[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 replac

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

Re: [Asterisk-Users] Voicemail volume adjustment

use: 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 O

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 recor

Re: [Asterisk-Users] GXP-2000 and Shared Line Appearances

ecall/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

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

Re: [Asterisk-Users] Asterisk queue log solution?

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.co

[Asterisk-Users] Need to Hire: PHP Programmer for PhoneCALL

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 emai

Re: [Asterisk-Users] asterisk management interface

g) 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 &

Re: [Asterisk-Users] asterisk management interface

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 Thanks! --Dustin moona ather wrote: As I know only php and no other langugae like perl or any other...

[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

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,

Re: [Asterisk-Users] Re: Web interface

Steve Totaro wrote: I don't see how any of these are better than AMP or [EMAIL PROTECTED] . What features do any of these have that [EMAIL PROTECTED] doesn't? The only problem with [EMAIL PROTECTED] is the nam

Re: [Asterisk-Users] Re: Web interface

st 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

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

[Asterisk-Users] PhoneCALL version 2.7-RC1 Released!

all/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

[Asterisk-Users] PhoneCALL v2.7 goes MultiLingual

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 __

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 a

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,

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

Re: [Asterisk-Users] Software only Asterisk PBX (commercial)

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&#x

Re: [Asterisk-Users] Phonecall or something as robust

g), 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

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 wa

[Asterisk-Users] PhoneCALL version 1.0 Administrative Manual - Released

ual 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 vis

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

Re: [Asterisk-Users] PhoneCALL v2.6.1 - Released

in 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

[Asterisk-Users] PhoneCALL v2.6.1 - Released

ent 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 o

Re: [Asterisk-Users] SNOM Hint for MeetMe

s 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 '

Re: [Asterisk-Users] SNOM Hint for MeetMe

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 Jungh

Re: [Asterisk-Users] SNOM Hint for MeetMe

tion 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

[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@

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

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

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,

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

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-

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: choo

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

Re: [Asterisk-Users] Asterisk on Soekris

-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,

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

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

RE: [Asterisk-Users] ADSI slow?

ECTED] 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

[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

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 wha

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 PROTECT

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

[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 A

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 ha

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,...

[Asterisk-Users] SIP Intercom & Paging (was Overhead Paging)

me 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. --- DUS

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 w

[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 li

RE: [Asterisk-Users] FW: Voice/Data mixed routing over Digium E1/T1 Card

land 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 currentl

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

RE: [Asterisk-Users] Answering Machine Detection

ction 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: [Asteris

RE: [Asterisk-Users] Answering Machine Detection

fter 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 wrot

[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 Machin

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 make consultative transfer on Cisco 7940 and 7960 phones?

[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 sof

RE: [Asterisk-Users] Cisco CallManager Image for 7940/7960

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 PROTECT

[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).

RE: [Asterisk-Users] OT - Headsets for Cisco 7940/7960

ed 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 > col

[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 m

RE: [Asterisk-Users] SetVar on sample.call

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"? >

RE: [Asterisk-Users] Is Asterisk ready for "real" use?

I'll be happy to work on this if you have some techinical information on how the intercom/paging features are to be setup with the Cisco SIP phones. Last I had saw - they needed an auto-answer type function in order for intercom/paging to be implemented. -Original Message- From: Ray Bu

[Asterisk-Users] SetVar on sample.call

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 WaitTim

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

RE: [Asterisk-Users] What is Newt?

Newt is a separate library from asterisk that is used to easily create a ncurses based program. If you want to see some examples of how to use newt, look under the 'zaptel' source for zttool.c and/or under the asterisk/astman source for astman.c Newt pretty easy to code off of and can make a quic

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] S

RE: [Asterisk-Users] Music on hold, Call Parking, etc

Did you add the SetMusiconHold option in your context? exten => s,1,Wait(0) exten => s,2,Answer exten => s,3,SetMusiconHold,default <--- That enables Music on Hold for incoming PSTN calls. . . -Original Message- From: Derek Beaumont [mailto:[EMAIL PROTECTED] Sent: Wednesday,