similar to: Create separate Voice Recording System..

I am looking for Video Conference Software (Open Source) , But but not for free Trial.. please give reference about it. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090427/cc9690ee/attachment.htm

Hi.. Iam new with asterisk PBX, and i have read about asterisk video call.: my question: 1. Is imposible to establish system video call (from Phone with GPRS/3G enabled to Computer Running Softphone like X-Lite) over Asterisk Gateway.. 2. If posible what requirement (Hardware and Software on my Asterisk,PC or My Phone) Thanks Joko Pitoyo -------------- next part -------------- An HTML

Greetings List, I have configured chan_mob for Nokia 7610. I can succefully dial from softphone to mobile and land line numbers, Softphone (PC) =====> Asterisk ====> FXO (Nokia 7610) ====> Destination Number When call is established I have to use Nokia 7610 for conversation. Is it possible to use softphone, dial via mobile phone and have conversation using softphone?

Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam -------------- next part

Not related to asterisk: but I figured someone here would have used them before? Has anyone tried http://www.wcell.com/ab/ZnJpZW5kLzEwNjI= yet? Looks like they have a voip app for your mobile handset sending voice calls out over your data service or wifi for 1c per minute calls both in the USA and internationally. Wondering if i should sign up with them for my trip to Australia

Hi, Is there an affordable HW solution to do a door phone on *? I do not mind using the solder iron to modify an existing door box. Thank you! Best regards HB Norway

Just built myself a little test server with an Atom 230 processor in it and am quite impressed with it so-far. Wondering is anyones used one in anger for a VoIP platform? I'm after something with a bit more oomph than the VIAs I'm currently using that I can use in a small box (mini ATX size) without going full-blown Xeons, etc. Cheers, Gordon

Hi, I'm in the process of setting up Asterisk in a SOHO environment using ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID is used for the business and the other is used for personal. The circuit already exists, but is presently being interfaced to POTS phones via a TA. This configuration is not very common in the US, but we are fortunate that our LEC offers it price

Hi all, Someone know how can I check for available members on a queue Before I queue the call, so I can do something else with it? Note that is not the case for joinempty Thanks, Gabriel Ortiz -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090718/462b725b/attachment.htm

Great for Chan_Mobile and GSM modem for SMS in Kannel or if Asterisk supports SMS over GSM modem. I know chan_mobile had SMS in the future at one point but have not revisited the project since. "America Movil's MVNO TracFone Wireless quietly unveiled a prepaid, nationwide unlimited offering for $45 per month that includes unlimited text messaging and 30 MB of data."

Hi, Quick question to the real world. Approx what specs would I need on server to handle 95 ZAP or Dahdi -> SIP gateway using G729 on the SIP to carrier side (nothing else, just media conversion)? Does the latest Asterisk/DAHDI significantly improve these numbers over say, Asterisk 1.2.X? Sure, there is plenty to read but nothing I could find quickly on my exact needs that was clear and I

Hi all, We have just set up trixbox latest with a Rhino r1t1 card, hooked up to a plain E1 PRI line. We call fine SIP to SIP, but as soon as we make a call from SIP to PSTN all sounds become unintelligible screeching or static kind of noise on both ends, when we call PSTN to SIP the PSTN side seemingly OK at least we hear no screeching sound, but the SIP side is a even worse screeching

On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen <joakimsen at gmail.com> wrote: > On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher > <tilghman at mail.jeffandtilghman.com> wrote: >> It is completely illegal in any country that recognizes patents. > > You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we

Not sure if anybody noticed, but it seems like Skype For Asterisk is out. $66 per channels, pretty pricey http://store.digium.com/productview.php?product_code=1SFA0001 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090817/cd8c6546/attachment.htm

Hello, What is the maximum number of simultaneous calls supported by asterisk. thks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090703/0794c554/attachment.htm

Dear All I installed a Digium card TE405P with zaptel and its running successfully with no alarms, but asterisk is not running . Any one have a cure or advice 03:09.0 Communication controller: Digium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02) Nov 14 07:56:58 localhost kernel: wct4xxp: Clearing yellow alarm on span 2 Nov 14 07:56:58 localhost kernel: wct4xxp: Clearing

I need your help: please help test the gender detection module at 575-613-4392. I wrote a gender detection module and thought I'd try it out. It only takes a second. I've been showing 90%+ accuracy and I want to make sure it's working correctly. Rain and significant background noise seems to throw it off, so I still have a bit of work to do. Have your friends and significant others

Hi I have problem with TE121 Digium card. I connected it to modem keymile Music 200 (provided by telco) but I can see 2 red lights on modem (both bellow words rx) and my card is red too. I tried to make experiment and made loopback (pins 1 4 , 2 5) and put it in card and card become green (I hope that is sign card is ok) but on CLI i can see following error message WARNING: We think we are cpe but

Hi, all. I've got an old Telrad PBX with an Emagen(?) voicemail box. The VM box, itself, is beginning to show its age. Big-time. We're thinking it might be time to look for a replacement. I'd love to install Asterisk with an FXO card or something, but I don't think it supports whatever protocol legacy PBX's used to speak to VM systems. If someone can tell me I'm

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the