similar to: Re: time to build an open phone?

> > I've never seen stats, but it's probably a safe assumption that the > > majority of IP phones are sitting next to a PC and the additional > > expense has been incurred because "people want a phone that looks and > > works like a phone". That's certainly been my experience far > > outweighing any technical issues with quality or reliability

Does anyone know of a piece of hardware that can allow multiple earpices to be connected directly to a server running Asterisk. I hope I am not being to vague but basically I am looking to allow a call center to user the server to do all of the "Pickup" and "Hangup" functions. The operators will merely have to have th earpiece in their ear. I have seen serial pieces of

Before committing to about 50 of the spa942's, I like to take a last poll from those on the list to identify any negative issues that might be associated with the audio, functionality, early failures, etc, on the spa942. Expecting to deploy these using existing cat5 cabling and both rj45 jacks. Been using three of theme in a short term demo with the customer, but the demo systems has

hi i've configured a Beronet BN8S0 Card with misdn beronet utility. the card is configured with all lines in TE and P2P mode, and it is connected with the special cable with an ISDN connection. i've turned on debugging to level 4, this is the output from the asterisk cli when i receive a call: P[ 5] MGMT: Short status dinfo 1000001 P[ 5] MGMT: SSTATUS: L1_ACTIVATED P[ 5] handle_frm:

Some smartcard readers have keypad to enter the PIN securely (i.e. such that it cannot be intercepted by a rogue (ssh) binary. PKCS#11 allows for enforcing this in hardware. Below patch allows for SSH to make use of this; against head/master as of today. Dw. commit 7f0250a8ae6c639a19d4e1e24fc112d5e2e1249a Author: Dirk-Willem van Gulik <dirkx at webweaving.org> Date: Tue Mar 17

I have been looking at chan_bluetooth, so far being unable to make it compile with Asterisk SVN trunk. I was wondering about the different ways it can be used. What I have read so far implies two possibilities: 1. Asterisk pretends to be a handsfree unit, and can use the cell phone for placing calls over the mobile network, or answer inbound calls from the mobile network. 2. Asterisk

I have a call center where we're looking at converting it from a traditional PBX w/ digital phone "agent" sets (keyless phones) that have headsets to a SIP based environment. I am having trouble finding anything on the market that resembles this in the VoIP world. For reference, we're currently using Inter-Tel Agent Sets, which are basically a digital phone with out any keypad,

So has anybody got one of these? http://www.amazon.com/exec/obidos/ASIN/B0007LQQUK/qid%3D1106972010/sr%3D11-1/ref%3Dsr%5F11%5F1/102-1529886-6420131 I'm thinking that it should be possible to connect it directly to an Asterisk box and not use their software, as long as there was Linux support for the USB dongle. Maybe it just looks like a standard USB audio device? But there has to be an

Greetings list, It seems that more and more phones these days are coming with XML mini-browsers. I'd like to have a go at developing something useful to use on them, but in all honesty, most of our customers use their phones to make and take calls and very little else. So I'm open to suggestions. What useful applications are you developing for these mini-browsers? What sort of things do

I've been playing with the outgoing call spooling feature a bit lately and it all works as it should with the exception of one irritation. I'm mostly using SIP to talk to the phones and using G.723.1 I copy the call file into the spool/outgoing directory and the originating phone rings. I pick it up and the remote phone rings. However there is dead silence from the originating

Use Set(CALLERID(num)=9999999999) instead of using CALLERID(all). Remember to set this BEFORE you Dial. -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Vieri -->> Sent: 06 February 2009 12:36 -->> To: asterisk-users at lists.digium.com -->>

Our basic system is as follows: P4 3.0 Ghz w/ HT, 1GB PC3200 RAM, 120 GB HDD, RH 9.0 OS, * from CVS several weeks ago, working OK for routing, VM, and AA, calls in on separate PSTN lines to Adtran TSU 600, into * server through T100P card. The hardware is not taxed at all with little over 20% proc utilization ever, low mem use, etc. All Phones are SNOM 200's with various firmware revisions

When talking to me, people are complaining the volume was not high enough. The phone only allows to change the volume of the speaker/earpiece. Is there an alternative solution? Is it possible to increase the volume in asterisk? Frederic

> > Different brand/model, but similar as they are both el cheapo, > entry-level headsets. I tried using them on a laptop, and I get > marginally better microphone output, even with its volume cranked all > the way up + automatic gain control enabled. > > I guess those on-board soundcards by Realtek aren't as good as a > quality microphones. I'll get a USB headset

hi .. I have an asterisk system with three TDM100P (single port FXO) cards and 10 Grandstream 100 phones connected to it .. 1st question: when i phone out or receive a call from one of the SIP phones onto the PSTN, there is a LOT of local echo in the handset .. the PSTN end of the call does not here this echo, but it's VERY annoying on the SIP end of things .. the echo seems to be about 0.3

Hello everyone. I've got this really annoying HFC Cologne card (or however I should call it - a single channel ISDN card based on the HFC chipset). It wrongfully detects lots and lots and lots of incoming DTMFs, to the point the card is not usable. Here's a sample out of CLI: P[ 1] I IND :DTMF_TONE oad:206361 dad:520101 P[ 1] --> mode:TE cause:16 ocause:16 rad: cad: P[ 1] -->

You're quite right. We'll need to see your misdn.conf file to check the settings. -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Vieri -->> Sent: 06 February 2009 13:49 -->> To: asterisk-users at lists.digium.com -->> Subject:

What headsets have people found work well with the Cisco 7940 and 7960 phones? To date, I have tried a couple of the headsets within the Plantronics H series (H41-N), and noticed that the volume of my speaking is lower over the headset than on the regular handset. I am currently looking for headsets that are known to work well. I do know that Cisco lists the H-91 and H-101 as certified to

Michel Send me the same output for a dial string that only sends the *31* Is this an ISDN line? What type of card/signalling/switchtype are you using? It looks as if the PSTN switch accepts the *31* and then hangs up so you can make the NEXT call with the *31* feature enabled. If so I assume the *31* feature will be enabled for the next call on the ENTIRE SPAN if it is an ISDN trunk group. If

I am trying to use misdn insted of zaphfc to drive two billion isdn cards zaphfc is ok, but the problem with cdr and the fact tha you always have to wait the bristuffed version of asterisk took me to try another way. so I downloaded the misdn installation script from beronet for the last version ( I am using asterisk stable 1.2, so now is 1.2.5) wget