similar to: Tired of trying to fix this echo problem

Then if let say instead of buying TDM400P cards I get this : Clipcomm CG-410 Quad FXO Gateway is it any good? They also sell Quad FXS Gateway. Clipcomm seems to sell the cheapest Quad FXO/FXS Gateways arround so I'm wondering if it's working fine with asterisk. I found this one too but at a lot higher price : AudioCodes MP108 8-Port FXO Analog Gateway (SIP) I need to buy a

I have searched for how to locate echo cancelation on SIP clients, but cant find anything and echocancel=y doesnt seem to have any effect. Configuration: CVS-HEAD from last month iPAQ h5500 with SJPhone (gsm/ulaw/alaw) Problem description: When I place or receive a call I hear a faint delayed echo of myself. The other party hears a really bad nonmuted echo that makes the call unusable. Aside

I already have one Digium TDM04B installed in my server working fine. I just received 2 more so I did added them to my server. When I booted again linux told me that it found new hardware I said ignore. Then I log in as usual. I did ztcfg -vv to see if it sees 12 channels now instead of 4 but it only see 4. So then I did again modprobe zaptel, then modprobe wcfxo and wcfxs hoping it would see

I am having a problem with a couple servers. They both have a Digium TE110P and a TDM04B card. I have setup the system so the TE110P uses channels 1-31 and the TDM04B 32-35. The problem is that when we reboot the server sometimes the TDM04B is recognized first and the TE110P second so the configuration fails. I do not know why this happens and to solve this I have to do a "service

I have a customer running Asterisk 1.2.13, Zaptel 1.2.11 with a TE110P, a TDM04B and an Astribank-32. They have been complaining that there is echo on calls even when they are IP to IP on the same network. There are 18 Aastra 9133i phones and 30 analog phones connected to the Astribank. I can understand there being a bit of echo on the analog phones, but I do not understand why there would be

Hi, A week or so ago I wrote about the problems I was having using a Digium TDM04B card in a Dell PowerEdge 750 IU running Fedora Core 1. Digium steadfastly indicates their card won't work in any PowerEdge 650, 700 and 750 series machines "do (sic) to a failure in interrupt handling between the pci bus and the card." My question to the list is: for those of you using a TDM04B

Hi people, I would like to read your suggestions as to where the issue might be. ATA286=Grandstream Budgetone ATA 286 in SIP mode. One Lan Port and one FXS port. TDM04B= 4 FXO signal fxls There is a 8FXO-to-SIP unit in this scenario that works perfectly so i will not make mention of it. PSTN----VOIPprovider---Internet---ATA286------tdm04b---Asterisk1.2.-13 Asterisk is being used as a meetme

Hi, I am just trying to figure out how to setup a TDM04B card for incoming/outgoing calls. I have 4 lines, which are provided as a rotary trunk group, currently hooked into a Nortel system, which asterisk will replace. I have setup a Dell 1800 (Tower) system with the TDM04B card, which seems to work. The question is how do I set it up that all 4 lines are part of a trunk group, such that

>?That beeing said I think the main task of scripts is to get things done via running them end to end in a fresh session. Now, it very well may happen that a lot of stuff has to be done. Than splitting up scripts into subscripts and sourcing them from a meta script is a straightforward solution. It might also be that some functionality is put into functions to be reused in other places. This

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 have a client who wants a Meetme box with 12 FXO ports, to connect to Analogue lines coming from an Ericsson PBX. It looks like I could do this with four different hardware configurations: a) three TDM04B cards (based on TDM400P) b) one TDM04B and one TDM808B c) one TDM804B (or TDM854B?) and one TDP808B d) one TDM2403B (half filled TDM2400P) Apart from considerations of cost and PCI slot

Hello I am (attempting) to run the astlinux version of Asterisk on a VIA embedded platform. I have a TDM04B and pretty sure zaptel.conf and zapata.conf setup OK. They worked fine with same card in traditional PC anyway. I think need the module wcfxs for a Digium TDM04B card. Is this module not part of astlinux? Do I need to download it? Or is it in opt? I see wctdm - but think that is

In the next week or so, we'll be turning up a new T1 Pri from Cox Cable in Omaha using a T100P (installed but not yet configured). While discussing the interface parameters with a very knowledgable Cox engineer, we decided on some of the basics including b8zs, esf, dchan=24, callerid, 5 digits of called number forwarded to *, and switchtype=national (NI2). They indicated their CO switch is a

I would like to install a TDM04B with Asterisk@Home 2.4 and Asterisk@Home 1.5 but I didn't find documentation about this installation. Thanks in advance, Nelson -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060206/85ff5252/attachment.htm

Hi We have static noise problem on our asterisk server. latest stable release. The card is a new TDM04B We have it installed on the following hardware Motherboard Intel SE7520BD2SCSI 2x POWER SUPPLY 730W INTEL I will not mention the other hardware because we have desactivated/changed all the other items The only 2 items that we have not changed is the mobo and the power supply. At first it

Ok I know I'm not the only one having echo problem with asterisk but the weird thing is that when I receive a call from a PSTN line on my TDM04B card I don't have any echo problem at the beginning of the call then after a few minutes I start having echo on my side only (the person calling from a regular phone doesn't have any echo), then it stop and come back all the way until the

Hi, Some weird problem (or I'm too sleepy) happens with a tdm04B revision G (4fxo) Steps: modprobe zaptel modprobe wctdm ztcfg -vv /etc/zaptel.conf fxsls=1-4 # TDM04B defaultzone=us loadzone=us /etc/asterisk/zapata.conf signalling=fxs_ls group=1 context=incoming channel => 1-4 modprobe zaptel and wctdm load fine, however ztcfg -vv shows: 0 channels configured Im using centos 4.4 with

Hello, I am looking for 4 port FXO low profile PCI card that could be used with Asterisk. Digium TDM04B sound like a good choice but it is half high PCI card and I can not plug it in my Dell box (small box). I am looking for adequate low profile PCI card (55mm high or similar but definitely smaller than TDM04B so I can plug it in). Does anyone know where to search for it? Thank you in advance,

I'm wondering if there's a way it will show on the phone when there's a new message. Here's what I'm trying to do : in my extensions.conf when someone call from a PSTN line on my TDM04B card they have a choice. When someone press 1 for sales then I have 3 phones ringing at the same time. Each phone as already there own mailbox because if someone know there extension

Hi, I've been lurking here for about a month and I've been putting together our companies planned migration to a new office and a new phone system. Could anybody tell me if there are any pitfalls to the following setup I may have missed? 1: TBD PC running * a: Digium TDM04b FXO connected to PSTN 1: lines rolling over via SBC sentrex to get multi-lines on 1 number b: Digium