similar to: Answer Supervision problem

Hi. I am using Digium TDM2400PFXO with Asterisk1.6. When call sets in the box , it answers the call even the phone is not picked. ideally it should answer the call when the phone is picked up. Its charging the clients. Please let me know how can I cover this ? Thanks in advance. -------------- next part -------------- An HTML attachment was scrubbed... URL:

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, 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

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

I have an older Newbridge Mainstreet 3624 upon which I'm terminating some analog DID lines. They are effectively loop-start trunks with battery supplied by me (ie. FXO) and consumed by the serving central office. One major part of DID is the requirement for providing Answer Supervision in the form of battery polarity reversal on the analog trunks. Without it wierd things start happening, like

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

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Hi! I'm using FastAGI (agi://) to make some calls. To do the dialing i use "EXEC DIAL Zap/g1/XXXXXXXX...". But how can I make "answer supervision" with FastAGI? DIAL command won't return until call is finished. Thanks in advance! -- Daniel

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

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

Hi, Is there any way of telling Asteirsk to withhold answer subversion on a call till I call Answer. My DP looks like this: [incoming] Exten => 18005551212,1,Noop() same => n,Answer same => n,Mset(__uid=${SIPCALLID}) same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV) same => n,Dial(Local/1 at dial_call_center/n&Local/2 at dial_call_center /n&Local/3 at

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.

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 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