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

...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

...quot; the following: (a) Costs less then $1000 / 750 euro (b) Has one or (preferably) two ISDN S0 interfaces. (c) Easy to set up. (d) Drivers offer proper echo-canceling OR has an hardware echo canceler. I might increase the $1000 a bit if I can get good hardware echo canceler... Thanks, 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

...sendfax) - and that command requires an chan_capi channel, it doesn't like the "local" channel. Besides, looping in the dialplan would probably interfere with the "Wait" option in the .call file so that's a really bad solution. -- Thanks for any suggestion, Cosmin Prund -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071015/692e5548/attachment.htm

...e 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 phone that supports a bluetooth headset? -- Thanks, Cosmin Prund -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071018/73bdccee/attachment.htm

...call files are designed to connect one channel to an other channel or one channel to an application. But I don't want to connect to an application like Voicemail, I want the system to behave as if I called the other way around and ended up into an arbitrary context. Thanks for any help, Cosmin Prund, Romania

...f) has an number of benefits: (a) I don't need to talk to the Telco (b) I *know* who called and I can call them back and (c) In a distant future I might use the capi channel's ability to transfer the call to a different POTS line since this doesn't use the B channel. Thanks, Cosmin Prund

...the agent would hear music, immediately followed by two tones, and then they would be bridged to the client. Perhaps you're running MeetMe() with those join tones disabled? Check out the docs for MeetMe. I think it's option capital i, as in Iberia. On Mon, Jun 23, 2008 at 12:19 AM, Cosmin Prund <cosmin.prund at adicomsoft.ro> wrote: > Hello. It's been a while since I last posted (probably because my "*" works > just fine). I'm working on something to replace call queues in my own > application-specific way and I'm using MeetMe rooms to bridge agents an...

...de enough connection for 1 simultaneous conversation, using IAX protocol with the comercial version of the g729 codec? I'm expecting this to be engough for more then 1 conversation (after all a single line analog connection is rated at 64kbit and I'm getting double that bandwidth) Cosmin Prund

.../lib/asterisk ASTERISK_HEADER_DIR=/usr/src/asterisk-1.4.0/include MODULES_DIR=/usr/lib/asterisk/modules CONFIG_DIR=/etc/asterisk //---------------------------------------------------------------------------------- If anyone has any idea what I'm doing wrong, please help me, Thanks, Cosmin Prund

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

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

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

...naged" and "Virtual LAN" in the biggest possible letters. Later, after buying two Intel 1Gb Virtual Lan Enabled network cards, I discovered my Trendnet switch doesn't do standard VLan, it only does VLan if linked to an other Trendnet switch - not useful at all! Thanks, Cosmin Prund

...g 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 one of Asterisk, chan_capi or the Diva driver have support for such functionality? -- Thanks, Cosmin Prund -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070419/77dac787/attachment.htm

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

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

...a call to googletalk using the chan_gtalk module . i am inside a NAT-ed LAN, and audio works in one direction only for the asterisk (SIP) - gtalk call. anyone else got asterisk - googletalk using chan_gtalk working? > >Message: 10 >Date: Fri, 02 Mar 2007 19:07:41 +0200 >From: Cosmin Prund <cosmin@adicomsoft.ro> >Subject: Re: [asterisk-users] gtalktovoip and Asteirsk >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <45E859DD.8080103@adicomsoft.ro> >Content-Type: text/plain; charset="iso...

Hi all !!! I would like to install asterisk in Xen domU using TDM400 hardware. Somebody know a howto or tutorial about that ? Thanks in advance roberto -- Ing. Roberto Pereyra ContenidosOnline http://www.contenidosonline.com.ar