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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:

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

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

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

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

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

...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 4/5/06, Matt <mhoppes@gmail.com> wrote: > > On 4/5/06, Matt Florell <astmattf@gmail.com> wrote: > > > In my experience capacity is a huge problem. You can't > > > have sphinx running on 48 channels at once. It is > > > limited to only a few instances at a time. Although I > > > only did trials with sphinx2. > > > > &g...

Hello, We've released another update to our astGUIclient suite: 2.0.2 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the astGUIclient client-side web app which extends your phone's functionality and the VICIDIAL call center suite. This package is free and GPL. (the suite is not an asterisk

Hello, We've released another update to our astGUIclient-VICIDIAL suite: 2.0.4 http://astguiclient.sf.net/ The client suite runs on most modern web browsers on almost any GUI-capable operating system, and it includes the VICIDIAL call center suite and the astGUIclient client-side web app which extends your phone's functionality. This package is free and GPL. (the suite is not an

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

Hi, Just downloaded the latest cvs from zaptel on my sparking new Athlon64 Centos4.2 system, but hitting a stumbling block... (sorry for the long post) #make cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64 -DSTANDALONE_ZAPATA -DZAPTEL_CONFIG=\"/etc/zaptel.conf\" -c -o gendigits.o gendigits.c cc -o gendigits gendigits.o -lm ./gendigits cc -I. -O4 -g -Wall -DBUILDING_TONEZONE -m64

Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example: If * detects Answering Machine or Voicemail, hangup call & the AGI will log (ANSWERING MACHINE DETECTED) and at that point,

The Asterisk Manager Interface is driving me nuts. Whoever wrote it should be drawn and quartered. Sometimes the data comes back separated by \r\n, and sometimes it's separated by \n. The whole thing is completely inconsistent, and trying to write any kind of API for it is -GHASTLY- Doug.

Ok... I am reluctant to ask this question as I believe that it may be like asking what someones favorite linux distribution is... but I need to make an informed decision. We are getting ready to upgrade from a TE210P to a quad T1 card with echo cancellation. I am trying to decide between the Sangoma card and the Digium card. I need this to have great quality and I need it to work well. I would

...we got the channel set > to the correct context all was well. > > > Again thanks, > Rod > -- >> TIA, >> Rod > > > > > ------------------------------ > > Message: 18 > Date: Fri, 21 Nov 2008 10:42:12 -0500 > From: "Matt Florell" <astmattf at gmail.com> > Subject: Re: [asterisk-users] How long will Asterisk 1.4.x > supported/maintained > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <asterisk-users at lists.digium.com> > Message-ID: > <61575c810811210742j6...

...happened on a number of users. Obviously there are fixes in 3.8.6, so I don't want to downgrade the phones again, but I can't see why they'd fail... Adrian Marsh ------------------------------ Message: 16 Date: Wed, 22 Aug 2007 07:31:02 -0400 From: "Matt Florell" <astmattf at gmail.com> Subject: Re: [asterisk-users] compatibility of PRI Two B channel transfers TBTC/2BTC To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <61575c810708220431p7522cdder75de1ef09c16ec08 at mail.gmail.com&...

Hello, We've released another update to our Asterisk GUI Client suite: 1.1.5 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 auto-dialer. This package is free as in GPL. (the suite is not an asterisk configuration tool) This package is geared towards

Hello, We've released another update to our Asterisk GUI Client suite: 1.1.6 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

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