similar to: AGGRESSIVE_SUPPRESSOR

I was wondering if anyone knew how to do the following.... Call comes in, gets put into a Queue, say `Sales`. Then the queue member is presented with the option to exit the queue, yet have the phone system sit in their place for them. When the virtual member reaches the front, call back the caller and connect them to the agent. Any ideas? Did i explain that ok? :) Cheers, Ben Merrills

Hi, I had been using 4 X100P cards in my Asterisk box, but 2 of them were sharing an interrupt. Therefore, periodically I would hear beeps and clicks that I had assumed were a result of this. So, I ordered a TDM400P with 4 FXO modules and installed it in the box last night. Today, we've had nothing but problems with it dropping calls. I installed the latest CVS of everything, and we've

Using TDM400's here and I have tried everything to cure the echo. I have used the Milliwatt test from the telco and from asterisk to tune RX/TX gain via a patched ztmonitor. What happens is I experience midcall echo. I turned on aggressive_suppressor and it seems to do great. The problem happens with misc. noise around the office will cause it to mute the other end of a phone call while

Hi iam installing zaptel 1.2.10 on my FC5 when i make iam getting following error any one suggest me whats wrong, i have installed source also in the same server. grep: /lib/modules/2.6.15-1.2054_FC5/build/include/linux/autoconf.h: No such file or directory ZAPTELVERSION="1.2.10" build_tools/make_version_h > version.h.tmp if cmp -s version.h.tmp version.h ; then echo; else \

I've got echo problems. *** I'm looking for paid support. *** I'll accept free support, but don't mind paying if someone really knows what they are doing. I've read the wiki, etc. Played with the settings in zapata.conf Using V400P PSTN->_T1->_ASTERISK->_BROADVOICE->_PSTN ECHO ON CALLED PHONE PSTN->_T1->_ASTERISK->_T1->_PSTN NO ECHO

I'm using bristuffed Asterisk with ISDN/ZAPHFC I have VERY ugly (outgoing) sound through ISDN/HFC if echocancel=yes in zapata.conf, but without echocancel I have bad (incoming) echo Through PSTN/FXO sound is ok with or without echocancel. I tried other echo cancellers (in zconfig.h) two times: ECHO_CAN_KB1 (this was default) ECHO_CAN_MARK2 ECHO_CAN_MG2 after any change I compiled (make

I greatly appreciate the help at last Monday's installfest and especially am grateful to Chris and Lenny. Now I am reconfiguring the installfest computer to add Zaptel using the instructions at http://www.voip-info.org/wiki/view/Asterisk+Zaptel+Installation There is now a Digium 4 port card installed. In regards to Zaptel there are some questions on zconfig.h. Should I: #define

I'm trying to get zaptel 1.0.2 compiled on FC2 or FC3 and I'm getting compile time errors. Systems include: FC2: Linux xxxxxxxxxxxx 2.6.9-1.3_FC2 #1 Mon Nov 15 14:46:43 EST 2004 i686 i686 i386 GNU/Linux FC3: Linux xxxxxxxxxxxx 2.6.9-1.681_FC3 #1 Thu Nov 18 15:13:22 EST 2004 x86_64 x86_64 x86_64 GNU/Linux /usr/home/bwright/zaptel-1.0.2> make linux26 ... make -C /usr/src/linux-2.6

Good Day all, I have a Fractional PRI connected to my Asterisk Box via a T100P card. When I initiate a call out to phone number 123-8888 the call sounds great no echo what so ever. If the person at 123-8888 hangs up and calls me right back (same handset on both sides) same trunk line The call always has echo on it. The Asterisk sip extension hears them selves echoing. The remote party

Has anyone tried integrating Asterisk and Rhetorical's rVoice software? We're evaluating different approaches to system announcements via T2S. Has anyone gone down this route that could give some advice? I've installed festival and wasn't too impressed, the demo one the website seems far better quality and clarity then the defaults in the source package. However I must admit

Hi all. Just wanted to let people know about a small project I started over the weekend to help me keep up with news about Asterisk. http://asterisktimes.xdev.net/ Some of the other new sites are either not there anymore or slow to update, so I've come up with a different idea for keeping Asterisk news up-to date and in one place. For the moment, I call it Asterisk Times. OK, so maybe not

Hi, I'm still strugling with getting an easy to use conference system implemented. I did have app_conference running, but today I upgraded asterisk to 1.0.9 and it stopped working. I've tried following the instructions for compiling app_conference on 1.0.7 but it didn't work. So I went back to ztdummy (I've not had any luck getting this to compile on FC2). Anyhoo, I've

I have a bad echo problem on my TDM400P with one FXO module installed. I have tried a few things, such as: * setting rxgain and txgain to 0 * setting echocancelwhenbridged to no / yes * settting echocancel to 64 / no / yes * setting echocanceltraining to 800 / no / yes * MG2 echo cancellation * MARK2 echo cancellation * KB1 echo cancellation * AGGRESSIVE_SUPPRESSOR option of MARK2 Each time

Is there a way to log all PRI events to a logfile? Cheers, Ben Merrills Griffin Internet T: 0870 8040862 F: 0870 8040805 W: www.griffin.com <http://www.griffin.com/> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041123/d983141f/attachment.htm

I have a 12 channel PRI with SNOM 190's and asterisk CVS from January. Most calls are fine, all incoming calls are fine, but I am getting echo on a significant number of outgoing calls. The person on the other side hears a perfect call, but the SIPphone side gets to hear themselves. It happens 100% of the time to some numbers (outgoing only), and only sporadically to others. Has anyone

Hi, I''m encountering another highlighting issue. (about the first one "range search and highlighting", i received no response. I don''t even know if somebody tried to reproduce and/or if it''s normal behavior) about the new issue, an example will be easier for you to reproduce: I''m filling an index with random data, i try to match for "*1*"

Is there a way of getting the dialled number from an AGI? Is it passed in the initial variables, or can it be pulled out or passed across from the dial plan? Cheers, Ben Merrills Griffin Internet -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040630/71e0bcda/attachment.htm

Hello, I''ve modified tools/xenstore/xenstore_client.c to include a new utility, xenstore-chmod. This utility permits developers and administrators to manually change the permissions on arbitrary locations in XenStore from the command line. This is often helpful if you''re trying to debug an application that relies on XenStore and is encountering difficulties with permissions.

Thank you for the suggestions Andrew. I have not come across some of them before and will give them a shot. Based on my reading, changing the motherboard should have minimal impact unless that motherboard and the TDM400P don't get along (aka. IRQ sharing). I have disabled everything that is not needed and I do not believe I have any IRQ problems and I am NEVER wrong ;). Calls are crisp and

Problem: Zap Channels Disappear @ random intervals. (Channels have disappeared on both gateways twice this week). My Setup: I have 2 Dell 1850 Power Edge Servers - Configured as. P4 2.8Ghz 512 ECC Memory SCSI Array (2 Drives Mirror) Configuration is really simple the boxes are setup as PSTN termination gateways: (SIP/ZAP) - We are running a SER/Asterisk Hybrid. My asterisk