similar to: Incoming DID call Voice Problems

Hello All, I've mostly solved my DID problem from a few days ago. Apparenly the lines weren't configured properly. Now heres the next question. 12 E&M wink lines from telco. I have them all plugging into an Adtran 750 with FXS cards. The Adtran ports are configured DPO. How do I signal this from Zaptel. I have them setup E&M in zaptel.conf and EM_W in zapata.conf. They

I've been trying to get a Win 2000 RAS server working with my asterisk PBX for quite some time, to no avail. I've googled, I've tried loads of configurations, I've rewired phone lines, and still I am not winning the battle. Here's my config. PRI->T400P->Asterisk->T400P->Adtran 750(L36 Firmware)->RAS Server. I have 4 Zap channels signalled FXO_KS to the 750

Has anyone on the list been able to locate and try out the 1.1.0 firmware? It was released in November, but I have yet to get my hands on it. The Polycom site has way more docs online, but the link to the firmware only brings up the release notes. -sb

Hello All, Quick question here on equipment selection. Is there any benefit to using a bunch of Adtran 850's with RCUs over 850s with BCU L1 to interface with the T400P. I'm looking at building a PBX with 7 channel banks. 6 x 24-port FXS and 1 x 24-port FXO. Would the RCU give me any advantage/functionality over the BCUL1? Thanks, Scott

Does anyone have any recommendations for overhead paging systems for use with Asterisk? Thanks, Randy Johnson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031113/82ae09ec/attachment.htm

Caller*ID used to work as some point, but I can't seem to get it going these days. The card is a x101p. I've tried going up and down the rxgain scale. Can the txgain effect it at all? When I plug in a phone into the line with a splitter it can decode caller id with no problems. Reading through the mailing list archives hasn't given me any move clues. Any ideas?

Hello All, I'm having a weird problem when connecting up to a TA 750 from adtran. The problem I'm seeing is that the third wire on my 66 block is behaving as the tip (or ring) for every extension. Is this indicative of a bad BCU? The only extension that works properly is extension Zap 2. Every other extension is crossed with Zap 2. Very weird. Anyone see this before? Did I get a

Asterisk apparently has five echo cancellation algorithms: STEVE, STEVE2, MARK, MARK2 and MARK3. The current default appears to be MARK2. My question is, has anyone had any experience with any of the others (other than MARK2), and is there some conventional wisdom as to when to use one over another? TIA Bruce Komito High Sierra Networks, Inc. www.servers-r-us.com (775) 236-5815

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Andrew > Kohlsmith > Sent: Thursday, September 29, 2005 2:23 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Best echo canceller? > > > On Thursday 29 September 2005 17:04, Claudio Canseco wrote: >

I just had the same exact problem this morning. The only thing I've done in the last couple of days is update update zaptel. I rolled back my zaptel to 2/11/04 from 3/8/04. And kept my libpri from 3/8/04. I never had this error before updated. I had other issues, but not this one. -sb -----Original Message----- From: asterisk-users-admin@lists.digium.com

Dear R users The following question looks simple but I have spend alot of time to solve it. I would highly appeciate your help. I have following dataset from family dataset : Here we have individuals and their two parents and their marker scores (marker1, marker2,....and so on). 0 means that their parent information not available. Individual Parent1 Parent2 mark1 mark2 1 0 0

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

Hi everybody!! I'm trying to assign boolean values to a column in a matrix depending on the number of times present in another column. I have no clue, I have been 2 days looking for a way, if somebody knows what kind of functions I need to use I will really apreciate it. example: Color Type Mark1 Mark2 Mark3 Red high 139 P alpha blue low 140 P alpha Green high

hi! In your experience what is the best choice for echo canceller ? Which one should work better: STEVE, STEVE2, MARK, MARK2, MARK3, KB1 ? Regards, Claudio -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050929/f556bf91/attachment.htm

I notice the zaptel Makefile option the mark2 option & KFLAGS+=-DAGGRESSIVE_SUPPRESSOR is now gone. Does simply adding these options still compile in a certain echo can - or is there an other method of activating a particular can. I have not had to update my machine that is connected to pstn for a while & I don't want to jump into echotraining without a way to quickly enable what has

Hi, I have scanned through the archives of this list and found a number of question about hardware, but I just can not find the answer to my question. I am new to phone systems, I got "drafted" to come up with a new phone system for our company (I guess they figure since I know computers I know phone systems as well :O). We have 5 analog (I guess they are called PSTN lines) lines coming

Hello, We have an asterisk setup at our client's site using a TE205P. The line to telco is a 23 channels T1 PRI, however the line has random echo problems (about 5-10% of the calls)! Can anybody tell me if echo cancellation is really needed even at a T1 PRI to the telco? Because people keep saying when they deploy voip solution in Hong Kong using T1 PRI, there is no need of echo cancellation.

This may be commonly known but I haven't come across it so here goes, maybe it'll help someone: I have terrible echo with asterisk 1.2, zaptel 1.2.12, and a TDM400P with 1 FXS and two FXO modules. The Mark2 echo canceller with Aggressive turned on was the only setting that would make it acceptable. I found fxotune with this zaptel version to be broken. I pulled the latest fxotune.c

Just updated from CVS 12-23-03 to tarbal 0.7.1. Identical settings in zconfig.h for echo cancellation (MARK2, aggressive OFF). The echo got worse, much worse. It takes longer to train and overspeak now disables cancellation, which it did not before. In fact, I now have echo on VoIP to VoIP on our local network, which I never had before. Was something changed with the echo supression which

Hi all, I was looking to build a SIP-to-PRI gateway using Asterisk (as in my other post), but there is also an alternative of using a Cisco router with something like an NM-HDV module with a T1 VIC module and DSP channel banks. The question is, would it be more reliable to offload all dahdi/zaptel/libpri type stuff to a dedicated gateway device (Asterisk or Cisco) and have the Asterisk PBX only