tools etc. The wall we are facing now is: choosing the right tool for * management. We tried AMP, very powerful but incomplete (CAPI is very important to us); it also suffers from its prerequisites: apache, mysql, php... too much things that should not go in a pbx We tried IPSwitchboard, but it seems only good as a monitor, not as a configuration tool (are we correct or are we missing something?) At this point we are thinking that we better abandon the idea of GUI tools and that we must go on the road of vi editing of .conf files. We would like to understand what other people are using for asterisk management, and to get some suggestion from the community. Any suggestion is welcome We are working on finalizing a production release of our PhoneCALL product, a GPL php/smarty configuration GUI for Asterisk: http://www.vecsector.com/phonecall I feel there is nothing wrong with having a web-based configuration utility, if set up correctly. Look at the WRT54G Linksys router, plus other countless devices that use an embedded browser for configurations. It can save a lot of time on training new employees, and syntax issues when starting out. Our goal is to have a GUI that is just as flexible as writing configurations by hand, but not having to write it by hand. ;-) PhoneCALL is not production ready yet, we are on 2.5-RC4 - but within a week or so, we plan to have a very nice/clean stable version that is production ready. We don't have CAPI support built-in yet, but open for any help anyone would like to lend. ___ 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] Cisco 7960 MWI
[EMAIL PROTECTED] wrote: I've google'd this to death, is there a simple way to make MWI work from * for my Cisco phone ??? Examples ??? ___ 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 Should be easy. Just add 'mailbox=extension' in your sip.conf under the entry. Example: [1003] type=friend username=1003 secret=mysecret nat=no host=dynamic mailbox=1003 does the MWI for Cisco phones. ___ 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] Tools for effectively manage Asterisk
Dan Perik wrote: Dustin Wildes wrote: I feel there is nothing wrong with having a web-based configuration utility, if set up correctly. Look at the WRT54G Linksys router, plus other countless devices that use an embedded browser for configurations. Just a nitpick, if I may. They have embedded http servers, not browsers. But I'm sure that's what you meant. Yes - you're right, I was in a hurry. :-) Having said that, I agree that putting streamlined apache/php on an * box isn't going to cause grief. Heck, I'm breaking lots of rules, and haven't running into problems (yet). I run _everything_ on my Athlon 3000+/1GB Gentoo machine. Apache, postfix, named, mysql, courier-imap, firebird / avg tcp server, nagios, samba, X/Gnome, and vncserver/Gnome! I even (gasp) play some games on it. I'm sure that slows down some of the server functions, but I haven't noticed any problems (yet). I'm hoping to get my own dedicated server box soon to offload all the non-client stuff, but until then, it all goes on this one machine. Yes, this is a home setup, but with ties to work functions. - Dan These are the same needs a majority of the businesses we have ran into - consolidation of services. And it's only a matter of time before more more companies will offer an all-in-one small business product to handle most/all of their business communications. So you need to have the hardware to handle the features, and well designed software to be efficient. :-) ___ 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] Tools for effectively manage Asterisk (kinda long)
We are working on finalizing a production release of our PhoneCALL
product, a GPL php/smarty configuration GUI for Asterisk:
http://www.vecsector.com/phonecall
I feel there is nothing wrong with having a web-based configuration
utility, if set up correctly. Look at the WRT54G Linksys router, plus
other countless devices that use an embedded browser for
configurations. It can save a lot of time on training new employees,
and syntax issues when starting out. Our goal is to have a GUI that
is just as flexible as writing configurations by hand, but not having
to write it by hand. ;-)
PhoneCALL is not production ready yet, we are on 2.5-RC4 - but within
a week or so, we plan to have a very nice/clean stable version that
is production ready.
We don't have CAPI support built-in yet, but open for any help anyone
would like to lend.
This looks interesting. I am curious as how your view Phonecall
compared to AMP.
Well, without trying to start a war - I'll give my view the purpose of
PhoneCALL. ;-)
AMP seems like a nice product, and looks to be a all-in-one
configuration for a SOHO-type setup.
With PhoneCALL - we are working on creating a highly flexible, scalable
interface - with nice, clean code that is written 100% in php/smarty.
This will make code management alot easier, and we've made every effort
to keep the code well designed so you can write any enhancement to the
product that you may need.
Not to say we couldn't power a SOHO office, but also adding the ability
to scale large enterprise-wide configurations as well.
We have a very nice groundwork for a macro/scripting interface, along
with a new call routing manager - that attempts to more logical with
handling a call.
Here's a quick run-through:
You create an auto attendant menu that plays a greeting file, and assign
the following digit actions:
Press 1 - Sends to extension 1021
Press 2 - Goes to menu 'Tech Support'
Press 3 - Goes to Support Queue
etc...
You create the 'Tech Support' menu that plays a greeting file, and
assigns the following digit actions:
Press 1 - Transfers to Level 1 support queue
Press 2 - Transfers to Level 2 support queue
Press 3 - Transfers to Level 3 support queue
etc...
