similar to: yes shady dial running now but not dialling

Dear Mongrel Users, It appears that Francis still isn''t satisfied with my attempts to apologize and he''s now asked me privately to retract my statements. So, just to be sure he finally gets it: I retract my statements that Francis and his friends are "shady" and "desperate for cash". He is in no way a shady guy, having dealt with this situation with the

wondering if anybody knows this......does shady dial work only with a zap interface or can it be configured to be used with SIP or IAX. Nauman -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of asterisk-users-request@lists.digium.com Sent: Thursday, July 08, 2004 5:48 PM To: asterisk-users@lists.digium.com Subject:

Looking for a good termination provider for US/Canada Please contact offlist. Shady -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061219/75bb266e/attachment.htm

I?m getting nowhere with this. From the website below I clicked on the ActivePerl 5.26 button which seems to lead me into creating projects on the Cloud rather than downloading the software to my hard drive. https://www.activestate.com/products/perl/downloads/ Can someone give me some advise? What the website is trying to do seems rather shady. Thanks, Philip [[alternative HTML version

On Thu, 19 Apr 2018, Valeri Galtsev wrote: > On Wed, April 18, 2018 8:36 pm, Always Learning wrote: >> Hi, >> >> I have a machine with a BIOS that does not permit DVD installation. It >> accepts everything else including some old superseded media types. >> >> Is it possible to download C6 combined parts 1 and 2 not using Torrent ? >> >> I have an

Is the "Whats New at Digium the Asterisk Company" message I got from digium at en25.com really from Digium? If so I suggest to send it from digium.com and not to use those shady Eloqua redirect URLs. Regards, Philipp Kempgen -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create new ones. Asterisk?

Hi, What OS are you on? It has been years since I used ActiveState, but it looks like you now need to create an account with them prior to downloading the installation files. They seem to give you the option of creating an account with them, or using Github. I would opt for the former, even though I have a Github account. If you are on Windows, an alternative Perl distribution is Strawberry

I'm working on an application that is backed by Postfix 2.3.8 and Dovecot 1.0.13, configured with virtual domains, for handling messaging. I need to be able to send the notification (such as "You've received a new message on SomeAwesomeService. Click here to check it out", It's evil, I know) to a different email address, such as a mobile device. Eventually, we want

Hi, I was just wondering if anyone knows how predictive dialers detect voicemail and answering machines, and if they could explain to me how that works. Thanks! Brian.

Just thought I''d let you know that your http parser worked great for creating an http protocol handler for Eventmachine. It wouldn''t take that much effort to use Eventmachine for Mongrel, might be worth a shot just to see how it does. A few things would need to be restructered, like HttpRequest where you read the rest of the body, and calling the handler. But it

Hi Zed, I checked in a pure-Ruby URI classifier to Mongrel trunk. Ola''s Java port of the TST had some bug, and I don''t think it''s necessary in the first place. The Ruby classifier is around 25 lines instead of the 400-odd lines for the C extension and the 200-odd for the Java extension. It uses a Regexp which is perhaps shady: @matcher = Regexp.new(routes.map do

As i m very begginner to AJAX and script.aculo.us , Kindly provide me very basic level tutorials regarding them with examples aslo Thanks --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Spinoffs" group. To post to this group, send email to

So I just saw this VoIP-centric article at slashdot (http://slashdot.org/article.pl?sid=04/05/22/1840220) which mentions e164.org. It's a "non-profit public DNS root designed to map phone numbers to Internet protocols." Is anyone on this list actually using this? They have asterisk config instructions: http://www.e164.org/config.php I wonder if someone can help me understand

Hi, suppose I have agents waiting on a queue and I configure asterisk to dial out and to forward the call to the first agent enqueued. Asterisk will do it even if the answer to the call is "busy". Is it possible to configure asterisk to detect the busy signal and, in that case, dial another number, without wasting agent's time? Thanks

Hey all, I'm having problems reliably dialling out my FXO card. About 30% of the time I'll get a "your call cannot be completed as dialed". I'm thinking it might be the dialling speed, but I can't find any configs that change that setting. Any suggestions for troubleshooting? Thanks, Brad Waite

Hello, I'm experimenting with Overlap Dialling in asterisk 1.6. I've enabled this in sip.conf and on the SNOM 300 phone. My problem is that asterisk dials out as soon as it matches an extension without waiting to see if the user is going to type in more digits. Is there a way to set a timeout per channel or globally? I'd like Asterisk to wait for a few seconds once its found a match

Hi :) I have a little 'slow dialling' problem. When I dial, e.g. 200# for the Asterisk 'echo test' demo application from my PBX extension 1010, I see this in the console the instant I press the # key: -- Starting simple switch on 'Zap/65-1' -- Accepting overlap call from '1010' to '200' on channel 0/3, span 3 then exactly 3 seconds elapses, and

hi asterisk users How can i make asterisk "ignore" invalid extensions, and go on playing the background soundfile? Normally, asteriks will take the user to the invalid extension if the caller presses anything other than 1 or 2 in the following context:: [example] exten => s,1,Answer() exten => s,2,Background(hello-world) exten => s,n,Goto(s,2) exten =>

Using 1.0.9: If I have: exten => s,1,Dial(SIP/5555&SIP/12345@192.168.1.1) How can I return the DIALSTATUS variable for the second SIP channel ONLY if the second SIP channel is busy, regardless of the dialstatus of the first SIP channel? What I want is, if the second SIP channel is busy go to n+1 or n+101 regardless of the status of the first SIP channel. tia

Hi, Red Hat 9.0 Asterisk 1.2.7.1 I'm having a bit of an intermittent problem with my SIP account. Often (but not always) when I start * or RELOAD my dial plan from the CLI I get this message: >Apr 30 11:01:20 WARNING[12785]: chan_sip.c:11822 add_realm_authentication: Format for >authentication entry is user[:secret]@realm at line 31 >Apr 30 11:01:21 WARNING[12785]: acl.c:244