similar to: WAV file volume

I am sending a call down a E&M wink trunk to a adtran tsu600 channelbank. The extension is setup like so... exten=>799179,1,Dial(zap/g2,20,D(9179)) exten=>799179,2,Hangup() It should Dial the Adtran and send some DTMF signals to a telephone on an fxs module in the Adtran. Asterisk is seeing the call answered when the T-1 is picked up by the Adtran not when the ringing phone is answered.

Can anyone share any advice / documentation on how to configure a Tellabs 2572 T1 echo canceller? I connected one between a T100P and an Adtran TA750 FXO/FXS channelbank, but when echo cancellation is active I get a LOT of snap-crackle-pop (and other problems) on the line. The 2572 has a bunch of configuration options and they're impossible to guess without documentation, which I cannot find

Dear Colleague, I am writing to ask your help for a book I am writing about Life Practices. I am looking for examples of those simple things that we all do that are essential to expressing and maintaining our values as we face life's ups and downs - our Life Practices. What I am seeking is a bit like folklore. Something with a story and an easy to remember slogan or catchphrase. I am asking

Greetings, I'm trying to setup an option in my greetingmenu that would allow the caller to select this particular option for emergency calls. That option would dial out on an available PSTN line to a cell phone number. Currently it is setup as such exten => 9,1,Dial(Zap/g1/<CELLPHONENUMBER> where <CELLPHONENUMBER> is the number it is calling out to. When option 9 is

Dear Colleague, I am writing to ask your help for a book I am writing about Life Practices. I am looking for examples of those simple things that we all do that are essential to expressing and maintaining our values as we face life's ups and downs - our Life Practices. What I am seeking is a bit like folklore. Something with a story and an easy to remember slogan or catchphrase. I am asking

Is there any way to take an incoming callerid string and remove the given "name" part of it and replace it w/ something arbitrary, or add to a blank name string (possibly by looking up the number in a database)? Thanks, John Lawler

I'm trying to get musiconhold to work w/ my Asterisk system, and I'm not having much success yet. First, I noticed that nothing happened even after I had enabled all of the options in zapata.conf & setup a sample extension in extensions.conf. Then I read something about how Asterisk uses mpg123 to play the files. I discovered that this had not been installed on my system, so I

On Tue, January 27, 2015 1:58 pm, Peter Lawler wrote: > On 28/01/15 04:47, Always Learning wrote: >> >> Saw this on the Exim List:- >> > <SNIP> >> >> I use Exim on C5 and C6 - should I be worried about Exim on C6 ? >> > > upstream references: > https://rhn.redhat.com/errata/RHSA-2015-0092.html When I read this I read that it is fixed in

Hi guys, I posted a somewhat similar question about a month ago and got a thoughtful resonse from Steven Critchfield, but I've got a quick follow up question to it. I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1

Hi all, I'm wondering how I could go about placing * into an existing office PBX system so as to capture callerid for further processing via AGI into an intranet app. I've already got the AGI scripts set up for what I want to do and have tested using IAX2 peering, however I have no knowledge of the target environment as yet, so would like to know what do I need to look out for (line

I have following setup Asterisk - T100P -> Adtran TSU600 P + FXOcard -> PSTN line When PSTN line is plugged directly in to analog X100P caller id is received by Asterisk but when I plug it into adtran I'm not getting caller id. Any ideas what kind of setup Adtran TSU600 requires to pass caller id to T100P ??? regards m.

I recently set up Asterisk with an Adtran TA750. All is well except the phones do not show the MWI. I have configured zapata.conf properly, as all phones will receive a stutter dial tone if there is a message waiting in it's assigned mailbox. Does anybody know how I might fix this problem? Thank you for your time __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free,

Hi guys, I've run into a problem that I can't figure out on a bunch of handsets I have running into a Rhino Equipment 24-port FXS channel bank hooked up to a T100P and running asterisk-0.9.0 and the associated stable Zaptel release. The sidetone (your own voice that you hear in your handset, built in for comfort) is noticeably louder than it should be, and it doesn't seem to

That could explain why it wouldn't work on any of my sip extensions I tried it on this morning when I first read about it and thought cool the things you learn. Is there anyway to make it work on Sip extensions? Cheers, Dean -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Aaron J. Angel Sent: Wednesday, 23

On Thu, April 16, 2015 8:59 pm, Peter Lawler wrote: > [OT ALERT] > > On 17/04/15 02:28, Valeri Galtsev wrote: > >> clamav is a scanner that is designed to detect viruses (virii I should >> use >> for plural as it is Latin word) > I believe this 'rule' in English is misunderstood by many and as a > general rule of thumb... > tl;dr: > Words from Old

All, I need help. I have an (actually 2) Adtran TA750's with 8 FXO ports. I get a terrible buzz on every FXO port. If I unplug the Adtran and put an analog phone on each incoming line, I have no buzz. I also have 2 Carrier Access Access Bank I's with 12 FXO ports. When I plug the same analog lines into either one of those, no noise or buzz whatsoever. I went so far as to move the TA750

We are a voip terminating company, we're using Channelbank with FXS modules, Rhino, CAC, etc.. What we're wondering is, is how to would you echo cancel a channelbank. Of course we're realizing that cancel'ing on the T1 (on Ast) does no good (we think?) because the analog conversion is at the channelbank. Suggestions? Lowering the gain helps but we're looking for a real solution

Hello, Did anybody successfully tried upgrade Adtran TSU 600 to firmware which is working properly with T100P and asterisk ? B.

Hi! Is there anyone that are using a E1-channelbank and have any tips about some type? Im looking at the TE410P and use one port for a PRI (Euro-ISDN, I think we're using some slightly modified version here in Sweden, but I'll check that tomorrow) and connect one port to a channelbank for 30 analogue telephones. It would also be great to get callerid on the analogue phones, so it would

Hello. I have been going thru the wiki and asterisk related sites and have not been able to find any documentation about how to configure an Adtran TA750 channel bank. The remote disconnect supervision doesn't seem to be working, when the remote caller hangs up asterisk takes up to 30-45 seconds to hangup the call. Can somebody help? Thank's