The same principle will apply to PSTN lines:
Incoming call on line 1 - during normal hours, send to Auto
Attendant menu (see above)
Incoming call on line 1, matches caller id of '111-555-' - send
to Tech support level 3 queue
Incoming call on line 2, call marketing director extension (2020)
during normal hours - calls marketing director cellphone after hours
The same logic is applied to Extensions within the system:
First, build a script to assign to an extension:
Script: Extension with Voicemail
Commands:
exten = s,1,Dial(${ARG1},20)
exten = s,2,Goto(s-${DIALSTATUS},1)
exten = s-NOANSWER,1,Voicemail(u${ARG2})
exten = s-BUSY,1,Voicemail(b${ARG2})
exten = _s-.,1,Goto(s-NOANSWER,1)
exten = a,1,VoicemailMain(${ARG2})
Next, Create your extension - and assign a script to handle the extension:
Extension: 1000 --- This is ARG1
Script: [Extension with Voicemail]
Voicemail Box:{ARG2}
[_] Send voicemail to this extension
or
Send voicemail to: [--drop down of other extensions--]
Now, whenever someone dials '1000' - it will run the 'Extension with
Voicemail' script (really an Asterisk Macro). If you ever update this
macro, you update all extensions assigned to this macro.
Now, combine this logic with the Asterisk macro facility, and you have a
very easy - yet flexible interface.
We are also implementing an export/import function within the
scripting/menus where you can quickly export all scripts from one server
and import them in another. Also if someone writes a very complex, and
detailed script that does alot of call logic - they could export the
script, post it on the community site for you to download and import
into your system.
With the import/export functions - you could quickly deploy hundreds of
PBXs with a default configuration, potentially saving you 90% of the
work per install - and creating a consistent install.
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Re: [Asterisk-Users] Asterisk on Soekris
Chris Mason (Lists) wrote: I thought I saw a Soekris embedded in the Digium booth photos, can you run Asterisk on one of these? How? I'd be interested in it for a back pbx, given the reliability. In fact, might want to move my home pbx to this also. Chris Mason ___ 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 If I'm not mistaken, the Soekris hardware does fine for a few voice channels - but not a very high performance piece of hardware. For example, if you wanted a full solution as a VPN, Asterisk server, media streaming via ICEs, web server, email server, etc... it will start to lack in performace when compared to a VIA EDEN system which can use DDR memory and such. ___ 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] Asterisk on Soekris
Maybe my point was missed. Hardware wise - a VIA MII EDEN based board will greatly outperform a Soekris system, which is why my embedded platform is based on the VIA hardware instead of the Soekris, because I AND my customers did want an all-in-one system, and small offices tend to want an all-in-one piece of equipment. Kristian Kielhofner wrote: Dustin Wildes wrote: Chris Mason (Lists) wrote: I thought I saw a Soekris embedded in the Digium booth photos, can you run Asterisk on one of these? How? I'd be interested in it for a back pbx, given the reliability. In fact, might want to move my home pbx to this also. Chris Mason ___ 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 If I'm not mistaken, the Soekris hardware does fine for a few voice channels - but not a very high performance piece of hardware. For example, if you wanted a full solution as a VPN, Asterisk server, media streaming via ICEs, web server, email server, etc... it will start to lack in performace when compared to a VIA EDEN system which can use DDR memory and such. Of course it would start to lack in performance! You'd have to be CRAZY to run all of that on a fanless $220 SBC! Like anything else, the Soekris is not an end-all, be-all solution. It does however, work surprisingly well in a lot of different applications and I am routinely impressed when I hear what people are doing with them (and AstLinux). :) ___ 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] Asterisk on Soekris
Not a problem Kristian! :-) Same here! Comments below: Kristian Kielhofner wrote: Dustin Wildes wrote: Maybe my point was missed. Hardware wise - a VIA MII EDEN based board will greatly outperform a Soekris system, which is why my embedded platform is based on the VIA hardware instead of the Soekris, because I AND my customers did want an all-in-one system, and small offices tend to want an all-in-one piece of equipment. Dustin, Yes, a VIA Eden board will greatly outperform a Net4801. My CL1 is actually quite powerful. However, several points on the mini-itx architecture that need to be mentioned: 1) Heat/Reliability. Much more heat generated, my mini-itx system has three fans. The Soekris has none (not even a heatsink). This makes the Soekris much more reliable - no moving parts. I am using the MII 6000 (no fans) with a heatpipe to replace the embedded heatsink - pushing to extruded fins. It does get warm, but not that bad. 2) Power usage. All though I have yet to measure it, my mini-itx system has a 90 watt power supply (which is ATX based, btw). My Soekris has a 12 watt power supply. Also on another note of reliability, I trust the Soekris power supply much more than the half breed ATX in most mini-itx systems. Yes, I do know that just because you have a 90 watt power supply you are not using all 90 watts, but the fact that the Soekris has a 12 watt power supply means that it is DEFINITELY not using more than 12 watts. I haven't measured the power either - but we have been using the morex power supplies for several months now, and no problems. But I not sure what the amount of wattage has to do with reliability? Personally, I'd rather have a board that could handle a bit more wattage if need be than not have enough. Would you say a 400watt power supply is less reliable than a 250watt? 3) Cases. Have you been able to find a reasonably priced case for mini-itx that doesn't look like some cheap home theater appliance? I haven't. One thing often looked for (especially in the embedded space) is for the device to look like an appliance. People are much less likely to mess with something when they don't know what it is. With a mini-itx case with upfront firewire and line-out, my 14 year old cousin would have his fingers in that case in a minute! You are right here, and they are not many good cases to choose from --- YET! :-) My company has already submitted plans to a few machineshops to build some prototype ITX cases as we speak. We just sent them in last week, so it'll be a few weeks. If anyone has an suggestions on the case style or anything they would/wouldn't like to see on a mini-ITX case, please speak now before we hit full production. We will be selling them to everyone, so if there is something you've been wanting in a mini-ITX, email me ASAP so we can look at possibly adding it to our prototype. When the 7501 comes out later this year there won't even be a point of arguing this anymore. That board is going to be killer! If the 7501 can perform to the degree we need, then you could be right. :-) Your point was not missed, but I don't think it is a good idea to include that much hodge podge functionality (web server, mail server, PBX, streaming media server, etc, etc) in one system. Also, most of my customers want reliability. Which the Soekris has over the ITX stuff, hands down. It depends on your market. Our market was for the small/home office with up to about 12 users, and they would like the biggest bang for their buck. If you could sell them one piece of hardware that could do everything they need, such as DSL PPPOE client, VPN, firewall, Intrusion Detection, web/email services, voicemail streaming to windows media/real player, plus full PBX options - it makes a nice little package. Of course, they don't have to use every feature there - they could always use a WRT54G for a DSL router/firewall, and only use our appliance for what they want/need, but at least they have the option/choice. ___ 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] Has anyone tried using a Sipura-3000 as an FXO device for *?
