similar to: (newby) IAX Trunk on low bandwidth connection

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

Hi All I am having a problem setting the outbound callerid number on a PRI E1 in South Africa. The outbound number keeps on appearing as the main PRI number. How does it work between Asterisk and the Telko? More importantly how do I get it working? Kind Regards Garth -- Garth van Sittert BSc (Physics & Computer Science) ----------------- Mobile: +27 (0)83 791 6662 Email:

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

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

Anybody tried one with Asterisk yet ? Views ? Best Regards, -- This message has been scanned for viruses and dangerous content and is believed to be clean. SplatNIX IT Services :: Innovation through collaboration

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

Sip to sip calls are fine, both local on Asterisk and over a SIP gateway, however some people who call on the PSTN line say we are very queit and vice versa, can the volume be turned up on the PSTN line? The volume buttons on the VoIP phones only turns up the others voice, so this is a fix for us, but how do I make our voices louder for the people on the PSTN line? Thanks. Paul.

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

Breeze to set up, too. To monitor and transfer to SIP/1000 / ext 1000: 1. Insert exten => 1000,hint,SIP/1000 into your default context (the context the extension is in) 2. In the monitoring phone's web interface, click Function Keys, pick a key, change it to Destination and type in SIP/1000. Once you submit the form it will change to a SIP URL, that's OK. 3. There is no step 3.

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

Hi, I am trying to send "404 Not found" reply, without any luck with the following: exten => 555,1,Playback(you-dialed-wrong-number,noanswer) exten => 555,n,Playback(check-number-dial-again,noanswer) exten => 555,n,Congestion() However the above results in "500 Service Unavailable" being send out. What would be the correct application/function to generate "404

Hi list! Does anyone have experiences with the updated model of the Kirk IP600? The V3 model is supposed to support SIP instead of only SCCP or H323 which would make the use with Asterisk a lot easier. I have only tested the Kirk IP600 V2 with SCCP / Skinny protocol which is still giving me severe headaches : - the standard Skinny driver in * doesn't work, only the version of Sergio

Hello, I had had some echo issues. I purchased a digium echo canceling card, and the echo issue seems to be reduced but not eliminated as I hoped it would be. I currently have it set to 128 'yes'. I've tried 256, but when I try 256 what happens is that instead of getting better echo canceling I get AWEFUL echo. Can anyone enlighten me? I am running 1.2.6 with a 4 port PRI card.

I am trying to set up a linux based faxing solution for a client, and have found that the modem they have (ancient dataplex external unit) just isn't up to the job. It talks to some remote fax machines but not others. A new external modem ranges from AUD$75 to AUD$400, which got me thinking of other possibilities... #1 FXO PCI card (more expensive for 1 port, probably cheaper for 2+) #2

In the list I found some messages that *8 doesn't work so well. Is there any possibility to create a extention that you can call, and if you are fast enough, pick up a number? (Also if you are outside your callgroup) like pseudo code: exten => 888, 1, EnterPhoneNumber() exten => 888, 2, EnterPass() exten => 888, 3, TransferCallToThisPhone() exten => 888, 103, Invalid()

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

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: 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 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