similar to: NICE analog recorders

Hi list, I'm wondering if there's a way for multiple users to share the same voicemail box and have their BLF flashing when voicemail comes, i.e. in a home phone system where there's a general vm for everyone. I'm using couple Grandstream GXP2020. Any suggestions? Kelvin Chan | Positronics Ent. Product Development | | unit 272

Hi guys, I'm building a server that need to host 2 digium TDM24 cards. I know any 3U server with 2 PCI-E slots would do. Since I do prefer supermicro server, but getting one configured is pretty darn hard. Any suggestions here? Cheers, Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St.

Hi guys, I'm looking for a affordable conference phone and a wifi phone that has a cradle. Polycom seems to make pretty nice conf phones but the price is a bit crazy for us. I saw the recommendation with ATA plus an analog Polycom phone but I do prefer a SIP phone. All because it's just too difficult to pull a phone cable into the current conference room. Is there any cheaper SIP

Hi guys, I'm trying to move away from meetme to loose the dependency on dahdi. ConfBridge seems to be a good fit but I can't get it going. The document sounds like an easy to use app. Am I missing any bridge_ modules? Asterisk 1.6.2.0~rc2-0ubuntu1.2 -- Executing [521 at outbound:1] Answer("SIP/109-b877a8c8", "") in new stack -- Executing [521 at

Hi guys, I have a bunch grandstream phones using ulaw and my users are complaining they are jittery when I use "canreinvite=yes". The data connection is an ADSL link dedicated for phone traffic. At any given time, I have at most 2 calls in parallel. I'm not a huge fan of asterisk being in media path doing buffering because the delay (jbmaxsize=80,jbimpl=fixed) is pretty long and

Hi guys, This has been bugging me for days and I can't figure out what's happening. Using the same channel on DAHDI, I can't make a call after previous one being hung up by remote side. However, it works like it should if my end hangs up the call. To make things interesting, DAHDI, or astribank rather, seems to "reset" itself after receiving an incoming call on that DAHDI

Hi, I am looking to connect 66 analog phones to an asterisk box. I was thinking of a Xorcom astribank 32port (2 of them and another 8 port). this is because the phones have no near connection to an ip network, so replacing the phones in favor of voip phones+network cabling is kinda out of the question. In your experience, will these units support all the phones talking at the same time with

Hi, I have a problem with people that are calling from analog lines. We have a block of numbers 12345 - 0 to -99. Most calls are transmitting the whole number including the extension. There's no problem with that. But people calling from analog lines are connected to our asterisk box as soon as they finish dialing 12345. They don't get a chance to dial an extension. Just inserting a

Hi, all My Asterisk uses a Digium TE120Pand I would like to add an Astribank zaptel_hardware sees is, but I cannot get it working pbx:~# zaptel_hardware Argument "IRQ" isn't numeric in numeric comparison (<=>) at /usr/local/share/perl/5.8.8/Zaptel/Span.pm line 114. usb:005/002 xpp_usb- e4e4:1131 Astribank-8/16 USB-firmware pci:0000:04:00.0 wcte12xp+

I have recently updated from Centos/*1.2 to Ubuntu Server and FreePBX 2.8.0.2. We have an Astribank with 4 T1 ports and 16 FXS ports. After updating, we had it working for a while with one NT PRI and one TE PRI and, in the process of trying to configure another PRI, I ran into a couple problems. (1) As my configuration changes didn't seem to affect the Astribank, I power-cycled it. I

The Asterisk development team is pleased to announce the release of Zaptel 1.2.13. This release contains a large number of bug fixes, an important performance improvement for most Digium cards, and support for new Digium hardware and some significant improvements in the XPP driver for Xorcom's Astribank hardware. In detail: * A modification was made to the drivers for all Digium PCI cards

The Asterisk development team is pleased to announce the release of Zaptel 1.2.13. This release contains a large number of bug fixes, an important performance improvement for most Digium cards, and support for new Digium hardware and some significant improvements in the XPP driver for Xorcom's Astribank hardware. In detail: * A modification was made to the drivers for all Digium PCI cards

I am running Asterisk on Ubuntu 2.6.32-25-server with asterisk 1.6.2.5-0ubuntu1 and dahdi 2.2.1-0ubuntu2. The machine has a passive HCF-based PCI ISDN card and an Astribank 8 attached. The ISDN card works fine. root at servaction:~# lsusb Bus 001 Device 002: ID 04b4:8613 Cypress Semiconductor Corp. CY7C68013 EZ-USB FX2 USB 2.0 Development Kit root at servaction:/usr/share/dahdi# ./xpp_fxloader

Hi, I have upgraded an Asterisk installation with a Xorcom BRI Astribank that was working under Ubuntu 8.04 to Ubuntu 9.10 and the device is no longer initialized. When I reload the udev rules, I see that the rules seems to be correctly loaded: udevd[452]: reading '/etc/udev/rules.d/40-xpp.rules' as rules file However, these rules do not seem to be triggered: # udevadm trigger

My equipement : 2x tdm 400p /4FXO/4FXS 16 PORT ASTRIBANK My first TDM400p is the sync master, so i set astribank to sync to it, but the quality is bad, like it is going fine and second after >>ROBOVOICE ;) any other devices (isdn/sip/tdm cards) are working fine, looks like astribank dont like to be sync 'slave'. Is there any way to change sync master manually, so every another

Hi Guys, An 8 channel Astribank is connected to Trixbox 2.8 and I ran freepbx-module-zapauto but I get the following when running these commands and can't make calls out: [Trixbox]# dahdi_genconf xpporder /usr/sbin/dahdi_genconf: warning - OLD DRIVER: missing '/sys/bus/xpds/devices/00:0:0/timing_priority'. Fall back to /proc pbx*CLI> dahdi show channels Chan Extension Context

Hello, I am trying to update to dahdi 2.2.1. But somehow the xorcom (2BRI) device does not want to work. I pasted below the information I have. pbx at loicvoip1:~# /usr/share/dahdi/xpp_fxloader reset pbx at loicvoip1:~# /usr/share/dahdi/xpp_fxloader usb 'xpp_fxloader'[4937]: --------- FIRMWARE LOADING: (usb) [1 devices] ..Got all 1 devices 'xpp_fxloader'[5003]: ---------

Hi Kelvin, The audio bandpass setting is only configurable when the encoder is instantiated (eg: start of a Voip call ) , but you can change the bitrate anytime.For example if you can read incoming RTCP packets , you can check if there's reported packet loss , and then lower the bitrate accordingly.Yes, the app has to be aware of the packetloss ?percentage. Cheers,Dragos From: Kelvin

I have a .csv table named mailing.csv as below. It consist a receiver, subject and sender. Receiver subject sender 1 Adrian Cole RE: [WHIRR-117] Composable services Tom White 2 Adrian Cole RE: [WHIRR-117] Composable services Tom White 3 Adrian Cole RE: [WHIRR-117] Composable services Adrian Cole 4 Adrian Cole RE: [WHIRR-117]

I'm having trouble getting a xorcom set up. A large part of the problem is that the box is a _long_ way away and I can't get to/at it easily, so while I could probably fix this in a few hours if the machine were in front of me, I'm struggling over a slow unreliable laggy link. Ok, enough whining from me. I have a new Xorcom plugged into the usb of a Suse 11 machine I built Dahdi