I found it was worse when using the G726 or G723 codecs, but if you used the G711 codec, the DTMF echo was hardly noticable. I was using the latest image: 2.0.9d -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Rich Adamson Sent: Wednesday, July 28, 2004 8:31 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Has anyone tried using a Sipura-3000 as an FXO device for *? On Tue, 2004-07-27 at 15:52, Carmi Weinzweig wrote: I am considering using Sipura-3000s as FXO devices for my * system. Has anyone tried them in that configuration? They interest me because they need no PCI slots and therefore no drivers. I would much prefer not to have any special kernel requirements for my system. A number of us are using SPA-3000s for this exact purpose, including myself. Works pretty well. -- PhoneBoy Have you found a way to get rid of the dial tone and dtmf tones when placing an outbound pstn call through the 3000? In my config, the call completes as expected however the dialtone and dtmf tones are slightly annoying. Rich ___ 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] ADSI slow?
I currently received two of the sayson 390 480 phones. I like the style of the phones, but was wanting some feedback from other users. My phones seem incredibly slow whenever connecting to voicemail. I've added the security settings to my adsi.conf file re-downloaded the script to the phone. It tells me it has a conflict with Slot1, so I'm unable to get the graphical menu - BUT, I'm able to see the caller information on each voicemail. I've used almost nothing but Cisco 7940/7960 phones with asterisk and they are very quick responsive to the voicemail, where these Sayson phones take app. 2 seconds to respond to key presses. What has your experience been if you have used one of these Sayson 390 or 480 phones? Thanks!! ___ 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] ADSI slow?
That explains a lot! Thanks! I guess the next question is - Is there anyway to speed up that 1200 b/s connection? Has anyone tried these on a Nortel or Altigen system? Are they just as slow? I requested a Programming reference guide from Sayson to explain the different KEYMODES and what options we have. If they give me something, I'll post it back to the list. That way we can work on addition Intercom/Paging, Call Hold, Parking, etc... to the adsi.conf sample config. I got these phones from the good guys at Netxusa, so they are unlocked. Just dealing with the learning curve right now! :-) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Alfred R. Nurnberger Sent: Friday, March 19, 2004 10:35 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] ADSI slow? ADSI is a slow inband protocol. You will notice that when pressing a key in voicemail that the system responds immediately but updating to a new screen takes a couple of seconds. This delay is caused by the downloading of new screen data from *. ADSI is based on the Bellcore caller id specs. So all downloading happens with a modem speed of 1200 bit/s. This is a very slow inband signalling system unlike Ciscos SIP phones where all the screen information is sent via a 10 or 100 Mbit/s ethernet connection. Your second problem is most likely caused by the passwords set in your phones. Check the archives there were a few posts how to change the passwords in your .adsi files. If you bought cheap phones of ebay you might be out of luck though - if they are provider locked then there is no way to program them without knowing the provider specific password (according to Sayson) -Alfred -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of DUSTIN WILDES Sent: Friday, March 19, 2004 6:26 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] ADSI slow? I currently received two of the sayson 390 480 phones. I like the style of the phones, but was wanting some feedback from other users. My phones seem incredibly slow whenever connecting to voicemail. I've added the security settings to my adsi.conf file re-downloaded the script to the phone. It tells me it has a conflict with Slot1, so I'm unable to get the graphical menu - BUT, I'm able to see the caller information on each voicemail. I've used almost nothing but Cisco 7940/7960 phones with asterisk and they are very quick responsive to the voicemail, where these Sayson phones take app. 2 seconds to respond to key presses. What has your experience been if you have used one of these Sayson 390 or 480 phones? Thanks!! ___ 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] Welltech FXOs
I have a 3802 and I was told by their support the SIP version doesn't support CallerID from the PSTN side. Also - mine was freezing occasionally on calls. I sent several debugs to technical support, but didn't get any response. My experience has not been that pleasant - please let me know what you find out from yours. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jorge Mendoza Sent: Tuesday, March 16, 2004 12:09 PM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Welltech FXOs Unfortunatly, no yet. The units are at customs at this moment. Keep you informed. Jorge Michael Graves wrote: Anyone make progress using the Welltech FXO adapters with *? Michael -- Michael Graves [EMAIL PROTECTED] Sr. Product Specialist www.pixelpower.com Pixel Power Inc. [EMAIL PROTECTED] ___ 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] MWI false light activity - msg0000.txt
This has been an occasional problem with us as well (around 45 users). If anyone has a fix - please share! :-) -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]On Behalf Of Darren NickersonSent: Friday, February 27, 2004 10:36 PMTo: [EMAIL PROTECTED]Subject: Re: [Asterisk-Users] MWI false light activity - msg.txt I can't offer you an explanation Rob, only thanks. We were going nuts trying to track this with SIP debugging, when in fact we had exactly the same problem on two mailboxes. In our case it was msg0015.txt causing the MWI to stay lit. -Darren -- Darren NickersonSenior Sales Support EngineeriFAX Solutions, Inc. www.ifax.com[EMAIL PROTECTED]+1.215.438.4638 ext 8106 office+1.215.243.8335 fax - Original Message - From: rjrae To: [EMAIL PROTECTED] Sent: Thursday, February 26, 2004 8:44 PM Subject: [Asterisk-Users] MWI false light activity - msg.txt Periodically when users delete voicemail a file gets left behind that triggers an inaccurate message waiting light. Users attempt to pickup/erase what they think is a legitimate message. /var/spool/asterisk/voicemail/default/*/INBOX/msg.txt Thanks for your help. Rob
RE: [Asterisk-Users] ATA-186 pass-through Flash
Cool - thanks Florian. I'll give that a try.
