similar to: What would cause a drop between two asterisk systems?

Can anyone recommend some kind of virtual t.38 fax software? I'd like to test/debug some of the t.38 stuff, but it'd be much easier if I had a software client that could just generate the faxes from a workstation, rather than having to sit with the fax machine + t.38 ata to source faxes from. There doesn't seem to be much out there, and the stuff that's out there is kind of

Does anyone have a suggestion on how to handle this? For example, if I have a list of numbers that I want to go out a certain sip channel and another that I want to go out the dahdi device, is there a way to do this? None of the numbers will fit into a pattern, so just plain pattern matching won't do. The most straightforward way would be to just define explicit patterns. Obviously that

Because some users have requested transfer beep confirmations I've switched our phones over to using the asterisk transfer feature instead of the built in transfer functions of the phones. While testing it was working fine, but I changed something in features.conf and suddenly any time I hit transfer (*2), I can only enter one digit before asterisk immediately tries to dial that extension.

I can't seem to locate any documentation on what this does. I tested it out with a simple static conference room: exten => conference,1,MeetMe(,1aMqw) and a static room defined in meetme.conf: conf => 123456,22,1 Users can get in with either of the pins, but I don't see that it does anything - I can't access the admin menu, nor does it set the user as marked to open up the

So when someone's brute forcing your server is there a way to identify the originating IPs without using a tcpdump? When I get a failed auth on the console it shows 'account at asteriskserver' then tag=as25ca5023 (or some random string, though it's a bit odd as alwaysauthreject = yes is on in sip.conf). Anyway, the logs don't show anything more useful either. Is there

http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Answer Can someone clarify what "early media" is? I noticed that NOT answering a call before dumping them into a queue that has music on hold will not set up a leg to push music back over the calling SIP channel. Tossing an Answer command into the dialplan just before moving to the queue alleviates this (in either situation the

Greetings - if you're curious about monitoring GlusterFS performance, be sure and sign up for tomorrow's webinar. We will also post the recording online should you not be able to make it. Introducing Gluster for Geeks Technical Webinar Series In this Gluster for Geeks technical webinar, Craig Carl, Senior Systems Engineer, will explain and demonstrate how to monitor your Gluster

?What is an Inflector Window Insulator? http://www.youtube.com/watch?v=21DiKS5mt4k Energy Efficiency Done Right presents information on the In'Flector See Through Radiant Barrier Window and Skylight Insulator and the Energy Efficiency Industry. We will examine the growth of the energy efficiency, conservation, energy independence, and carbon emmission industries and explain

Hi, I'm getting the following error from an asterisk 1.6.0.9 installation: [May 20 06:07:18] ERROR[18517]: chan_dahdi.c:10515 dahdi_pri_error: Asked to delete sched id -1??? [May 20 06:07:18] ERROR[18517]: chan_dahdi.c:10515 dahdi_pri_error: No more room in scheduler This repeats a few times, then asterisk crashes. I can't seem to locate any info on this error at all. I'm using

Hi, I just deployed new Aastra 9480i phones and when I attempt enter digits on other systems, like host pin in a GoToMeeting, the servers on the other end do not get my entries. I am assuming this is a DTMF issue but do not see anything in this phones config other than turning on the display of the digits. I have the DTMF method set to "SIP INFO". I am using AsteriskNow w/FreePBX.

I wanted to call out Marco Agostini, who so kindly wrote up a post on using various tools to monitor GlusterFS performance: http://community.gluster.org/p/some-tools-to-monitor-performance/ Please review and comment as you see fit. If you find it helpful, let Marco know by clicking the "like" button. While we're on the subject of Monitoring GlusterFS, Gluster Superstar Craig

Dear colleagues, This is to inform you that I will be giving a webinar on the apcluster package on Thursday, June 13, 2013, 7:00pm CEST (10:00am PDT). The outline of the one-hour webinar is as follows: - Introduction to affinity propagation (AP) clustering - The apcluster package, its algorithms, and visualization tools - Live apcluster demonstration - Question and Answer period To register

In my old office, for conference purpose , gotomeeting was used. also for the lecture delivery, same gobomeeting was used, most the time , we need to listen voice only. also, we use to share desktop screen. But as far as I know SIP is the standard for video telephony. SIP can handle video +Audio. now, I am thinking that, I can give solution like selling a server which has Asterisk over it. AFAIK,

Tomorrow, Friday March 15 Maybe you missed Part 1 of "The Evolution of Regression Modeling from Classical Linear Regression to Modern Ensembles " webinar series, but you can still join for Parts 2, 3, & 4 > Register Now for Parts 2, 3, 4: https://www1.gotomeeting.com/register/500959705 > > Course Outline: Overcoming Linear Regression Limitations > > Regression is

Hands-on Webinar (no charge) Advances in Regression: Modern Ensemble and Data Mining Approaches **Part of the series: The Evolution of Regression from Classical Linear Regression to Modern Ensembles Register Now for Parts 3, 4: https://www1.gotomeeting.com/register/500959705 **All registrants will automatically receive access to recordings of Parts 1 & 2. Course Abstract: Overcoming Linear

I will be giving a webinar on glmnet on Friday May 3, 2013 at 10am PDT (pacific daylight time) The one-hour webinar will consist of: - Intro to lasso and elastic net regularization, and coefficient paths - Why is glmnet so efficient and flexible - New features of the latest version of glmnet - Live glmnet demonstration - Question and Answer period To sign up for the webinar, please go to

Maybe you missed Part 1 of "The Evolution of Regression Modeling from Classical Linear Regression to Modern Ensembles " webinar series, but you can still join for Parts 2, 3, & 4 Register Now for Parts 2, 3, 4: https://www1.gotomeeting.com/register/500959705 Download (optional) a free evaluation of the SPM software suite v7.0 (used in the hands-on components of the webinar). As a

Hi I am using libvirt (0.9.12) with openstack and xen. It looks like libvirt is not creating ebtables rules against arp spoofing etc. Here are my configs: VM definition: <domain type='xen'> <uuid>d49b777f-32f1-4093-ae47-a12efd0efd2c</uuid> <name>instance-00000168</name> <memory>2097152</memory> <os>

Hi! I have an T1 line for using with IVR AGI. I receive the calls in my T1 but my dialplan has an error but my extensions doesnt have the error that show me asterisk. I dont know from where asterisk take extension 8 and how is playing ss-noservice because in my dialplan is not exist. Any help or any cluees? Verbosity was 5 and is now 7 -- Starting simple switch on 'Zap/1-1' ==

Hi all, I have setup a Dell PowerEdge 2550 with a Sangoma A200 card with 2xFSO and 1XFS modules. The PowerEdge specs are 1 x P3 1133MHz, 512MB RAM. Sangoma A200 has 3 analogue PSTN lines connected. This server is based in Office 1, with 5 users all with a Linksys SPA942 VoIP Handset. There is another Office (Office 2) connected to here using VPN. There are two users in Office 2 with the