similar to: Static on ZAP channels

Hi All I've got an asterisk system, using a couple of Xorcom Astribanks to provide FXS ports. (I'm using the zaptel 1.2 branch, if that matters) I've noticed that the ringing volume is a lot louder than on our old phone system, and people are starting to complain it's too loud. (This is the noise the phone makes when it rings, not the noise in your handset when you ring

Hi, this is probably a stupid question but can someone point me to where I can find the pinouts for the connector at the back of the TDM2400P card from Digium. I am assuming this is some kind of telecomms standard but I need to connect to 8 existing analogue lines and therefor need to wire this connector to a patch panel or R11 jacks. Also most of the docs do not mention this card - am I

I upgraded from 1.0.9 to 1.2.1. In 1.0.9 everything worked perfect. Now, I call in my IVR, and after navigating in menus when I get dialtone for dialing extension, Sound is choppy and I get bunch of messagess: -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence suppression is disabled (option_silence_suppression=0 chan->timingfd=30) -- Silence

I am having issues with a TDM2400P. It appears when the ZAP channel dials out, it randomly chops the first digit off of the number. I have tried relaxdtmf=yes, turning up and down the txgain, turned off and on the echo cancellation, generated new zaptel (with updated spinlock.h)... I am at a loss. Can someone please offer some help? Thanks. TJ

Hi i have a tdm2400p and installed asterisk 1.4.11 with zaptel 1.4.5 im having an error message when in running asterisk with the tdm card in. here's the error from the console of asterisk: [Nov 18 10:30:44] ERROR[5557]: chan_zap.c:7489 mkintf: Unable to get span status: Inappropriate ioctl for device [Nov 18 10:30:44] ERROR[5557]: chan_zap.c:10466 build_channels: Unable to register channel

HI, im having problem with callerid. Im using tdm2400P and i get this from asterisk logs -- Starting simple switch on 'Zap/4-1' [Mar 24 02:07:48] ERROR[2358]: callerid.c:564 callerid_feed: fsk_serie made mylen < 0 (-1) [Mar 24 02:07:48] WARNING[2358]: chan_zap.c:6416 ss_thread: CallerID feed failed: Success [Mar 24 02:07:48] WARNING[2358]: chan_zap.c:6516 ss_thread: CallerID

Hi, I'd like to change 3 TDM400p with one TDM2400p to avoid echo (my intention is to use a TDM2400P echo cancel module). It TDM2400p working good with asterisk 1.2.1? Or I need to install a new asterisk version? TIA Giorgio Incantalupo

Hi, during gradual migration to Asterisk, I put Asterisk in front of a legacy Alcatel PBX: PRI PSTN <--> Asterisk <--> E1 cable <--> Alcatel PBX After successful deployment of VoIP phones, it's time to drop Alcatel PBX! I'd like to keep some of analog lines to support modem, fax and some older stuff. What's the best choice? A channel bank or a TDM2400P card? Can I

Hi, I am getting half channel audio on cisco 7960? Any idea why? Details below. Jerry This same phone has been working for MONTHS using a TDM2400p with no issues. Today we got a T1 installed coming into Box A with all incoming calls going to Box B (the TDM2400P box). The TDM2400 is no longer in use. All incoming and outgoing calls are done with the T1 and Box A and Box B use SIP between them

Hey Guys I've been browsing the list looking for more information on asterisk behavior for large system. As for now I've got a project with 300 SIP Extensions to start, with future growth (scalability) Capability of recording all extensions simultaneously during peak time. And keeping the call recording for 30 days. SIP Calls being terminated over a Cisco 5400 Gateway Extra 100(+) FXO

Hi all, I have TDM2406E with 24FXO ports connecting to 10 POTS line sitting in my office. the out going calls symptom like when called party pickup the phone but the calling party still hearing the ring tone from the IP phone. Please light me up. it been many sleepless night by googling around trying to get the right answers. The digium card running on Intel 915G chipset. Below are my zaptel

I've been asked to assess the cost of implementing Asterisk with a single T1 line in one of our offices. I've had plenty of experience w/ TDM400 cards, but T1 is new for me so a couple of questions: 1) Will I need a digital or analogue interface card? I expect digital is the answer, but the Digium web site said something about analogue cards being able to support "provider T1

Hi, I have a box with Debian Sarge, Asterisk 1.2.1 (and zaptel 1.2.1) and a TDM2400P with echo canceller. I installed the card but no echo cancellation is being made...seems like the echo canceller module does not work, infact the software cancellation is working. My zapata.conf has echocancel = 128 and echocancelwhenbridged = yes but no echotraining parameter which gives a warning. I found

Hello, Polycom ip soundpoint support ACD login/logout . Can we configure asterisk with polycom ACD support? Regards Harry ___________________________________________________________________________ Nouveau : t?l?phonez moins cher avec Yahoo! Messenger ! D?couvez les tarifs exceptionnels pour appeler la France et l'international. T?l?chargez sur http://fr.messenger.yahoo.com

A big G'day to everybody on the Asterisk list. I am having a lot of trouble getting the TDM2400P card working in asterisk. I will give the important details below, please let me know if I am doing anything obvious or ideas for debugging this. SUMMARY: I get the right voltages on the line with the phone on or off the hook, but no dial tone, no ringing in or ringing out. INSTALLATION AND

How many TDM2400P cards can I safelly install in one PC? I'm loking for answers from whoever has a working scenario with * and a number of cards higher than one. Thx, Juan __________________________________________________________ Mensagem enviada usando o Webmail da ViaLink ver. 2.7.8

For example, how much power would a server with 3 of the TDM2400P fully loaded cards draw? We're trying to figure out what APC power we should have to achieve approx 30 minutes back-up time. Would appreciate any learnings or methods on calculating the power draw and determining the appropriate back-up device. Thank you. Melisa.

Hi list, I have this problem: when someone is making a call, with asterisk and a TDM2400P connected to 8 fxo lines, the sound is good, but if three, for people are calling at the same time the sound got worse and worse. Using other voip cards the sound is much better even with all user calling at the same time. What can be, the problem? Someone else has having the same

Hi, I have a problem with TDM2400P card. The card is detected ok, I can make a call but only with pulse dialing (not tone dialing) without hear sounds from the headset. When I receive a call, I can to establish a communication, but without hear sounds from the headset. When I dial any phone key, I can hear dtmf tone. I'm using Elastix 1.5.2. These are my configuration files:

We just bougth a tdm2400p with all the modules for FXO, but we are having some troubles with the card, cause it aparently is stripping some digits from the dialed number, we tested the same server with a tdm400 and everything worked as expected. We?ve already added "w" before the dialed number with no results, is there any way to solve, is it a bug thanks