I guess there isn't a away to just pass the native flash via SIP yet?
-Original Message-
From: Florian Overkamp [mailto:[EMAIL PROTECTED]
Sent: Monday, January 19, 2004 2:30 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] ATA-186 pass-through Flash
Hi,
-Original Message-
How do I pass the flash button to the PBX? It seems the
ATA-186 wants to control the flash by putting my first call
on hold and prompts me to dial another extension. DTMF is
fine, just can't use the native Flash functions of our PBX
with the ATA-186 and asterisk.
You might be able to make a set of extensions that use the Flash application
on the Zap channel connected to your PBX, so you could use the flash
transfer
with asterisk and then a special dialset to indicate it must flash your
PBX first:
Exten = _*.,1,Flash(yourzapchannel)
Exten = _*.,2,Dial(yourzapchannel/${EXTEN:1})
Or something like it.
Florian
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[Asterisk-Users] ATA-186 pass-through Flash
Hello all! I have an FXO port on a cisco router that is directly connected to our PBX. Our ATA-186 (firmware version 3) registers with asterisk, which connects to our cisco router's fxo port to give me a dialtone on our PBX from the ATA. How do I pass the flash button to the PBX? It seems the ATA-186 wants to control the flash by putting my first call on hold and prompts me to dial another extension. DTMF is fine, just can't use the native Flash functions of our PBX with the ATA-186 and asterisk. Anyone done this or know which direction to go? I've tried changing the audiomode options, but nothing helped or I didn't get the right hex-to-binary conversion right. Any help would be greatly appreciated! Thanks!! ___ 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 Call Manager for Windows 0.0.1 (Alpha)
I think this is a great addition!!! Thanks for the app! -Original Message- From: Steven Sokol [mailto:[EMAIL PROTECTED] Sent: Friday, November 21, 2003 3:32 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Call Manager for Windows 0.0.1 (Alpha) If anybody is interested, I have an early version of my Call Manager for Windows application integrated with Asterisk. CMW is an application bar (like the task-bar) that docks to the top of your desktop window. It provides the following functions: 1. View Call-Related Information (Caller ID, Call State, Call Direction) 2. Monitor Status of Asterisk Stations (Channels) -- BLF or Busy Lamp Field 3. Place outbound calls. 4. Record (monitor), transfer, or drop connected, active calls. 5. Speed-dial inside and outside numbers. 6. Remote access to Asterisk CLI functions (show channels, reload, etc.) 7. AstManager COM component (can be used to add Ast functionality to other apps). Here are four screenshots with various features in use: Basic Screen: http://www.sokol-associates.com/images/AstMgr.jpg With Command Window: http://www.sokol-associates.com/images/AstMgr2.jpg With Settings Window: http://www.sokol-associates.com/images/AstMgr3.jpg With Monitor Config: http://www.sokol-associates.com/images/AstMgr4.jpg With Debug Window: http://www.sokol-associates.com/images/AstMgr5.jpg I kind of think of it as a SoftPhone-Lite application. It works as a soft-phone enhancement or add-on to your VoIP hard-phone. It is currently buggy and rather feature-poor. I hope to add lots of additional features. These will include: 1. Redial 2. Voicemail Box Monitoring 3. Enhanced Conferencing 4. Outlook/Act/Goldmine Integration (PIM stuff) 5. Call History (both inbound and outbound) 6. Redirect Option on Ring (VM, Application, Transfer, etc.) 7. Automatic mixing and delivery of monitored (recorded) files. I also plan on adding a DDE and COM extension so that it can easily act as a CTI Client to execute screen-pops. I may even throw in a scripting function that allows scripts to be executed on each incoming call. A copy of the source code (let's call this LGPL for now) is available here: http://www.sokol-associates.com/Downloads/AstMgr.zip It's written in VB6 (yes - barf, gag, whatever). The only thing required beyond the integral VB6 controls is the Windows Scripting Runtime which most PCs should have. I will work on an installable version soon. I may also port it to something more cross-platform. Please bear with me as I am just learning Gnome/GTK/X-windows. Please let me know what you think of the idea. Constructive criticism only please. I am fully aware that Windoze sucks, VB sucks, Bill Gates sucks, etc. I don't need to be called a script kiddie by anyone. (And FWIW: Asterisk will go much further by playing nicely with the evil but predominant operating system out there.) Thanks, Steve P.S. Note: the Monitor editing dialog is not working yet. I shows the devices/users but does not actually edit the file. The file (Monitor.conf) is stored in the root directory for the application and can be edited using notepad or whatever. The format is pretty self-explanatory. Obviously the PSTN and APP technologies can't be monitored. Regs, SMS ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Scope of the h extension..
