similar to: Callback auto dialing

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

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

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, 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. 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

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: 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

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,

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

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

My Trendnet 26 port managed switch gave up on me so I'm shopping for a new switch. I learned the hard way NOT to trust marketing material from anyone so now I'm asking the list: what am I looking for in a managed, VoIP switch? P.S: For those that don't understand WHY I can't trust marketing material, let me tell you something about the Trendnet switch that's fast becoming

Hello everyone! I've got a Eicon Diva Server BRI card into my "*" box working just fine, but I wander if there's anything I can do to improve voice quality for my operators. I'm thinking something along the lines of "auto gain" and sudden noise suppression (like when you hit a fax machine or the other party accidently touches the dial pad on the phone). Does

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

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,

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 know this one must be easy but I'm an newbye so please help. In my extensions.conf I want to have a line like: Exten => 9XXXXXX,1,Dial(Zap/4/${SOMETHING},40,r) Ie: I want to dial all the XXX-es, but not the 9; How do I do that? What do I write in place of ${SOMETHING}? Navigating the wiki didn't provide any usefull advice... Thanks.

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

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. 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] -->