similar to: Echo problem

Hello all, I know this has been covered on list but can not find the answer I need, lots of references to no authority found, but none with an answer. I have two * servers, one behind firewall with nat the other on a dmz with nat. Both servers register with each other successfully. home is today's CVS-HEAD away is Asterisk 1.0.7 on away: Registered to '165.xxx.xxx.xxx', who sees

Hello everybody, I'm aware that I can try to balance gain levels with PSTN cards using the ztmonitor tool, as described in http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html (Adjusting the rxgain/txgain Settings). Is there a similar tool for mISDN? If not, what is your approach to gain setting in mISDN? Thanks in advance, -- Dr. Andrea Spadaccini

The value of 14800 is correct. I had issues with my TDM400p with 2x FXO's installed and using the Xlite client. I could not get the echo stable at the initial call. Changing to a hard phone made everything work correctly. I still had problems with the off location I called, but mostly worked great. To see the TX gain I created an extension with the Milliwatt command attached to it and called

Hello, We've got an Asterisk server with a Digium TE110P card, connected to a Tadiran Coral Flexicom IPX 500 switch using a T1 card. We are having echo problems on the lines coming in from the Digium card. I was wondering if anybody is successfully using a Digium card and Asterisk with a Tadiran switch, and if so whether they could share some configuration information? Thanks! ----Scott.

Here's our config: cisco 7960's running 6.3 sip code latest cvs of * t100p zaptel card adit 600 channel bank 7 pots lines and 2 fax machines on the adit 600 dialing out from the cisco phones gets sent out via the zap channels, but I'm having some serious echo problems. I currently have the adit set to +3 rxgain and -6 txgain, with my zapata.conf containing: echocancel=128

http://www.asteriskdocs.org/modules/tinycontent/index.php?id=4 Can I make a suggestion that some documentation is provided for the Tortoise CVS download of the asterisk docs. I've tried every combination and I cant get it to work. I'm assuming it must work otherwise it wouldn't have been listed but for 60 seconds more work it would be a bigger benefit to the asterisk

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

Hi, is it possible, that my * identifies himself as ip-phone? I.e. Im using a grandstrem 100 phone and if I use * as proxy, the authentification string should be changed. Im not sure where looking for this. Hfh, Oliver

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.

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

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

> -----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: >

A while ago on the download page on www.asterisk.org, there was a stable branch for the asterisk source tree. It seems to have disappeared now, at least the instructions on that web page are gone. What's the story on this? Can we have it back please? thanks __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages!

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, I get fascinated when I dial someone and get an IVR play " for accounts department press 1, for sales, press 2 or hold the line for an operator" and then have MOH play before the line is picked up at the desired extesion. Please, permit me as I know this will be one of the dumbest questions to ask in a group like this. I'll apprecaite any specific guide/instruction. Thanks

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 got my TDM11B and am trying to get it to work on my PC. However, I'm having difficulty getting the wcfxs driver to load. I've Googled this problem, and while there are others who have ran into the same problem, none of the solutions work for me. I would very much appreciate it if you could suggest things I could try to resolve this problem. Details follow. The PC is somewhat

Skipped content of type multipart/mixed-------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 186 bytes Desc: OpenPGP digital signature Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20061103/b35592e2/signature.pgp

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