Your inheritied context is including the exten = h,... for dial-out internal
because your sip.conf is pulling both via your local context.
Something like this should fix it:
[local]
include = extensions
exten = _9,,1,Goto(dial-out,${EXTEN},1)
That will only execute the exten = h,... entry for matched outgoing calls that use
9.
Hope it helps!!
-Original Message-
From: WipeOut [mailto:[EMAIL PROTECTED]
Sent: Thursday, November 20, 2003 10:26 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Scope of the h extension..
I have the following setup..
[extensions]
; all extensions defined here.
exten = 1234,
exten = 1235,
[dial-out]
; PSTN dialout config
ignorepat = 9
exten = _9,
exten = h,
[local]
; phone context in sip.conf is here..
include = extensions
include = dialout
The question is where will the h extension be active?? it appears to
run for ALL, both internal and PSTN calls, not just the calls to the
PSTN.. Is that correct?? is there any way to limit it to PSTN calls??
Later..
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RE: [Asterisk-Users] Overhead Paging
Title: Message I feel this needs to be a separate application in Asterisk, like app_sipintercom The application would connect to all available auto-answer SIP phones, play a short frequency tone for the intercom alert, only allow one-way streaming to the phones, then disconnect all phones whenever the originator hangs up. Same is true for a paging application, app_sippage The application should work the same as intercom, but allow two-way audio streaming. I was starting the design of these two applications unless anyone else has a better idea or has already begun work? Feedback welcome -Original Message-From: Jerry Gibson [mailto:[EMAIL PROTECTED]Sent: Friday, November 14, 2003 10:41 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Overhead Paging Scott: The operation you describe with multiple phones set to ring on the same extension is correct. The first one that answers, gets it. However, the meetme setup you describe also works great. I have that set up for a conference bridge where one person sets up a conference with one click on a web page which calls multiple Snom phones into a conference. This is a full conference where everyone can talk to everyone. However, The Snom phone also allows you set it upwith the mic permanently muted, which would work great forpaging. Jerry -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott (7805)Sent: Friday, November 14, 2003 9:25 AMTo: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Overhead Paging Jerry, Do you have it setup so that multiple phones answer one extension? I tried that setup with two Cisco phones, however, only the quickest responding phone answered. If you have a config that rings multiple phones and all of the phones answer the same call, I'd be interested to see the config. I guess theway to do it would be to setup a meetme conference and then dial all parties into the conference then speak -sb -Original Message-From: Jerry Gibson [mailto:[EMAIL PROTECTED]Sent: Friday, November 14, 2003 8:52 AMTo: [EMAIL PROTECTED]Subject: RE: [Asterisk-Users] Overhead Paging We do the same thing with the Snom phones. They can be set up for auto-answer, and they have a speaker jack in the back that is the same levels as a sound card on a PC. And the Snom phone automaticly hangs up when the caller hang up is detected (the SIP BYE message). Jerry -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Bisker, Scott (7805)Sent: Thursday, November 13, 2003 6:17 PMTo: '[EMAIL PROTECTED]'Subject: RE: [Asterisk-Users] Overhead Paging Our setup is to set the OSS device to autoanswer. The output of the soundcard feeds into a bank of overhead speakers. If the channel is in use, then the call gets put in a queue until the OSS device is free. -sb -Original Message-From: Johnson, Randy [mailto:[EMAIL PROTECTED]Sent: Thursday, November 13, 2003 5:34 PMTo: '[EMAIL PROTECTED]'Subject: [Asterisk-Users] Overhead Paging Does anyone have any recommendations for overhead paging systems for use with Asterisk? Thanks, Randy Johnson
[Asterisk-Users] SIP Intercom Paging (was Overhead Paging)
I wasn't thinking of using the conference system as the basis. I was thinking more along the lines of: 1) Setup a second extension on the Cisco phone named INTERCOM enabled for auto-answer 2) Create a call group on asterisk to dial that INTERCOM extension on every phone that will participate 3) Add a feature code that would dial the intercom extension and connect to all phones in the group This model could also be used for the paging feature since the INTERCOM extension has already been setup. -Original Message- From: Chris Albertson [mailto:[EMAIL PROTECTED] Sent: Friday, November 14, 2003 11:54 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Overhead Paging I'd hate to see conference bridging use for paging. A lot of wasted CPU and bandwidth. Could you multicast the UDP packets? We assume you don't need to page across multiple Asterisk servers but if you did the software wuld need to be smart enough to know which groups of extensions could be in a multicast and whci need to be bridged. Basically check to see if the SIP phone are on the same subnet. --- DUSTIN WILDES [EMAIL PROTECTED] wrote: I feel this needs to be a separate application in Asterisk, like app_sipintercom The application would connect to all available auto-answer SIP phones, play a short frequency tone for the intercom alert, only allow one-way streaming to the phones, then disconnect all phones whenever the originator hangs up. Same is true for a paging application, app_sippage The application should work the same as intercom, but allow two-way audio streaming. I was starting the design of these two applications unless anyone else has a better idea or has already begun work? Feedback welcome ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT - (Cisco 79xx) SIP ver 6.0??
