similar to: FXO port on TDM400P hangs!!

Hellow everyone, here's an other newby question. I've got a * configured with the card in the subject line. At times Asterisk fails to notice a disconet from the incoming line going into one of the FXO ports. Consequently it just keeps the line off-hook for ever and that causes my provider to mark the line aut of order. Is there any way to "help" Asterisk notice the disconect?

Hello everyone. Is it possible to do some very basic voice recognition from within Asterisk's dialplan? What I'm aiming at is the ability to speak the digits I want to dial from my mobile phone. Dialing digits on my mobile phone while driving is not all that safe... Thanks for any input, Cosmin Prund

Googling arround I found a number of pocket pc softphones. Of those I was only able to install SJ-something (removed it). Is there one (pocket pc softphone) that works? Thanks, Cosmin Prund

There's a large number of gadgets one can buy that work with Skype through the API. One of the things I'm interested right now is the ability to properly use a mobile phone headset with a SIP/IAX softphone. Is there an softphone that emulates the Skype API? Are there legal implications in writing an softphone that emulates the Skype API? Should I just give up and buy a Siemens DECT

Hello everyone. I'm working on an application that needs to automatically send faxes. To send the faxes I create .call files but the .call files mostly fail because my lines are always congested within business hours! Is there any trick I can use to give the end user a better chance at actually receiving the faxes? I already tried using the local channel for dialing (so I can put in

hi, i was able to get this working with google talk. i entered myusername@gmail.com using the gtalk2voip.com website's "invite" box, and as a result, saw a request from service@gtalk2voip.com to be added as a buddy in my google talk contact list. i accepted the request. in my asterisk dialplan, i have this entry... exten => 3501, 1,

As usual, this is most likely a easy question, but here it goes any way: How can I change the caller id on a transferred call so the called party knows the call has been transferred from a colleague and it's not coming directly from our outside lines? The story goes like this: 1) Client calls. All phones ring. 2) Someone picks up the phone. 3) The phone gets transferred to someone. 4) The

Hello everyone. I've got this really annoying HFC Cologne card (or however I should call it - a single channel ISDN card based on the HFC chipset). It wrongfully detects lots and lots and lots of incoming DTMFs, to the point the card is not usable. Here's a sample out of CLI: P[ 1] I IND :DTMF_TONE oad:206361 dad:520101 P[ 1] --> mode:TE cause:16 ocause:16 rad: cad: P[ 1] -->

As the subject line says: Is PING a good indicator of network latency? If not, how can I measure latency? Thanks, Cosmin Prund

Hi, Occassionally Asterisk will go down and I have to restart it.. not often.. but sometimes. When it does the manager interface stops working, as does the CLI. My thoughts was to poll the manager interface once every 5 minutes for a value. If I don't get the value back then alert me that the server is possibly down. Does anyone know what a good value to poll for might be? I was thinking

Hello everyone. I need a BRI ISDN card that works in Romania. I already have one of the "Cologne HFC-S" PCI cards and it doesn't work right, it's junk. I get waaaay too much echo using it. I'm now "shopping" for a better card. Can anyone recommend me a card that "fits" the following: (a) Costs less then $1000 / 750 euro (b) Has one or (preferably) two

Hello everyone. This is an other question from a relatively newbie. I'd like to provide auto callback ability for my *. From my mobile I want to be able to call a number on the * and have it call me back on my mobile. I know how to generate a .call file from a script and I know how to call a script from the dialplan (in order to get the .call file generated). I also found the scripts on

Hello everyone, this is my first post to the list, so hello again. We're a small company in Romania and we're trying to set up a really small version of "call center". That is, we want to get a few land-lines from our telco in different countys and "bridge" all calls to our HQ, in order to make it cheeper for our clients to call us. Unfortunatelly there's no ISP

Hello everyone: using chan_capi 0.7 and asterisk 1.4 Quick question: How can I detect the number of "voice" channels (B channels) in use at a given time. I'd like to call "Busy" if two B channels are used on my BRI to give the calling customer a Busy signal. Long question: On my single-line BRI (two channels) I'd like to give the 3rd simultaneous incoming call an

Hello: I got into a trap. As far as I know I do not need to pay any royalties to use G.729b in Romania, so I should have used other drivers. The installation procedure looked difficult so I decided to get one from Digium - it's not that expensive, my time is much more expensive. Made the payment, got they key, downloaded and copied everything as in

I've got a brand new Eicon Diva Server BRI card and I want to configure it with Asterisk. I managed to get asterisk and zaptel to compile and install, I've compiled and installed the drivers for the Diva card and now I need to compile and install the chan_driver for chan_capi. Unfortunately this fails miserably. I get the following messages: I'm using: Kernel 2.6.16.37.4,

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

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'm unable to get faxes working over tdm400p (4fxs modules) Too many errors sending and receiving faxes with an analog fax 1) echocancel=no on the zap channels 2) ztmonitored the channel for a good/low audio volume I'm trying to send fax between zap fxs channels. No way to get it working right Has someone else the same problem?

The way I understand things, there's no way for a analog line to "reject" a call (give the caller an busy tone) if the line is not actually busy. Would a digital EURO-ISDN line give me this ability? Thanks, Cosmin Prund