similar to: Problem with chan_iax.c implimentationcausesbadaudio?

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 11:36 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio? On Tuesday 21 March 2006 11:19, Adam Robins wrote: > All switches and routers give

We upgraded all five servers to 1.2.4. We tried trunking/notrunking. End users use an IAX2 softphone on their desktop PCs. Agents are VLANed and all IAX2 traffic is QoS'd on all LAN and WAN legs. Calls flow from the agents to the local Asterisk server as IAX2/ulaw. Then they went over the IP-VPN as IAX2/g729 (we tried ilbc and straight ulaw as well). Calls get to the PSTN from the

We have three remote call center Asterisk servers communicating with two central Asterisk boxes over a private IP-VPN with QoS. All systems were running Asterisk 1.0.7 communicating via IAX2 with little or no quality issues at all. Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was horrific. We tried with/without jitterbuffer. We messed with every jitterbuffer parameter. We tried

Just looking for unsolicited thoughts on the Aheeva product? Anyone have anything to say? Thanks, Steve Totaro http://www.asteriskhelpdesk.com

Hello all, I tried to compile Asterisk from CVS yesterday. I would like to try gnophone as IAX client on the same Asterisk server with a sound card. I didn't have any problem of compilation. Here are the config files in /etc/asterisk: /etc/asterisk/asterisk.conf /etc/asterisk/extensions.conf /etc/asterisk/iax.conf /etc/asterisk/indications.conf /etc/asterisk/logger.conf

Hi All, I do have asterisk installed for a call center and I would like to know if it is possible to create a scipt and execute it from a PC connected to the Network without accessing the server. This script should restart asterisk and another service related to aheeva. The problem now is that each time I have to access using PUTY to the server to start and run services manually. Service

I received an e-mail from a vendor who says: "We have recently become aware of an issue in the chan_iax2 implementation of IAX2. This issue leads to degraded audio quality. Due to this we are urging everyone to move to SIP." I don't want to discount what this person is talling me, but I'm curious to know why I would only be having issues connecting to his servers, and also what

Hello! Continuing my PRI saga - I have a PRI setup and appears to be answering calls OK, but my carrier is cutting all the calls after 15 seconds. For example, when I call from my cell phone, it goes straight to a busy signal - however the CLI shows the call coming in and being answered. Additionally, when I call from another ground line, it will ring once or twice, again show as answered, but

RJ11 Plantronics 1 3 2 4 3 1 4 2 RJ11 Pin 1 is on the left when looking at the contact points. Plantronics Pin 1 is on the left when looking at the contacts (through the plastic sheild) My multimeter battery is low, so YMMV, but: Pin's 1,4 are connected with ~160 ohms Pin's 2,3 are connected with ~1400 ohms On my Plantronics head set. Chad -----Original Message----- From:

Hey all, after hooking up some Plantronics to some Snom's (3 320's 1 360), I noticed my client is having some form of feed back on the phone. Because of Snom's "inner oddities" this is how I got it to work. Plantronic --> RJ11 --> SnomHandset Port (on Snom Base) Handset --> Plantronic jack (bottom base in the front) If I placed Plantronic(RJ11) --> Snom's

Tom - here are a few suggestions Plantronics 510SL Bluetooth - they make a bundle that comes with a Bluetooth transceiver and a handset lifter. This is nice if you have users that already have Bluetooth headsets for their mobile phones, as it allows you to sync with their office phone as well. http://www.plantronics.com/north_america/en_US/products/cat29880043/cat2 9880054/prod5460016

Hello All, I do have asterisk installed for a call centre with aheeva application and i would like to know how to configure the sound for the inbound calls and if there is any possibility for agent to receive a file with the phone number and name of clients: For your information there is no problem related to the outbound call An help would be appreciated Kind Regards Salah. --------------

I have echocancel=yes in zapata.conf but when I do a zap show channel 1, I notice echo cancellation is turned off. I followed the article that talks about the order in which the statements need to be in zapata.conf to get echo canceling to work: http://lists.digium.com/pipermail/asterisk-users/2005-June/110615.html But it is still not working. Does anyone know how to get echo

I have a system that is using Intel HDA as its output, and a Plantronics USB headset as the microphone input. I would like to have my wine configuration emulate this setup, I currently have been unable to confirm that it isnt... I just cant get Steam to choose the plantronics headset as the default voice device (this is the goal). Would appreciate any insight!

What kind of call center: inbound, outbound or both? how many lines per agent will you have? what kind of trunks will you be using? do you need to tie into an existing database? do you want screen-pops? MATT--- -----Original Message----- From: Zeeshan [mailto:ztahir@gmail.com] Sent: Tuesday, August 02, 2005 7:27 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject:

Not strictly asterisk related, however... Here's an odd one for you.. I got a Nokia E90 and setup it's SIP client which runs via Wi-Fi (anyone know if I can make it work via GPRS/3G?) Anyway, in a fit of idleness, I thought I'd see what codecs it supports, as I couldn't find it in the manual... And it supports: ilbc g729 ulaw/alaw No GSM! How odd is that, given

Steffen Kaiser <skdovecot at smail.inf.fh-brs.de> writes: > On Tue, 10 May 2016, Jon Kristensen wrote: > >> Which Pigeonhole Sieve extension should I use to process/test a >> message >> depending on whether the message is in a particular mailbox? >> (Is it the >> mboxmetadata extension by any chance? If so, I would appreciate >> if >>

Anybody using Plantronics USB headsets? What softphone are you using and whats your overall experience? Any comments/suggestions?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi! Which Pigeonhole Sieve extension should I use to process/test a message depending on whether the message is in a particular mailbox? (Is it the mboxmetadata extension by any chance? If so, I would appreciate if someone could tell me what the reason is for it not being enabled by default.) The reason I want to perform this test is that I have

I am looking for SIP with power over ethernet desktop phones that will work with asterisk and Plantronics HL-10 Handset Lifter for Remote Answering. Any suggestions? I am considering buying about 150 of these desktop phones for a new call center.