Hey guys - hate to beg, but my Cisco ID has expired (yes - I'm renewing) and I can't get the latest ver 6.0 image for my SIP Phones - could anyone send me the .scp .bin? Of course this email never happened! :-) Thanks!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Answering Machine Detection
Thanks for all the info! So I take it I would need to either build an additional APP to asterisk like (voice_detection) or into an AGI and have that application or AGI run after the call is Answered? Fortunately it's not a telemarketing system! :-) It's an appointment reminder system for some of our employees. Calls them up and reminds them of important tasks like meetings and stuff. -Original Message- From: Michiel Betel [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 8:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Answering Machine Detection See http://resource.intel.com/telecom/support/documentation/unix/SR50_linux/html _files/vox_feat/contents.html#TopOfPage chapter 2 for a basic insight on Dialogic does it... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: woensdag 29 oktober 2003 3:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Answering Machine Detection Humans tend to say Hello? (short burst of audio followed by silence), and answering machines tend to say I'm sorry I'm not here right now, please leave a message after the beep (long burst of audio followed by a beep and silence). So, basically you need to decide 1) what is audio and what is background noise and 2) how long should there be audio followed by silence. On Tue, 2003-10-28 at 19:25, Alastair Maw wrote: On 27/10/03 21:57, DUSTIN WILDES wrote: Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? There's obviously no nice way of doing this. If you're doing telemarketing, and you're playing pre-recorded audio, which of course is a nasty thing to do, the algorithm is something like: 1. Dial out. 2. Wait for answer. 3. Start playing audio. 4. If you hear something that sounds like a beep, either hang up and try again later, or stop the audio, pause for two seconds and start playing it again. 5. Hang up when finished playing audio. Step 4 is accomplished by doing a FFT on the incoming audio into frequency buckets and taking a rolling average of the mean and standard deviation, such that you can detect when a fixed monotone beep occurs at the other end. If you don't want to play audio files and wait for beeps, and want to connect real humans to each other, then there's no decent way to do this, as the only difference between humans and arbitrary answering machines is that the answering machines give you a beep prompt to record your message. Regards, -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Answering Machine Detection
Because I need detection for the logging functions. Otherwise I won't get accurate logging results. -Original Message- From: Bryan Nolen [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 8:38 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Answering Machine Detection Why not just ask them to press-any-key ? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of DUSTIN WILDES Sent: Thursday, 30 October 2003 12:30 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Answering Machine Detection Thanks for all the info! So I take it I would need to either build an additional APP to asterisk like (voice_detection) or into an AGI and have that application or AGI run after the call is Answered? Fortunately it's not a telemarketing system! :-) It's an appointment reminder system for some of our employees. Calls them up and reminds them of important tasks like meetings and stuff. -Original Message- From: Michiel Betel [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 8:11 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Answering Machine Detection See http://resource.intel.com/telecom/support/documentation/unix/S R50_linux/html _files/vox_feat/contents.html#TopOfPage chapter 2 for a basic insight on Dialogic does it... -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling Sent: woensdag 29 oktober 2003 3:12 To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Answering Machine Detection Humans tend to say Hello? (short burst of audio followed by silence), and answering machines tend to say I'm sorry I'm not here right now, please leave a message after the beep (long burst of audio followed by a beep and silence). So, basically you need to decide 1) what is audio and what is background noise and 2) how long should there be audio followed by silence. On Tue, 2003-10-28 at 19:25, Alastair Maw wrote: On 27/10/03 21:57, DUSTIN WILDES wrote: Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? There's obviously no nice way of doing this. If you're doing telemarketing, and you're playing pre-recorded audio, which of course is a nasty thing to do, the algorithm is something like: 1. Dial out. 2. Wait for answer. 3. Start playing audio. 4. If you hear something that sounds like a beep, either hang up and try again later, or stop the audio, pause for two seconds and start playing it again. 5. Hang up when finished playing audio. Step 4 is accomplished by doing a FFT on the incoming audio into frequency buckets and taking a rolling average of the mean and standard deviation, such that you can detect when a fixed monotone beep occurs at the other end. If you don't want to play audio files and wait for beeps, and want to connect real humans to each other, then there's no decent way to do this, as the only difference between humans and arbitrary answering machines is that the answering machines give you a beep prompt to record your message. Regards, -- Sample configs, scripts, more : http://www.fnords.org/~eric/asterisk/ BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card
Title: FW: Voice/Data mixed routing over Digium E1/T1 Card I'm currently using this setup for a channelized T1 for voice and data. First 9 channels of the T1 are voice - the rest are data for internet. Works extremely well! This is being used for a production server that receives/places around 500 calls per day. -Original Message-From: Ray Burkholder [mailto:[EMAIL PROTECTED]Sent: Wednesday, October 29, 2003 9:01 AMTo: [EMAIL PROTECTED]Subject: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card The documentation mentions that the Digium channels can be split into some voice channels and the remainder of the channels used for routing IP traffic. Does any one have this in use in conjunction with Asterisk? Does it work well? Would you recommend it for a production server? Obviously, if this works, this makes for a cost effective platform where you obtain one E1/T1 to a provider, and they can provide TDM and data over the one circuit. No separate router required. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses dangerous content at One Unified and is believed to be clean.
