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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

...is to make sure APIC and IO-APIC support is turned on, if your system supports it. (All SMP motherboards do, and many single-processor ones do as well.) This seems to give access to 32 interrupts instead of 16, minimizing interrupt sharing. -----Original Message----- From: Dennis Webb [mailto:dwebb@delta-express.cc] Sent: Friday, March 11, 2005 2:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM04B lock up Should ACPI be turned off in the kernel? In the bios I can only set cards to the 1-15 interrupt range, but linux and acpi it seems moves...

Disabled echo canceller because of tone (tx) on channel 10 I understand that the PSTN companies use their own echo canceller's, send a tone across 2100hz, the problem we're having is people are complaining of echo on random calls. I'm assuming this may be the cause. Is their anyway to 'ignore' the disabling of EC? Or would be just be a manual code change.. Matt

Im completly new to the whole PBX thing. I have a toshiba unit now and we're moving our office in the next few months. I want to use asterisk but would like to test it out first. Does it support a basic analog phone line like the one in my house? Is that FXS? Are there any FAQs I should read to learn more? Thanks for the reply! -------------- next part -------------- An HTML attachment was

Any help on this would be great I have 2 TDM400P's, 2 asterisk servers (running on powerful boxes with FC1 and * v CVS 1.0.02), and 4 analogue PSTN lines from BT and whatever I do, I cannot get rid of this damn local echo. Ive tried setting the echoTraining, echoCancel (in phone.conf and Zapata.conf) , echocancelwhenbridged to every possible combination , Ive even tried running the fxotune

I have searched the lists and voip-info and having trouble with a call parking issue. When I park a call and it times out, it seems to immediately tries to goto exten s in whatever context the person who parked the call is in. Voip-info under config features.conf that it will ringback to the original extension. Is the original extension the person who parked the call, or the original extension

Not everybody here has voicemail and we have more than one person answering the phones. I would like to know that if Person A transfers to Person B, that after a predefined timeout, it would then transfer the call back to Person A. Right now when a new call comes in, it rings 3 phones. When Person A transfers to Person B, it tries for 15 seconds, then transfers the call back to all 3 people

Ok, I've pretty much decided to try the Adit route. Somebody who has experience with these tell me if I'm missing something. I have 15 incoming PSTN lines. T1 is not an option at current location. I want to put in an Adit 600 with 2 8-port FXO boards. The adit will then connect to * via a digium t1 board. I configure zaptel.conf for the T1. What other parts would be needed? How do