similar to: echo cancellation: the never-ending quest fortruth

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

Folks, I've been following recent discussions regarding echo and echo training with much interest, since it's a problem we've never been able to eliminate here. We're facing two challenges presently, and they may be related (or not): a) Cisco 7960s in the office here echo back to our staff, but the customer hears decent sound. b) We've yet to be able to pass faxes through

Hi, I am using SJPhone here for testing ivr with Asterisk. I haven't seen any problem here yet. I have tried different things for that and finally got it working. I am not an expert to explain more about that, but here is the general section form my sip.conf. dont know whether it will help... [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ;

Folks, I'm experimenting with bringing multiple (8) analog lines from our local telco into a Carrier Access ADIT 600 channel bank with an FXO module, then having this talk to Asterisk via the T1 TDM controller on the ADIT and a TE405P card. I don't know if this will work well (ie: give me decent echo cancellation, call disconnect supervision, caller ID etc) but I haven't had much

Folks, We have several people using SIP softphones in the office. When they leave for the day, they power down their workstations, causing their registration with Asterix to quickly timeout. Here's the entry for one such extension in extensions.conf: exten => 8102,1,Dial(SIP/someone,20) exten => 8102,2,Voicemail(u8102) exten => 8102,3,Hangup exten => 8102,102,Voicemail(b8102)

What kind of channelbank/FXS port are you connecting to? I've seen problems connecting to some of the older versions of the Adtran Total Access 750's. I wouldn't doubt there would be problems on other channelbanks with older firmwares. Of course, no firmware on CAC AB1's I have the AAstra 480, Adtran 750 Channelbank (updated firmware), T100P card, and it worked fine on the

> -----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: >

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

T.38 is often put forward as the solution for reliable FAX over VOIP. Just wondering for anyone using T.38 (with any equipment), how well does it work as compared to a FAX PSTN call? Thanks. Jim James H. Thompson jht@lj.net -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041129/3ea08b2c/attachment.htm

Hello, I have spandsp setup to accept incoming faxes and receiving tif files via Email. Using tiff2pdf, or tiff2ps -a2 or even tiffsplit, the last page of the fax is cut off and the quality of the text looks "squished". I "figure" it's a tiff parsing thing, as opposed to a problem with my spandsp installation (heh). Has anyone experienced the same thing, or can

I have been doing some research on echoes and echo cancellation. Here is a summary of what I've learned: 1) When solving echo problems it is extremely important to locate the source of the echo, and if possible, eliminate it at the source. The most common cause of echo is an unbalanced hybrid at the PSTN to TDM interface. This is where the 2-wire POTS line is divided into a 4-wire

Hello, I'm a newbie user of Asterisk, i'm sucessfully install it it's great but, i get some problems with echo in a adsl line. My system is a TDM440P with 3 FXO Ports and 1 FXS. Asterisk 1.2.13 Zaptel 1.2.11 Line 1 - Analog Line 2 - Analog with ADSL It's installed with two analog lines one of them have a ADSL splitter. The location are Spain, i load the tdm driver with opermode

What are your experiences with faxing through Asterisk to the PSTN? We are using g.711u as a codec, and are originating/terminating with Broadvox as well as through our own PSTN gateways. We have had some luck with incoming faxes coming into our network from Broadvox DIDs. They work 50% of the time. Not sure yet on PSTN incoming since nobody that is using FAX is in our local rate centers.

I have a gentoo 2.6.14 box with 4 nics, LAN/DMZ/PUB1/PUB2 LAN and DMZ have a 1918 /22 each, PUB1 and PUB2 have a /29 each of which 5 ips are assigned. Using the mangle table, I give all packets a mark (according to local policies) in the range 1-10. Using ip rule, i pass marks 1-5 through the pub1 route table, and marks 6-10 through the pub2 routing table. Using the nat table, I SNAT to one

I just installed the new version of SJPhone and it appears that it cannot work with * anymore? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040111/25c910bb/attachment.htm

We're having some difficulty connecting a T1 fax board card to asterisk, with E&M Wink. We've successfully connected it to other E&M Wink T1s, but have been unable to get asterisk to hear it's dialing. Asterisk can dial down to it with no problem, and it hears all the digits, but whenever it sends digits down to asterisk, * always only "gets" the first one. This is

Hi, I would like to set up a high density FAX/PBX server, am looking at using Eicon Diva E1 card with Asterisk and Hylafax sharing channels, is this possible. I know Extensions can be reserved for voice OR fax with the combination of chan_capi used for * voice, capi4hylafax on fax but then channels must be dedicated rather than shared ie. would like to be able to answer voice/fax on same DID.

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

I'm trying to get a Carrier Access Corp. Channel Bank I working with a Digium T100P without success. What is stranger is that the status lights on the channel bank and T100P seem to change almost each time I power cycle the channel bank or reset the T100P. The channel bank has three status lights: T1, Framing, Status. T1 is green, Status is yellow, and Framing is usually red but sometime