RE: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card
All of the setup is running on RedHat 8.0 - no other router or CSU is needed. Don't use RedHat 9.0 yet in this setup since the ZAPTEL_NETWORK flag will not compile with the new implementation of HDLC in the kernel. I'm using the T100P for the interface. They have all Cisco SIP Phones, no everything is digital. So basically I have my T1 which plugs into my T100P - and that's it. Linux is providing the PSTN termination Internet connection along with firewall proxy. The carrier is ITC Deltacom - I'm located in the Southeastern side (South Georgia, North Florida). -Original Message- From: Scott Stingel [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 9:45 AM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card Hi Dustin- That's interesting! What is the physical setup that you have? IE: Routers, etc Also, where are you located and who is the carrier? I'm interested in setting up a similar channelized T1 here in my office (PacBell-SBC) Thanks Scott Stingel Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of DUSTIN WILDES Sent: Wednesday, October 29, 2003 2:23 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card I'm currently using this setup for a channelized T1 for voice and data. First 9 channels of the T1 are voice - the rest are data for internet. Works extremely well! This is being used for a production server that receives/places around 500 calls per day. -Original Message- From: Ray Burkholder [mailto:[EMAIL PROTECTED] Sent: Wednesday, October 29, 2003 9:01 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card The documentation mentions that the Digium channels can be split into some voice channels and the remainder of the channels used for routing IP traffic. Does any one have this in use in conjunction with Asterisk? Does it work well? Would you recommend it for a production server? Obviously, if this works, this makes for a cost effective platform where you obtain one E1/T1 to a provider, and they can provide TDM and data over the one circuit. No separate router required. Ray Burkholder [EMAIL PROTECTED] http://www.oneunified.net 704 576 5101 -- Scanned for viruses dangerous content at One Unified and is believed to be clean. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Answering Machine Detection
Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example: If * detects Answering Machine or Voicemail, hangup call the AGI will log (ANSWERING MACHINE DETECTED) and at that point, don't queue that call to re-dial later. If * detects a Human picks up the call, the AGI will log (PERSON PICK-UP) and log that entry. I'm currently running a cron job everynight that queries a database of records to call select individuals for reminders. A call.id file is generated and placed in /var/spool/asterisk/outgoing for dialing. If you need more info - please let me know, any input/suggestions welcome!!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT - SIP Auto-Answer for Cisco 7940/7960!!
I've been digging around with some cisco engineers for about a week I finally got an encouraging response to the Auto-Answer issue with the SIP Phones. Here is their reply: === == FROM CISCO == === Auto-Answer feature is introduced in SIP IP Phone 6.0 version. This software version is expected to be available for customers shortly. Please let me know if you have any questions. == == END CISCO == == Hopefully we'll finally have an intercom/paging solution with Cisco SIP Phones!!! Thought I would share the news with any/all who are interested. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] consultative transfer cisco
Yes -Original Message-From: Bartosz Jozwiak [mailto:[EMAIL PROTECTED]Sent: Thursday, October 16, 2003 2:31 PMTo: ASTERISK USERSSubject: [Asterisk-Users] consultative transfer cisco Hello, Is it possible to makeconsultative transfer on Cisco 7940 and 7960 phones? -- Bart
[Asterisk-Users] Cisco CallManager Image for 7940/7960
Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Cisco CallManager Image for 7940/7960
Ah - just got it finished!! In case anyone else has this problem, here's what I done: 1) Extract the cmterm- file from cisco in order to get the data1.cab (I used linux's cabextract) 2) Extract the two files 'P00305000200.bin' 'P0030500200.sbn' of the data1.cab file (I used i6comp for windows) 3) Put them in your /tftpboot directory 4) edit your image OS79XX.TXT file to P00305000200 Reboot your phone Presto - back to Skinny. -Original Message- From: Matthew Hardeman [mailto:[EMAIL PROTECTED] Sent: Friday, October 03, 2003 1:45 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Cisco CallManager Image for 7940/7960 Dustin, It's quite a pain to get those without a CallManager... However, there are some tools for extracting the compressed files from an InstallShield image and I have successfully done so with those files in particular and was able with some tweaking to get a phone back to Skinny without having a CallManager. Good luck. If you need a pointer or two, drop me a line at [EMAIL PROTECTED] Matt Hardeman PaperSoft -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of DUSTIN WILDES Sent: Friday, October 03, 2003 12:39 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco CallManager Image for 7940/7960 Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] OT - Headsets for Cisco 7940/7960
This is Off-Topic for Asterisk, but I wanted to get some feedback on headsets for Cisco 7940/7960 phones. We have about 10-20 people who wants/needs a headset for their phone was hoping to collect some real-world input. Thanks!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] OT - Headsets for Cisco 7940/7960
Thanks for all the great info!!! That's great about the adapter for existing headsets!! I have several that went to some Nortel phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, September 03, 2003 11:09 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] OT - Headsets for Cisco 7940/7960 There are two options that I've used: 1) Build a headset adapter so you can use a cheap computer headset. Instructions here: http://www.rvs.uni-hannover.de/people/einhorn/headset/index_e.html 2) Buy an adapter: http://shop.store.yahoo.com/founderstelecom/dirconcabfor.html and the a compatible plantronics headset. Both of these options plug into the back headset jack of the phone. The easiest, but not cheapest option is #2. The sound quality was slightly better with option #2, but this is probably cause I am not very talented with a soldering iron. - Justin On Wed, 3 Sep 2003, DUSTIN WILDES wrote: This is Off-Topic for Asterisk, but I wanted to get some feedback on headsets for Cisco 7940/7960 phones. We have about 10-20 people who wants/needs a headset for their phone was hoping to collect some real-world input. Thanks!! ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SetVar on sample.call
Ahh - I'll review over the pbx_spool.c code to see what else I can find.
