similar to: Ribbit Demo

Interesting use of their voip api. http://www.eweek.com/c/a/Enterprise-Apps/Ribbit-Rolls-VOIP-for-Salesforc ecom/?kc=EWKNLVPU050508STR1 Cheers, Dean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080505/4589b730/attachment.htm

Strange issue.... when I record a file from a phone to the asterisk system I get a blip in the recording every 30 seconds. It's a very small blip, but it is there. It seems like it's only if I'm recording, not when I'm playing back that the issue happens. My SATA drives, ETH0, and my Sangoma card are all on seperate IRQs. Any thoughts on what might be causing this and how to

Hi, I have an IVR that sounds just fine and dandy over ZAP. However, when I dial in through an 800 number from a provider that I connect to via IAX I get this 'blip' in the sound file. At first I thought it was just packet loss, but it happens at the exact same spot every single time. There are several parts to the IVR menu and several places where this blip happens every single time

https://blip.tv/opensuse/episode-5622859 Does anyone have the right tools to extract this video? I refuse to use flash over here, and as it's not on youtube I can't use youtube-dl. Rich. -- Richard Jones, Virtualization Group, Red Hat http://people.redhat.com/~rjones virt-p2v converts physical machines to virtual machines. Boot with a live CD or over the network (PXE) and turn

On 2/20/2014 6:55 AM, Charles Lepple wrote: > On Feb 19, 2014, at 12:50 PM, Ted Mittelstaedt wrote: > >> Worse, however, is if there's a power failure right near the end of >> the 2-days-off cycle. That happened to me last week - it was a >> short duration 15 second loss - and the upscode2 driver decided it >> needed to issue a forced shutdown. >> >>

What codec should be used to connect a * box to a cisco router which has a t1 with 24 trunks coming in? My router voip dial plan looks like this: dial-peer voice 2 voip destination-pattern [1,2,,3,5,8].. session protocol sipv2 session target ipv4:10.x.x.x dtmf-relay cisco-rtp codec g711ulaw no vad ! The problem I have is when more than one call is on it, sometimes the quality gets very

Hmm beginning of the end of free trixbox by the sounds of it. It was good while it lasted but time to download the latest iso while it's still available by the sounds of it. Regards, Dean Collins Cognation Pty Ltd dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). ________________________________ From: trixbox

Anyone else around using apcupsd? I seem to be seeing a problem, and I'd like someone to check me on it: I edit /etc/apcupsd/apccontrol to replace the value of SHUTDOWN from /sbin/shutdown to /bin/false (we don't want 3 or 6 servers shutting down over a 2 second or so blip, which it really wants to do). What's happened is that a machine shut down the other day, and looking at it, I

http://bugs.digium.com/bug_view_page.php?bug_id=0001589 Has anyone else heard an audible blip, break or garble between answer and the native bridge attempt using sip? If I change the usleep(500000); to usleep(5000); in rtp.c the proble totally goes away... even the note above it says it needs to be fixed. bkw -------------- next part -------------- An HTML attachment was scrubbed... URL:

Sorry if this dupes, didn't seem to post to the list when I emailed it a few hours ago. Regards, Dean Collins Cognation Pty Ltd dean at cognation.net +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). ________________________________ From: Dean Collins Sent: Thursday, December 06, 2007 2:49 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: "Happy

Hey all, I am trying to figure out a weird problem. I have a two node ctdb samba setup where the first node acts as expected setting the default gateway (startup shutdown takeip recoveryip) if there is a network blip / link down, but the second node does not set the default gateway properly on the initial recovery of the link after the blip has occurred. If I restart the machine or restart the

I have what amounts to a simple search engine running through a table using input from a form. Everything works fine when looking for "title" or "body" But, when I want to couple that with an AND statement the search still works fine, but it seems to disregard the AND statements. Code below: In the view: <%= start_form_tag :action=> "show_search" %>

Anyone on this list have extensive experience with BTS? http://deancollinsblog.blogspot.com/2009/03/open-bts.html Please email me, particularly if you have experience in deploying over multiple cells covering large geographical areas (200k's sq). Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York

The bar is getting raised yet again http://www.voipmonitor.net/2007/06/05/New+Features+And+Services+For+Pack et8+Virtual+Office.aspx I personally use Snapanumber $30 or there abouts (after trialing a few other TAPI solutions and finding them sub-par) and think it's a great product but interesting to see how more people are expecting desktop/phone integration applications. Does anyone

Is anyone out there using Druid? After the switchbox announcement today I've been looking into some other gui's and as I'll probably do a trial install this weekend of the free switchvox iso but I thought I'd ask is there any other guis I should be burning trial ISO's of as well? Regards, Dean Collins Cognation Pty Ltd dean at cognation.net <mailto:dean at

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

I need a country, state, city database for a web application. Anyone have a free version they can email (or drop.io) for me? Looking for something like this at $197 but may as well ask in case you know of a free source. http://www.globixdata.com/pop.cfm?db=world&v1=l&v2=s&v3=a&pricing=99 Regards, Dean Collins Cognation Inc dean at cognation.net

Hello list- I'm having an extremely odd issue with an installation of mine. The system is running * 1.2.12.1 and currently handles around 100 handsets. With the exception of a few Grandstream DTA's, all devices are Polycom 320, 430, or 601's. After a recent power outage, I'm having an extremely odd issue with one of the handsets. One of the Polycom 601 units simply reboots every

Does IPKALL still exist? I am after a free SIP trunk - who is still giving these away these days? As I noticed Stanaphone is no longer in business? Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York +61-2-9016-5642 (Sydney in-dial). +44-20-3129-6001 (London in-dial). -------------- next part --------------

Meetme cant handle more than 5 users in a call?? Hmmmm http://www.voip-news.com/feature/asterisk-voip-pbx-right-choice-032707/ hmmm I'm all for commercializing a product, but this FUD from Fonality seems to be taking it just a little too far Regards, Dean Collins Cognation Pty Ltd dean@cognation.net <mailto:dean@cognation.net> +1-212-203-4357 Ph -------------- next