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All, We are in the process of trying to setup our network to use Verizon's SIP "trunking" product. They say that since Asterisk is not on their certified list of approved devices, we need to go through a field trial to get it approved before allowing us to use their service. Where we are at is getting the design approved. We are trying to watch our budget at the same time. We

I have a couple of questions about Digium's T1 cards, such as the TE410P. Any answers would be greatly appreciated. 1) Do they support standard T1s or are they ISDN-only? 2) Do you know of anyone offering support for configuring T1s for Digium cards, and if so at what cost? Thanks, Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian

I am looking for estimated hardware requirements for running a 50 to 100 seat call center off of a single Asterisk server. The Asterisk server will have one quad T1 card installed (probably a Digium TE410P) with two T1s connected. The OS is Debian GNU/Linux (woody) with a custom 2.4.xx kernel installed. It is preferable for the server to have a single CPU and no shared IRQs. I would really

Hello, Does anyone know the developer(s) of the app_queue.so application? I'm looking for the PauseQueueMember and UnpauseQueueMember features of this application for the open source version that only seem to be available on the business edition of Asterisk. Thank You, Timothy Karl tkarl@imminc.com

...;unload" and "load". I didn't have much luck looking through the archives or wiki on this, but I was wondering if anyone had any suggestions. Any feedback is appreciated. Thanks much, Franklin Webb Assistant IT Project Leader Inter Medi@ Marketing Solutions 610-701-9670 fwebb@imminc.com

...run into this issue any maybe have a solution, or does anyone know of a good way to get that call back onto the Asterisk switch from another extension prior to calling chanspy? Thanks much, Franklin Webb -- Franklin Webb Asst Project Manager Inter Medi@ Marketing Solutions 610-701-9670 fwebb at imminc.com

Is anyone out there using SIP Witch in conjunction with Asterisk? It claims to be able to "enhance existing IP-PBX solutions such as Asterisk", so maybe it can be used as a simple means to provide secure/encrypted calls. GNU SIP Witch - Summary <http://savannah.gnu.org/projects/sipwitch> GNU SIP Witch - GNU Telephony <http://www.gnutelephony.org/index.php/GNU_SIP_Witch>

Greetings: I'm in the situation where I'm trying to splash information picked off by an asterisk IVR into a Cisco call center environment. I'm under the impression that the ONLY way to do this is to setup socket connections with the Cisco "voice processor", or CVP, and send packets corresponding to GED-125. Cisco has a detailed 100+-page document detailing the internals of

Greetings all, We are settng up a fair sized call center on Asterisk, but we are having some issues with our agents not knowing if they have logged in and logged out. Prior to beginning our migration to VoIP the agents logged into our nortel phones and confirmation was displayed on the phone. My question is has anyone out there done anything from Asterisk that can change the display on

Hi, I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is "cut". Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? my extension.conf for meetme: ;switch =>

Friends, I have updated my Asterisk 1.2 presentation with the latest information. It is still available in the same place as before: http://www.astricon.net/asterisk1-2/ Please continue to test the beta of Asterisk 1.2, available at ftp.digium.com. We need all the feedback we can get. If you are a developer and have some time for community work, please check in with the bug tracker and help us

Hi Asterisk Users, we are using Asterisk 1.4.18.1 on debian 4.0 etch, pwlib 1.10 and openh323 1.18. We are using MeetMe for conference calls and with two participants there is no echo problems, but with more than two participants there is a lot of echo that sometimes disappear for a short time and all function well. Someone have some suggestions?? Do you ever used app_conference

...------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070329/ddbfcf c3/attachment-0001.htm ------------------------------ Message: 17 Date: Thu, 29 Mar 2007 11:52:02 -0400 From: "Matthew J. Roth" <mroth@imminc.com> Subject: Re: [asterisk-users] maximum simultaneous calls To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <460BE0A2.8070206@imminc.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Mark Quitoriano wrote: &...

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

List users, Did anyone else recently receive spam from DIDForSale with the subject "DIDForSale 2012 achievements"? I suspect that they are using this list to harvest email addresses and think they should be called out on this poor business practice if that is the case. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer

I beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or open ports, and bang them real good. We need two things: a) disable in sip.conf the reply for INVITES that have wrong user

I'm looking for suggestions as to the best multimedia headsets for a call center environment. A few considerations: 1) USB headsets are preferable, because they don't require a soundcard. 2) Omnidirectional microphones are problematic, because they pick up too much background noise. Thanks, Matthew Roth http://www.voip-info.org/tiki-index.php?page=Running%20Asterisk%20on%20Debian

Hello, Can anyone recommend a hardphone that has the following qualities... Both headset and handset ports Headset port has amplification built in Supports SIP using G729 We are switching from a Nortel switch to Asterisk. If anyone is familiar with Nortel phones, the Nortel 2216 phone has the features we're looking for in a SIP phone. Any help would be greatly appreciated. Thank you.

Hello, I'm trying to find out if Asterisk will support plain G729 or G729b. I've read all over that it supports G729, but I can't seem to find any explicit remarks regarding the specific versions of the codec Asterisk will support. I noticed that Digium allows Asterisk users to register and download G729a, but refers to it as G729 on it's pages. I also noticed that on

Hello, I'm trying to write a php script that issues the QueueStatus and Queues manager API commands to Asterisk and records the returned event data into a MySQL table. On voip-info.org, I see that QueueStatus returns such things as Queue, Max, Calls, Holdtime, Completed, Abandoned, ServiceLevel, and ServiceLevelPerf. However, there are no descriptions for these values. I've tried