I'll post any changes to the list for review.
-Original Message-
From: Richard Lyman [mailto:[EMAIL PROTECTED]
Sent: Monday, August 25, 2003 12:59 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SetVar on sample.call
as far as i know only the extension/context/priority (NOT
application/data side) has SetVar code. meaning you can't use
whats not there.
look in ..asterisk/pbx/pbx_spool.c line 189, notice that
ast_pbx_outgoing_app isn't passing 0-variable like line 192,
ast_pbx_outgoing_exten does.
(this was cvs as of last friday)
DUSTIN WILDES wrote:
Hi all!!
Does anyone have a short example or even better - a working AGI script that uses
GET VARIABLE' from a /var/spool/asterisk/outgoing call that uses SetVar?
Here's what I've tried with no luck so far:
sample.call
=
Channel: SIP/1000
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Application: Agi
Data: playTasks.agi
Callerid: Nightly Processor (999) 888-777
SetVar: taskID=300 //This ID is queried from my mysql database so the
playTasks.agi should be able to retreive this value to do another query to play
information
playTasks.agi (Derived from the agi-test.agi)
==
#!/usr/bin/perl
$|=1;
while(STDIN) {
chomp;
last unless length($_);
if (/^agi_(\w+)\:\s+(.*)$/) {
$AGI{$1} = $2;
}
}
sub checkresult {
my ($res) = @_;
my $retval;
$tests++;
chomp $res;
if ($res =~ /^200/) {
$res =~ /result=(-?\d+)/;
if (!length($1)) {
print STDERR FAIL ($res)\n;
$fail++;
} else {
print STDERR PASS ($1)\n;
$pass++;
}
} else {
print STDERR FAIL (unexpected result '$res')\n;
$fail++;
}
}
print GET VARIABLE taskID\n;
$result = STDIN;
$taskID = checkresult($result);
print STDERR TaskID: $taskID\n;
print STDERR Result: $result\n;
print SAY NUMBER $taskID \\\n;
$result = STDIN;
checkresult($result);
==
I always get 'zero' played back at the prompt the result(s) don't display my
$taskID.
Anyone got any recommendations or how to fix it?
Thanks!!
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FW: [Asterisk-Users] Sip codec preferences
Did anyone have a way to make codec negotiation work with Asterisk? This is something I would love to have working as well. I won't need PSTN - G729 mixing. Just SIP - SIP using G729 for calling remote offices via VPN, but everything else use G711. -Original Message- From: Brancaleoni Matteo [mailto:[EMAIL PROTECTED] Sent: Wednesday, July 16, 2003 11:32 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Sip codec preferences Hi. I'm experiencing a issue (not big, but important) I have an asterisk installation with a buch of sip phones analog ones. I have 2 1 sip phone that's outside in the world, and is nat'ed. I'm using g.729 with it. I wanna use g.729 only for the remote phone, and ulaw for the local ones, since they're on a lan. What happens? when I call the remote phone, g.729 is used, but when the remote calls ulaw is used... beside there's a disallow=all , allow=g729 is the user definition. so seems that when we call it, the codec definitions are taken from the user config itself, but when it call us, the codec defs are from the global settings. that's the same if we call remote (or receive) from an analog or iax phone. Here's a snippet of my sip.conf: ; ; SIP Configuration for Asterisk ; [general] port = 5060 bindaddr = 0.0.0.0 context = local tos = lowdelay disallow = all allow = ulaw ;local phone definition [200] accountcode=localphone mailbox=200 type=friend secret=secret username=200 host=dynamic callgroup=1 pickupgroup=1 ; remote phone definition [250] accountcode=remotephone type=friend secret=X nat=yes username=250 context=local reinvite=no disallow=all allow=g729 canreinvite=no host=dynamic qualify=1000 callgroup=1 pickupgroup=1 Any hint? -- Matteo Brancaleoni Powered by RedHat Linux 8.0 Linux User #153521 -BEGIN GEEK CODE BLOCK- Version: 3.12 GS d? s:- a- C+++ UL P+ L+++ E- W+++ N++ o K- w-- O- M-- V-- PS PE- Y PGP++ t 5 X+ R tv- b++ DI D+ G e h! r++ y --END GEEK CODE BLOCK-- -- Matteo Brancaleoni Espia System Administrator Email : [EMAIL PROTECTED] Web : http://www.espia.it Phone : +39.02.70633354 - ext 911 IAX(2): [EMAIL PROTECTED] - ext 911 or tel:17005662458 - ext 911 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Whoooaaa!!! Feaky - but in a good way
If this is through your Telco, they may have turned on the Callerid-Name field along with your number. I had mine turn on the Callerid-Name field for us. -Original Message- From: Andy Powell [mailto:[EMAIL PROTECTED] Sent: Sunday, June 15, 2003 3:25 PM To: [EMAIL PROTECTED] Subject:[Asterisk-Users] Whoooaaa!!! Feaky - but in a good way Ok, this has really freaked me out, but in a good way - sort of.. I've made no changes at all to my system, save messing with ADSI. However this has nothing to do with ADSI. The thing is all of a sudden my DECT phones have started reporting caller id, and not just the number, the name too! They have never done this before in the couple of months that I've had * running. I'm pleased that they have decided to work, but I am confused and concerned as to how and why it suddenly started ... anyone got any ideas? Andy ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
