similar to: odd 'digital' sound artifacts

Hi, I've had some decent luck with the mark3 echo canceller from the zaptel driver, echos on about 20% of the calls, people I've called say I sound great now, but our side hears echos. I was wondering if there was any way to tweak the current software cancelers into using more CPU (and hopefully doing a better job, close to a hardware canceler), I only have 10 lines, and a single call

Hi list, I was wondering if anyone had any solution to either one of two issues I'm having: I have a cisco 7962G with the latest (from cisco) 8.5(4) SIP Firmware, it works very well for the most part, but after less then a week of heavy usage, eventually the phone gets into a state where it won't accept or let you place any more calls, the screen flashes "no free lines

Hi list, This is not necessarily an asterisk issue, but a lot of you guys know way more then me, so I have a question: someone at my company sets his phone to forward calls to his cellphone, so someone calls our office, call is forwarded to his cell, and the callerID that shows up on his cell is of course our office number, because asterisk originates a new call to his cell and then bridges

I have several Cisco 79xx phones (7905, 7920, 7940, 7960, 7970, ATA 186) and I need to buy firmware for them. I have contacted http://www.cdw.com and http://www.insight.com/ but they didn't respond. Can anybody tell me where can I buy SCCP and SIP firmware for my phones? BTW, I'm in Croatia (Hrvatska). I heard that location does matter. P.S. My local Cisco reseller wants to sell me

When a call is placed out the Zap interface there is a long pause followed by an error message from the telco that the call can not be placed as dialed. We have a tdm2413e with 11 1FB (POTS) lines. The number being dialed is a working local number, all dialed numbers get the same error. What should we be looking for ? Best regards, Duane Pudenz Network Infrastructure Manager Shasta

Hi everyone, I'm having a bit of an issue after upgrading from asterisk ~1.2.24 to 1.6.2.11, with the old version when the user would go to transfer a call, they would press Transfer, then the speed dial button for the extension, optionally introduce the call, and then press Transfer again to complete the transfer. Now, with the new version, when you hit the speed dial button, asterisk

We want some of our users to use G729, and some others to use ULAW. Our PSTN gateway provider supports both, so that's not a problem, and if I force him (the PSTN gateway) to allow G729 only, the outgoing call will take place with G729. The problem is that I want to have my PSTN provider configured to allow ULAW as a first priority, then G729. I did it like that: [mypstngate] type=friend

The REV I card shows up in the PCI table as: 02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or 02:05.0 Class 0280: e159:0001) Subsystem: Unknown device b119:0001 But the REV E/F shows up as: 02:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface (or 02:0d.0 Class 0780: e159:0001) Subsystem: Unknown device b100:0003 One

Been around asterisk for two-plus years, but need a little input from the list on this topic. Have a potential client that wants to replace their old key system with *, but they want to integrate a commercial message service (they pay a monthly fee to have special MOH messages generated) into their system. The messages are essentially delivered to this customer via older generation audio

Hello List, I work for an IP communication provider in upstate NY as the engineer assisting our technical support team. We provide a number of different Telco systems to residential subscribers; and in an effort to more effectively trouble shoot termination problems I came up with the idea of creating a click to call system that will allow our agents to effortlessly place test calls. On a

I just bought an I Audio mobile device (Btw, these thigns are the coolest players on the face of the earth.. might want to check em out). It plays ogg vorbis files rather nicely. My whole music library is sitting in flac format. When I encoded my flac files (the majority a year ago) I used replay gain to save the album and track information if the flac file. Unfortunately, the I audio doesnt like

Hi, I have a requirement to limit the calls to our agents via a queue to 5 minutes. I had posted this to a previous thread by name "Maximum talktime in a queue?" One work around that was suggested was to use the S(x) in the dial command to the agents, so that all calls to that extension would be terminated after x seconds. So I modified the dial command to the agent as: exten =>

Guys. I have an issue with a te110p card and also some tdm04b cards on the same system: Zttest returns this for the tdm04b cards: [root@mollendo ~]# /usr/src/zaptel-1.2.4/zttest 38 -v Opened pseudo zap interface, measuring accuracy... 8192 samples in 8191 sample intervals 99.987793% 8192 samples in 8192 sample intervals 100.000000% 8192 samples in 8192 sample intervals 100.000000% 8192 samples

Hi, I have a TDM400P running with only one FXO port running on a VIA EPIA MS10000 (1Ghz Via Eden). When I run zttest from Zaptel 1.2.1 it hang and when I interrupt it with Ctrl-C that is the result: ?anyone have some idea about why isn't working? Some additional info: # /usr/src/zaptel/zttest -v Opened pseudo zap interface, measuring accuracy... --- Results after 0 passes --- Best:

Hello. I thought I'd introduce myself. I'm the guy that wrote kexis way back when. For those of you that dont know, kexis was written before flac, and pretty much does what flac does, although its not nearly as nice or robust. I wrote kexis actualy cause I was interested in playing around with lossless compression and shorten was about the only tool that was linux friendly, but had a

Hi, My asterisk server was working with a 4-FXO analog card (TDM400P). I recently added two digital cards: a TE120P (1 PRI) and a B410P (4 BRI). The B410P is still unconfigured but inserted in a PCI slot. The TE120P's jumper is set to E1 as it will connect to a commercial PBX's PRI card also configured as E1. My analog channels used to be 1-4 but since I added the new cards I changed

Hello, I have a TDM400P with 4 FXO ports, currently using three. When sending or receiving calls on this card, there is a nearly constant popping/clicking sound, it is related to the echo cancellation?. I adjusted my gains properly, but to no avail. I even found that setting echotraining=no in zapata.conf didn't change the scenario at all. I've plugged analog handsets into the

All the 'unknown device' means is that your 'lspci' doesn't know what the card is. That's all. Nothing more. --Rob ________________________________ From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Aryanto Rachmad Sent: Tuesday, 4 October 2005 7:43 AM To: asterisk-users@lists.digium.com Subject:

--- Wayde Milas <wmilas@rarcoa.com> wrote: > On Tue, 2004-01-20 at 23:44, Josh Coalson wrote: > > Miroslav did some experiments with searching for optimum blocksize. > > from what I remember it made at best a couple percent difference. > > there was a thread about it here a while back. > > > Did his changes make it in? I can think of a coupla ways to approach

--- Wayde Milas <wmilas@rarcoa.com> wrote: > The last one is truely frightening. AT&T hold the patent. The link > for > it is at: > http://patft.uspto.gov/netacgi/nph-Parser?u=/netahtml/srchnum.htm&Sect1=PTO1&Sect2=HITOFF&p=1&r=1&l=50&f=G&d=PALL&s1=5025258.WKU.&OS=PN/5025258&RS=PN/5025258 > Trying to read and grok it is mind numbing.