similar to: Static on outgoing calls using either X100P or TDM400P

>Asterisk and it works fine untill the following >situation: > >- one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that >line) - both via old and new PBX. >- zap show channel <n> would show that line as 'Offhook', though no telephone is off hook. > >If physical line would be

Hi All, I'm attempting to use two X100P cards and one TDM400P card. What driver do I load for the TDM400P? Is their any documentation for this card yet? Also, I know this has come up before but I can't seem to find it in the archieve... how do I configure the second X100P card? Anyone have a working sample? Thanks in advance, Randy

Hi, I have configured tdm400p alone and it works fine for me. But now i tried to add 2 x100p in the same machine with the following configs, but asterisk refuses to load, can someone guide me what i have done wrong. Following r the configs and output. TIA, Azher >> ./asterisk -vvvvvc [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found --

I am sure that quite a lot of people would like to have Caller ID working with their X100P and TDM400P cards outside of USA. Judging from previous threads this is just a matter of implementing this support in the software driver! So, I was thinking, if we get together and put few $(USA DOLLARS) into a basket, we could then ask Digium to actually properly implement Caller ID for non USA

Hi All, Sorry if this has been around a millions times.. I have been off this list for a few months now.. I have installed the latest asterisknow (upgraded asterisk to 1.6 as well) and I am having a hard time getting my X100P and TDM400P working.. Its all new to me with dahdi because my old server is still running Asterisk 1.0.2 so there have been lost of changes.. Can someone point me in

Hi, I'm thinking of purchasing a TDM400P card and was wondering if anyone has experienced any echo issues with phones off these cards connecting to the PSTN via the X100P cards? I have had my fingers burnt with a Voip phone > X100P. Cheers, Taff. --------------------------------- ALL-NEW Yahoo! Messenger - sooooo many all-new ways to express yourself -------------- next part

I've searched the Wiki and this forum with little success. I have a TDM400P in my server which functions fine. Except it will continue ringing about 3 times after hangup. I.e. it's failing to detect the hangup tone. I was previously running a Sipura 3000 and had the same issue. After researching and some timely assistance I was able to determine the hangup tones applicable to

I'm looking to add a single FXO port to my Asterisk box. It looks like my options are a Digium Wildcard X100P off eBay for $6.99, or a Wildcard TDM400P with an FXO Module from Digium for $125. Can anyone explain the tradeoffs (other than the ability to put 4 FXO/FSO modules on the TDM400P). What about RTC for the system - I know the TDM400P provides it. Does the X100P? Thanks!

With one IVR payment system, I noticed quite a difference in DTMF transmission between these two cards. The IVR missed nearly all digits from X100P, while receiving digits from TDM fine. Since neither card process or synthesize audio, what can the difference be? (This particular IVR has problem with some regular phone devices, too.) Yuan Liu

Hi All, This must have been asked before, but despite my best googling efforts, I can't find a solution or reason why it won't work. Please accept my Apologies. Is it possible for an outgoing call on the FXO port (PSTN line) to not always register in the call logs as being answered? I've tried adding the 'r' option to the end of the Dial command, but that does not appear to

Calling out on PSTN over a TDM400P seems to generate a Red Alarm - whatever that is. I have another extension on the PSTN, and I can dial out over that. zttool shows no alarms. asterisk*CLI> zap show status Description Alarms IRQ bpviol CRC4 Fra Codi Options LBO Wildcard TDM400P REV I Board 1 OK 0 0 0 CAS Unk YEL 0 db

Hi, I had ordered a TDM40B and developers kit a few months ago. I have everything installed and working, with one exception - sound quality. When placing a call it sounds like a very bad cordless phone - lots of hiss / static in the background. This even happens with the dialtone, though it is much worse one the call is connected. This does not occur when the phone is directly connected to the

Hello. Well I finally rx'd my dev kit (new batch of TDM's apparently. I'm on Mandrake 9.1 There were no hardware install instructions, it would have been nice to know whether the 4-way power connector was to be used or was for some other future or expansion purpose. It came with a floppy disc, no label and it wasn't even write protected. The only readme file was

Hello I setup Asterisk@Home with purely VoIP and it worked fine. I then added an X100P card so I could call out / take inbound calls via PSTN and that went fine. But I have just added a TDM400P card (specifically a TDM30B) and now problems. Here is some of the output. Any ideas on what I should be looking at next? When I run genzaptelconf -s -d I get lots of erors on screen - bit I can

Hello. All. I am a bit new to asterisk, started from half a month ago. I am setting up a home asterisk server with analog card. I am using asterisk 1.4.27. At the moment, I bought a X100P card and installed it on my computer. I used it to connect my home phone line. For the moment, it works fine when dial in. Soon I noticed when I dial out through it to my mobile, it can't hang up

All of a sudden I am getting the following warning "Ring/off-hook in strange state 6 on channel1" from chan_zap.c and I cannot answer calls. I can place calls out without a problem though. Any ideas what can be the problem. I have checked zapata.conf and zaptel.conf and they both seem fine. Thanks in advance. Dan -------------- next part -------------- An HTML attachment was

Outoing calls I make using Motif Google Voice Calls continue ringing even after the other end picks up. I have to restart Asterisk to resolve the issue. I don't see any errors. It's not recognizing that the other party picked up the phone and restarting Asterisk fixes it only for a day. -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336

Hi! I currently have the X100P and TDM400P Asterisk cards. For obtaining Asterisk PBX: I have no direct connection to internet, so I use ftp and download the neweast packages: zaptel-0.7.0, libpri-0.4.0, asterisk-0.5.0. The environment I compiled the tool: PC (HP): Intel PIII 500 MHz OS: Red Hat Linux 8.0 (Upgrade from Red Hat Linux 7.1). When I compile the first package (zaptel-0.7.0), I

Dear all I have now install TE210P 2 port E1 card on asterisk 1.4.10 on centOS 5 but thing is that i have connect 1 E1 port with avaya E1 back 2 back and second E1 card on Direct Telcom for outgoing for outside now i got this error when i call on avaya PRI asterisk think PRI_CPE and remote end also CPE i have configure /etc/zaptel.conf span=1,1,0,ccs,hdb3

Skipped content of type multipart/alternative-------------- next part -------------- A non-text attachment was scrubbed... Name: zapata.conf Type: application/octet-stream Size: 590 bytes Desc: zapata.conf Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040811/dfc853bc/zapata.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: zaptel.conf