similar to: Review: Digium TE405P v2

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 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

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

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

Hi Everybody, Is there anyone who already uses Asterisk with TE405P card? Is this card going well with the old E400P or T400P drivers? Are there a lot of differences? In the card's data sheet writes this sentece: "Digium has designed the TE405P to be fully compatible with existing software applications..." Is this meaning the the old drivers are also OK for this card? Thank you

Hi, Can please someone help me. I have successfully installed Asteriskathome 2.5 on a server with a Intel Server Board SE7525RP2. May issue is after placing the TE405P in the server, it is not booting anymore. Has anyone in here have the same experience? Can someone please point me to the right direction. Thanks in advance, Leonimar __________________________________________________ Do You

lspci dont show Digium card. The motherboard is Intel 945gnt. 00:00.0 Host bridge: Intel Corporation 945G/P Memory Controller Hub (rev 02) 00:02.0 Display controller: Intel Corporation 945G Integrated Graphics Controller (rev 02) 00:1c.0 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 1 (rev 01) 00:1c.2 PCI bridge: Intel Corporation 82801G (ICH7 Family) PCI Express Port 3 (rev

Jared Smith, Jim van Meggelen, and Leif Madsen of the Asterisk Documentation Project, in conjunction with O'Reilly Media are pleased to announce the official release of Asterisk: The Future of Telephony on Friday, October 14, 2005 at AstriCon 2005 in Anaheim, CA. In the true spirit of Open Source, the authors and O'Reilly Media have published the book under the open, Creative Commons

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

Hello all, I really checked voip-info.org but it still seems to be not very easy and I just hope that there is anybody with a simular config. We have one PRI (euroisdn with 30 channels) coming from the DTAG. The second PRI should be connected to a Siemens Hicom 150E Pro Office PBX (was cheaper than a channel bank :-) Carrier ----S2M------ * -----S2M------Siemens | |

Hi, we successfull managed to bridge a PSTN (E1) switch over the TE405P card to our old PBX. So now we could migrate to the * server. But, there are two things we can't live with: 1. A call from the outside to the old PBX is missing a leading 0 before the number. Ex: caller has number 0123456 -> * routes to old pbx -> old pbx sees 123456 as caller number. 2. A call made from a SIP

Did anybody know, Is it possible to establish a ISDN DIAL up Connection and Analog Dial up Connection (V90) trough asterisk with Digium TE405? Thanks a lot for help. Nico Giefing -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060328/3b657058/attachment.htm

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 getting ready to purchase my first Digium card to start experimenting with Asterisk. Before I make my purchase, I wanted to make sure I'm not going to have issues with these cards (need to see what the specs are on my box, 5V or 3.3V PCI ). I will be using Asterisk @ Home, so will be Asterisk v1.0.9. I took a quick poke at the lists, and it appears several people have been having

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

Anyone care to elaborate on the differences between the T400P and the TE405P? -Darren -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050908/e817f8bb/attachment.htm

Forget the sound card. It isn't related. The subject above should have read 'TE405P No Voice Problem' or something similar. It appears to be a zaptel timing issue, but I have found a workaround. For those of you just tuning in, here is the story: I have a CentOS/Intel 865 box currently running Asterisk 1.2.7.1 and zap 1.2.5, both compiled from the source available off the main

This works fine. Message: 26 Date: Tue, 7 Jun 2005 17:22:23 +0800 From: "Ma Zhiyong" <mazhiyong@sohu.com> Subject: [Asterisk-Users] te405p and dell poweredge To: <asterisk-users@lists.digium.com> Message-ID: <004501c56b42$6a12fcd0$7301a8c0@hylaclient> Content-Type: text/plain; charset="gb2312" Hi, I want to install * and te405p on Dell Poweredge 1850. Can I

Hello On a Sipura SPA-841 handset (and also at other end) you hear a sea wave like sound - it gets louder then softer and continually repeats. I don't remember hearing this when using other handsets. But what is this effect? How can I reduce it? Angus