similar to: Sangoma A200D analog card with fxo's

Hi all, I've been having problems with my A20002D lately - callers from the PSTN don't hear me when I answer, but I hear them. Disabling echo cancellation in zapata.conf brings the audio (and echo) back. This used to work fine, until two days ago. The only weird thing in the logs is this: May 12 07:42:53 steerpike wan_ecd: wp1ec: The H100 slave has lost its framing on the bus! May

Been using a few different FXO interfaces (X100P, Voicetronix, and SPA-3000) at a couple sites and they've all run into nasty issues at times. I'm looking at moving one of my offices and am in the market for a solution for connecting 4 FXOs. I got excited about recent offerings from Digium and Sangoma in the way of new cards but I hear they're causing troubles for more than a few.

hi list , is having problems when sending a fax with hylafax and a card sangoma A200D, when he sends it arrives to the destination but it paginates appears in white this is my log Aug 20 16:11:08 voz FaxSend[6715]: MODEM Supports 40 ms, 20 ms/scanline Aug 20 16:11:08 voz FaxSend[6715]: MODEM Supports 40 ms/scanline Aug 20 16:11:08 voz FaxSend[6715]: MODEM WWW.SOFT-SWITCH.ORG spandsp/ Aug 20

I would like to hear from anyone good or bad as what their experience has been in recent times with STABILITY of current builds of Asterisk and drivers for TDM400P. The sort of configuration is: 6 incoming POTS lines. ie. 2 TDM400P cards. I am not concerned with: price points, or the advantages or disadvantages of using POTS vs ISDN technology, but simply RELIABILITY & stability of the

Hi List... I have an asterisk box with one TDM2400, it has 8 FXS's and 12 FXO's, and it also has the echo canceller... I'm running Asterisk 1.2.13 and Zaptel 1.2.12 with the Kernel 2.6.9-34.0.2.EL I'm using Polycom's 501 with the SIP 1.6.2.0041 The problem is when someone dials to or from the PSTN through the TDM2400, the voice quality is crappy...Instead of hearing:

Simple question that google hasn't helped much with (likely poor search terms) I just installed a Sangoma A200 with overall good results. Initial tests with both incoming and outgoing calls were very positive. Until I made a normal call that lasted more than 30 seconds. I setup the FXO with kewlstart signalling, and the outgoing call is never registered as 'Answered' and the dial

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

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 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,

Good Day List, I'm having some issues with echo cancel on my Asterisk box, and have done extensive reading and have gained some useful pointers from this list but have a couple of hopefully fairly simple questions. The Asterisk box is connected via 20 FXO ports on a TDM2400 with the Hardware echo cancel module. Echo cancel almost works, but the users hear what they describe as a

Hi. Am new to this concept but have been requested to add VOIP capability to a small office phone system. They currently have 4 standard analog lines running into a PBX feeding 16 phones, with all the usual features, call transfer call hold internal calls etc. would the following seem reasonable ? asterisk server:- ( what specs ) cat5 > broadband (VOIP) 4 FXO's for incoming PSTN

I own a TDM2400 board, with three FXO modules and one FXS. I'am having trouble with analog sip phones, from two different equipment. (Grandstream GXW-4024 192.168.0.105, and Audiocodes MP202), sometimes when I am calling someone, then I press flash, and then call someone else, both calls stay connected after I hang up. [Sep 29 07:18:06] VERBOSE[3218] logger.c: -- Called g2/16 [Sep 29

Hello, After solving the other "low hanging fruit" audio issues in our Asterisk PBX, we are left with occasional cases of severe echo which we have not found a solution for yet. Our system: - Asterisk 1.2.0-beta1 - TE110P on a PRI - TDM04 and TDM40, but these are unrelated to current echo issues - Fedora core 3 - Echo canceller KB1 Most calls have minimal, acceptable echo levels. But

Hi I have been having a rough time getting a Sangoma A200/Remora FXO/ FXS Analog AFT card set up properly. The main issue is that the card has four ports and as far as I can tell Asterisk is only seeing two. On the two that it recognizes the "Green" FXS ports are not green, they just are not lit. The "RED" FXO ports are indeed red, but from what I have read your not

Hi All; To overcome the echo problem, what mainly I have to do in the configuration other than the following line in the system.conf under dahdi directory? echocanceller=mg2,1-16 1) How can I know if the digium card supporting echo cancellator? 2) If I am getting a message in the consol that unable to enable the echo cancelator, then what does it means? The hardware is not supporting echo

I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is any registerable incoming volume from these lines. I've been running them at rxgain = 25 (zapata.conf) to make the audio audible, however this creates poor call quality issues

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

We are installing a number of systems with 2 TDM04B cards. Have done all the isolation to unique IRQs, etc. All inbound calls seem to work fine. However, outbound calls are hit or miss. Sometimes they work fine and other times we get a "you must first dial a 1 or 0" message back from telco when dialing out standard POTS lines. We are running AAH 1.0 which is Asterisk 1.0.7. Six

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, 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