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...r my new SNOM200 phone: *CLI> *CLI> NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '<sip:mattf2@10.10.10.15>' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" <sip:mattf2@10.10.10.15>' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line 4713 (handle_request): Registration from '"Matt Florell" <sip:mattf2@10.10.10.15>' failed for '10.10.10.14' NOTICE[1125329600]: File chan_sip.c, Line...

I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced

This is a global message to all to announce our callcenter / dialer / predictive dialer / vicidial program is now open. Codecs: G711, GSM, G729, G723 Protocols: SIP Duration Rate : 30/6 (6/6 with monthly minutes over 100,000) Channels : 100 to start with , more on demand. We are predictive dialer friendly , your account will not be shut off. Contact us to do a test run. Mike

Hello, Now that we've had our new Digium TE406P card in production for 4 days we have discovered audio drop problems that happen randomly across all channels. Here's more about our setup: P4-3.2GHz 2GB ram Slackware Linux 10.1 with custom kernel 2.4.29 Asterisk 1.2beta1 Digium TE406P quad T1 card with the following attached: - 2 x RBS D4/AMI 24 channel T1s - 1 x RBS B8ZS/ESF 24 channel

Hello, We have finished our tests of the new Digium firmware on the quad T1 cards(TE405P/TE410P). Overall it is a big improvement over the version 1 firmware. Here's the review: http://astguiclient.blogspot.com/2005/09/digium-405p-v2-review.html MATT--- -------------- next part -------------- An HTML attachment was scrubbed... URL:

...must not be disclosed or distributed. Quark Group Pty. Ltd. T/A Quark Automation, Quark AudioVisual, Quark IT > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] > On Behalf Of Matt Florell > Sent: Thursday, 6 April 2006 01:45 > > The load on the system will crash your server with that many instances > of real-time sphinx running. Take a look at 'top' while you run it on > tow channels at once an see what the load is. > > MATT--- > > > On 4/5...

So this is just a general question, Is Asterisk really good? Reliability? Functionality? Customization's? I am coming from a Nortel world, were you pay for everything, and you can't delve into the software. But it seems that customization would be a great thing. Like, setting up a war-dialer to customer lists, incoming/outgoing faxes (that's possible with Asterisk, right?) and

Hi, Matt: What?s your server specifications that did you use? Best Regards, Cleviton. -----Mensagem original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]Em nome de Matt Florell Enviada em: sexta-feira, 23 de junho de 2006 11:38 Para: Asterisk Users Mailing List - Non-Commercial Discussion Assunto: Re: [Asterisk-Users] Meetme max users We've had over 100 participants spread across 30 meetme rooms on a single server before, and the most we've had in a single mee...

Hello, We've released another update to our VICIDIAL/astGUIclient call center suite: 2.0.5 http://astguiclient.sf.net/ The call center suite client applications run on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite. This package is free and AGPLv2. This package is geared towards Asterisk installations with SIP,IAX or Zap

I currently have Asterisk deployed in my office with a TE411P. On the first port of this card is a T1 from the telco setup for D4 AMI. Unfortunately, I'm not receiving caller ID on inbound calls from this line. The caller ID information is arriving in the form *ANI*DNIS*. In zapata.conf, I have signalling set to em_w. The DNIS always arrives correctly, but I'm never receiving the ANI

I am trying to resolve an issue and I believe it is my configuration. The scenario is that I have a SIP detected on the server. The dial plan then makes a local connection to another part of the dial plan. The new dial plan extension then places another SIP call out to a SIP phone. When the call is accepted there is streamed from the calling SIP phone. When the audio is complete a DTMF is

It sounds like Naren and company has their own CRM application. They need a predictive dialer that allows third party app integration. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Matt Florell Sent: Monday, March 13, 2006 8:03 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Need help implementing call center featuresofAsterisk Hello, There are two GPL Asterisk-based outbound call center systems available, GnuDialer and VICIDIAL. You can fi...

Hello, A client has asked for Two B channel Transfer capability (known as TBCT or 2BCT, similar to other features such as ECT, RTL and Q,SIG Path Replacement) in a new Asterisk system and so I researched the capability and came up with quite a few gaps in documentation.

Hi guys, Recentlly i did a asterisk gateway and use it with an alcatel pabx. All is working, i have only two problems. 1. When call incomming to asterisk, it forward to digium card to PABX Alcatel. The user that start the call can't hear the control tone of ring ring ring. Tha calls stay without sound until the called part answer the call. At this point, conversation follow normaly. 2. When

Hi everybody Im the IT Manager for a new call center and my bosses has assing to me a very dificult task i have to configure the call center using Hp 5520 thin clients, asterisk and some kind of dialer that allows outbound calls. I triyed using terminal services but it dind worked because the lack on the sound and the microphone do not work on the thin clients using terminal services, we tried

Kevin P. Fleming wrote: > Falck Kenneth wrote: > > Thanks, I was misguided by > > http://www.voip-info.org/wiki-Asterisk+Variables which > didn't mention > > this. > > You are more than welcome to edit the page to make it obvious > to the next reader :-) You're quite right - I added a little note there to warn others now. It's a great Wiki anyway,

I received a TE406P card from Digium yesterday and have done several tests on it and cannot get it working reliably with any Stable release or even CVS v1.0. What are other people that are using the TE406P card using for an Asterisk version? I tried 1.2.0 beta1 and it seemed to function well except for an unrelated issue with the beta release where the manager interface chokes under

Hello, We've released another update to our Asterisk GUI Client suite: 1.1.7 http://astguiclient.sf.net/ The client suite runs on Windows, UNIX and Mac, includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL client-side web app auto-dialer. This package is free as in GPL. (the suite is not an asterisk configuration tool) This package is

hi building a new setup, we want to try using sangoma cards. can these be used as time sources the same way as TE410Ps? thanks roy

On Fri, 2006-10-13 at 10:50 -0600, Joseph wrote: On Fri, 2006-10-13 at 07:27 +0200, Remco Barendse wrote: > > On Thu, 12 Oct 2006, Eric "ManxPower" Wieling wrote: > > > > > Matt Florell wrote: > > > > If you downgrade, let us know if it fixes things for you. > > > > > > > > It's strange that there were so many changes in the 1.2 SVN branch > > > > after 1.2.7.1 that seem to be complete changes in how some things > > >...