similar to: Detecting disconnect on TDM400P with 3 FXO ports and 1 FXS port

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

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

Test <b>Html</b> code as there is no <pre> Preview </pre> button -- Posted via http://www.ruby-forum.com/.

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

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. 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

Hello! This is my first post so hello to the ROR community :) I am implementing an online editor using the prototype.js lib. On one of the ajax.request calls I make something weird happens. In the debugger I get this output: =================== 10:36:04:796 [DEBUG] FileBrowser.onServiceResult > inResult = success 10:36:04:812 [DEBUG] FileBrowser.onServiceResult > inResult = exception |

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, 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

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. 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

Just ran into the same problem, but with `-DLLVM_BUILD_TOOLS=ON` since the tools link against shlib and use the C++ interface I can't use Cosmin's solution. I managed to get through with using `RelWithDebInfo` instead of `Debug` On Sat, Jun 22, 2019 at 5:20 AM Cosmin Apreutesei via llvm-dev < llvm-dev at lists.llvm.org> wrote: > Update: looks like the the problem was that the

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

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] -->

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,

I received an additional module for my TDM400P yesterday. I installed it last night. Before I installed it, I had a working 1x1 PBX. After installing the physical hardware, I spent a couple of hours unsuccessfully attempting to get dial-tone on the new FXS part. I gave up, made sure the old FXS still rang and went to bed. I should have tested further. This afternoon, I got a call at work

Hello, I want to use Asterisk PBX in front of my old, legacy PBX. The legacy PBX can be outfitted with caller-ID and is already able to handle Calling Party Control Signal Detection (this is a Panasonic KX-TD1232 Super Hybrid PBX. My question is how would one enable Asterisk to control the TDM400P/FXS port to provide to the /FXO CO port on the legacy PBX, support for proper answer supervision/CPC

I thought I had posted this, but I didn't see it in the archives, so I guess I hadn't. I've got FXS lines going to a legacy IVR. When I Dial into one of these lines and then hang up, FXS plays the Congestion tone until the IVR drops voltage. I would like the IVR to hang up sooner. I could do this by either making the IVR recognize the standard Congestion tone, or changing the