similar to: Stability of Asterisk with 2 x TDM400P cards (6 analogue lines)

Hello everyone. This is a message I've sent before on Sunday, no one replied so I'm reposting it (guess not everyone's at work 7/7) I've got this really annoying and beyond-my-knowledge-to-debug problem. The line connected to my FXO port gets marked "out of order" by my telco operator. I don't know how to explain this further. If I dial my own number from a

I have a TDM400P with some FXO ports, and I wanted to connect the two POTS lines from my Pipeline-75 ISDN router into the FXO interfaces on my Asterisk server. Hooked it up, seemed fine, called in and it answered. The problem is when the call is hung up on, the FXO port never drops. So of course then the P75 just holds the line off hook and you get a busy. So it's good for the first

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I need to ask, to refresh, is the aux power connector on the TDM400P card *only* to power the ringer on any analog phones/devices on the system? Can I still use this board, to "terminate" POTS lines and use all SIP Phones? Due to circumstances, I end up with a 1u server that has no aux power connectors available. I "have to" use this server, so am considering abandoning

I have a TDM400p with 3 FXS mods and 1 FXO mod. I have all set up with what seems to be correct settings (according to digium and asterisk wiki). As soon as I plug in my POTS line into FXO mod the line goes into offhook state (whether I have power to the card or not). Should this happen? When I try to call * box all I get is busy signal. I've installed stable version, cvs version, change

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

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

Hi; I'm trying to connect a TDM400P with an FXS module to a Valcom V-9940 Paging adaptor. This port on the TDM400P was connected to a 2500 Set and was working I just re-connected it to the Valcom (which is known to work on a Telco POTS line) and its not picking up. The Valcom docs say it need a minimum of 75 Volts at 20-30 Hz to recognize a call... So the question is what ring voltage

Dear users. My current setup uses a euroISDN E1 with 8 cahnnels for incoming and outgoing calls from a digium te110p. Currently all phones use SIP. However, I need to add some faxes lines and some POS credit card machines. These will require POTS lines with a fixed DDI. I have purchased the tdm400p and 4 FXS modules. My problem is with the zaptel.conf and zapata.conf. I am a little confused as

What are peoples experience with the reliability of the TDM400p. Specifically in the 2 FXO, 2 FXS configuration, which is the 022 (?) model. Is this board prone to random failures? joe a.

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!

Does anyone have trouble with dialing in to an Asterisk Server and having the DTMF digits recognized? We have some clients who are calling in with cell phones, notably those with SprintPCS service, who's DTMF is just never recognized. I have tried relax_dtmf on and off, with no improvement. My rxgain is currently set to 3. Can anyone suggest possible solutions? Incoming calls are

As best i could figure it out, I've installed dahdi and rc4. My TDM400P doesn't answer fxo or fxs. /etc/dahdi/system.conf: loadzone = us defaultzone=us fxoks=1,2 fxsks=4 /etc/asterisk/chan_dahdi.conf: [house-phones] context=internal ; Uses the [internal] context in extensions.conf signalling=fxo_ks ; fxo_ks Use FXO signalling for an FXS chanel dahdichan => 1 ;

Okay everyone - I'm moving away from using sipura 841 phones. I'm starting to test with Polycom IP 501 phones. We plan to upgrade our server to a dual t1 but, for now we have a digium tdm400P with 4 analog lines coming into it. So my question is will upgrading the IP phones with my existing digium tdm400 card be enough to satisfy my users ? or is it really a combo deal needing to

I've spent the last week or so troubleshooting echo problems at my Wife's business, and I've been able to clear up about 99% of the echo, but there is still a little residual echo that I can't seem to "tweak out". The users describe it as "buzzing or crackling", but what it sounds like to me is a slight echo, but just of one syllable of the word.

Hey all - we have two TDM400P cards in an SMP Redhat9 box, with 4 FXO ports each running thel latest Asterisk CVS. Users connect to this to share POTS lines using analog phones connected to Sipura SPA-2000 boxes. All works reasonably well, except every day (or more) a line will get "locked up". No other way to describe it really -- the line just stops working, so if someone tries

Shane Wegner wrote: > Hello all, > > I have two handsets connected to FXS ports on a TDM400P, > both GE models but one rings and the other does not. The > phone models are not identical. The phone which doesn't > ring on the TDM does ring when connected to a regular POTS > line and I tried connecting another phone to the port and > it rings fine. Do you have the

I'm a complete Asterisk novice and have an installation based on the Asterisk@HomeCD. I've installed my TDM400P with 2 x FXO & 2 x FXS, but every time I try to dial out, I get a message "No circuits available". Can someone confirm the pinouts for connecting the FXO's to a UK BT Line - I have RJ11 connectors on the back of my TDM400P card, so ideally I'd like to

On an Asterisk 1.2.12.1 system using a TDM400P with two FXO and two FXS modules, servicing two POTS lines: When dialing a number, such as a bank, or pharmacy, where it is required to enter a long series of numbers via the phone's keypad, an unexpected hangup occurs. The hangup does *not* occur when entering the numbers, as one might initially expect, but after the called end begins to

Hi, after finally getting my testsystem to work well I'm planning to buy some hardware. Now I was wondering what the advantages / disadvantages of using a Wildcard TDM400P solution over using real sip phones like the snoms is? The only thing I came up with is that the Snoms can use a LDAP adressbook on the server. What about calling a sip adress like person@server.com. Can this be done by