similar to: User in two queues receive two calls at once

Hi all, We need to get better echo cancellation on an Asterisk gateway. Currently it has two TE410P (1st gen) cards. So would it be possible to just buy two VPM450M cards ? Or do we need to buy two new TE412P cards ? In that case a Sangoma A108d card would be nice as well ? What configuration gives the best audio quality ? Thanks, Leon de Rooij leon at scarlet-internet.nl

Hi all, I have Asterisk working great for a while now, and until now, only needed to forward calls which originate from one country (Holland, +31). We have a Wildcard TE410P card and configured it in zapata.conf as: switchtype: euroisdn pridialplan: unknown ...etc This was all fine, but now we want to also forward calls to the voice-switch with a calling-number with a different country code.

Hi, I needed to create config files for downloading to Grandstream devices and made a little script for it. It seems to work fine for the HT486. The script needs a config file (cfg.in) which is in this format: P2 = blah P10 = hrm ...etc... The configfile may contain comments (starting with '#') and empty lines. Mind that the 'gnkey=0b82' shouldn't be in the configfile, as

Hello all, Since a while some of our SIP users complain about gaps (sometimes multiple seconds of silence) in the RX audio stream (direction pbx -> phones). Our configuration is an Asterisk with two Wildcard TE410P cards that are connected with E1 PRI's to an external server running callcenter software. The SIP users use Snom and Polycom phones. The gaps are not present in

It took a little longer then expected, but here it finally is, a field test for the idefisk for linux iax2 softphone. Freely downloadable from http://www.asteriskguru.com/tools/ You will probably need to copy the iaxclient lib into your library directory and run ldconfig before starting the phone. Please note that this is the first copy in the wild of the linux version and is not as tested

] Hey ho, A few days ago we released the linux version of the phone, today we are very happy to have the mac version ready for a little field test. Freely downloadable from http://www.asteriskguru.com/tools/idefisk_mac.php At the same time, we also put a newer version of the windows and linux versions online. Let us know how you feel about it, a more mac look (brushed metal) is coming.

We just uploaded the latest and greatest version of the idefisk iax2 softphone, version 1.24 Freely downloadable at: http://www.asteriskguru.com/tools/idefisk_beta.php Changes since the last release include: - history panel is working - receiving messages and urls (sendtext command in asterisk) - some bugfixes (the annoying hangup bug is finally gone!). A big thanks to everybody who sent us

Hi, I''m still a new with ROR, but it''s really cool :) I just have this problem with a project of mine. I want to store email in a database. I already wrote a simple import script, that works, so that''s not a problem. Showing the email (efficiently) in a ROR site is. These tables are in the database: emails id email_id (foreign key -> emails.id) seq

Hi, I'm new to telephony Software and Hardware, so please excuse my questioning. I plan to set up a little system, using Asterisk and VoDSL via Belcacom or Scarlet here in belgium. We are yust a little 2 man company and we are not always in our office. My idea is, to get VoDSL and set up a system that works as following: A customer sends SMS or phones to our office-numbers, if we are out,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Dear samba team, I created a folder on my website (french linux website) about this huge movement of Novell. And for make my articles, I want to have the official position of samba team about this accord. thanks St?phane Purnelle - -- St?phane Purnelle <stephane.purnelle@scarlet.be> Site Web : http://www.linuxplusvalue.be -----BEGIN PGP

Hi, 1. Has anyone got any success when send a TIFF file form one zoiper softphone to another ? I tried using Zoiper 2.18 free edition in windows but I'm seeing 415 Unsupported media replies. 2. Here (http://www.voipinfo.org/wiki/view/Asterisk+T.38), you can read : "Also, try using: t38_udptl=yes t38pt_rtp=no t38pt_tcp=no ... in the general section of the sip.conf and under the VoIP

List users, Over the last few days we have been working with MCI's development lab to test our Asterisk setup. We were using a piece of hardware called an Abacus 5000 that is capable of creating and terminating thousands of SIP calls. Initially, we could not get past 64 simultaneous digitally recorded calls without having call quality issues including dropped calls. We identified an

Hi list, I have 2 softphones, 1 Idefisk (IAX), 1 Xlite (SIP) registered to Asterisk. One call the other-one, is it possible to order Asterisk to force answering the call ? i.e. Xlite call Idefisk, Idefisk is ringing, I send a command to Asterisk which force answer, so Idefisk answer the call without clicking on "Accept" button. Greg -------------- next part -------------- An

I've been working my way through the softphones listed on voip-info over the last few weeks and I've not really found anything to fit the bill. Has anyone had more luck? The environment is a small call centre of 5 users. Operators often need to be able to transfer calls to other operators with different specialties, so the softphone needs to be easy to use and quick to transfer calls.

I am new to the R language. I am trying to plot multiple figures on one page through a loop, but the code just produce one graph on one page. Can someone show some light on what's wrong? Here is my code: library("quantreg") tcdata<-read.table("mydata.txt",header=TRUE) postscript("myfigure.ps") basins<-

Hi, Did someone ever managed to make Astertest (http://www.asteriskguru.com/tutorials/astertest.html) work ? I followed all the instructions of this tutorial and corrected the mistakes pointed by the users but it still doesn't work. I can compile it and load app_securax_cpuinfo.so. When trying to load app_securax_serverload.so I have this error : WARNING[31477] : loader.c: 325

Cool, i'll help out a bit with the windows port, i will start right away with a new project on asteriskguru making nightly executable builds and installers - will post the links in -users when i'm done. Well done luigi, this will make it a lot easier for a lot of non linux guys to make their first steps in the asterisk world Crossposted to -users. Zoa Luigi Rizzo wrote: > As a

The Asterisk Development Team today released Asterisk 1.2.9 and Asterisk 1.0.11 to address a security vulnerability in the IAX2 channel driver (chan_iax2). The vulnerability affects all users with IAX2 clients that might be compromised or used by a malicious user, and can lead to denial of service attacks and random Asterisk server crashes via a relatively trivial exploit. All users are urged to

Hi, This is the first time I've posted, and I apologize if I formulate this incorrectly. I am analyzing data from a multi-region carrot variety trial. 35 varieties of carrots were grown in 3 randomized complete blocks in organic and conventional fields in Wisconsin, Indiana, Washington, and California. In this example I am comparing the heights of the carrot tops at harvest. In other

Hey gang, I've been able to use sipp to produce some call volume on our asterisk server. The server has no problems handling 50 simul calls. But then again, no RTP is being done. I tried to use the rtp echo ability of sipp but that doesn't seem to work right. I also setup a fake number in asterisk that when called by sipp, would dial another number via PRI, hoping that some 729