similar to: TDM04B Echo on Only One Channel

Hello All, I got my TDM04B card installed and configured. Everything works fine I can receive calls and route to appropriate extensions. The only problem I am facing is Slowness. When I dial the PSTN number which is connected to Zap 1-1 after two ring it answers and then run the AGI script. What I did was assign it to a specific extension. So all inbound call on that PSTN number should

Hi, I'm in the process of setting up Asterisk in a SOHO environment using ISDN for trunking. More specifically a BRI 2B+D circuit where one SPID is used for the business and the other is used for personal. The circuit already exists, but is presently being interfaced to POTS phones via a TA. This configuration is not very common in the US, but we are fortunate that our LEC offers it price

Hello, I'm interested in implementing DID functionality with the Digium TDM04B adapter. Is DID supported with POTS? Are there any caveeats or drawbacks that I should be aware before proceeding? This pbx will be implemented in eastern europe. Thanks in advance. -- Martin Spasov <mspasov@techno-link.com>

Slightly OT, but I'm wondering if anyone here has come across a "soft ATA". That is, software that will perform the functions of a basic POTS line ATA on Linux with a zaptel driven card. I have a Linux machine with a zaptel card in it and I want to have another Linux machine running Asterisk utilize the zaptel card in the first Linux machine to make outgoing and receive

I used to get bad echo and also voice breaking up during a conversation, so I upgraded the machine (from a PII 300 to a Athlon 1.8GHz) and have resolved the voice breaking up issues but still seem to have a fairly bad echo on *both* sides of the conversation. Currently I am only using the system for outbound calls, so I am always calling from the digium USB port phone and out to the telco using a

Hello all, I am having a big issue that i just cant get fixed for a while now and its causing me lots of grief. It seems like since we got FIOS installed (including switching to fios phone lines which are supposed to be the same on our end) i am having massive problems with asterisk not hanging up dead calls for days, even weeks if i dont catch it. It slowly builds up randomly not ending a call

Hello, I've spent a couple days searching and posted into the forum with no luck, apologies to anyone who reads the Digium forums for the cross-post. I'm having a problem with an asterisk set up where I have a TDM402B connected to a POTS line. Also connected to the POTS line are plain telephones, non SIP, just plain old telephones. When one of the normal handsets goes off-hook,

We have occasional echo, mostly on lines that come across as quiet.. I am trying to troubleshoot how to disolve the echo. i have tried the steps on voip wiki, but have not recently been able to make any difference. the echo isnt horrible most of the time, and seems extremely random in that i can call a number once without echo, then dial the same number a second time and get echo. things i am

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Hello All, I saw an add in my latest Linux Journal advertising Sangoma's new "AA series" of FXO/FXS analog cards with on-board echo cancellation, but I can't find any information at all on them. Even the link given in the advertisement is a dead end as far as I can tell. Anybody else seen/heard anything about this? -Seth

Hello: Have a TE110P laying around and decided to see if I could build a router around it. I've tried compiling several versions of zaptel .1.4.x with the same results. I checked the zaptel changelog and can't find anything about it no longer being supported (or that it ever was for that matter). ztcfg: Zaptel Configuration SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1)

The TDM01B is 4 port capable but has only 1 FXO module. I'm running asterisk 1.2.7 and zaptel 1.2.5. I cannot seem to get the TDM01B working. When I do the zttool it shows 4/1/0. I can dial out from a POTS phone up to the point that the cable plugs into the card. Here is my /etc/zaptel.conf loadzone=us fxsks=1 and here is my /etc/Zapata.conf [channels] language=en #include

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Hi all, Does anyone know of any providers that can offer local numbers based in the UK via IAX or SIP? We're looking at getting a number based in the UK. Thanks! -- jeremy bogan [ jeremy@segpub.com.au ] segment publishing - design.develop.host

Asterisk 1.2.24 seems to crash repeatedly under any substantial call load (and sometimes without a substantial call load - just one SIP leg is enough to do it) when using the G.729 pre-compiled binaries from: http://asterisk.hosting.lv/ As per: http://www.voip-info.org/wiki-Asterisk+G.729+Licensing Time to crash is variable, but seems to require at least an hour of production performance

I can ping it just fine. I am on gw5.voicepulse.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040615/054c83f4/attachment.htm

We are running asterisk with a single POTS line for local calls and a voip line for long distance. Whenever we receive a call on the POTS line it is more than likely, but not always, going to have significant distracting echo. In addition to that there is occasional heavy crackle or static. I have tried to follow the guidelines at :

This is a question about removing the intercept in a binomial glm() model with categorical predictors. V&R (3rd Ed. Ch7) and Chambers & Hastie (1993) were very helpful but I wasn't sure I got all the answers. In a simplistic example suppose I want to explore how disability (3 levels, profound, severe, and mild) affects the dichotomized outcome. The glm1 model (see below) is

Hi, I recently got a TDM04B and after installing and getting asterisk up and running I connected a handset to one of the ports. Unfortunately I don't get a dial tone when I lift the handset. What could be the cause of this? Could someone point me in the direction of a proper config for a TDM04B? Thanks.

I need your help: please help test the gender detection module at 575-613-4392. I wrote a gender detection module and thought I'd try it out. It only takes a second. I've been showing 90%+ accuracy and I want to make sure it's working correctly. Rain and significant background noise seems to throw it off, so I still have a bit of work to do. Have your friends and significant others