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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

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

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

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

...t;fits" 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

...break; case ''exception'': ... break; default: ... break; } I used jsLint to clean the code and right now I''m clueless so any hint would be hugely appreciated. Thank you ------------------------------------------------------------------------------------------- Cosmin Cimpoi Web Developer - Team Technology (OT) ------------------------------------------------------------------------------------------- monogroup Srl piata unirii nr 4-5 400013, Cluj-Napoca, Romania tel. (+40) 264594667 fax. (+40) 264594667 mail. cosmin.cimpoi-qZ0t5ilwAP73oGB3hsPCZA@public.gmane....

...ommand(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

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 cmake script in > tools/llvm-shlib was set to...

...erly 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 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

...it a wide 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

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

...s that 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

...y behalf) 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

...ption, 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 age...

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

...set up 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=&qu...

...IX=/usr/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

Hi, Trying to build llvm.dll (8.0.0) with mingw64 (gcc 8.1.0) using cmake. After linking for hours and consuming some 7GB of RAM, ld.exe finally bails out with: X:/tools/mingw64/bin/../lib/gcc/x86_64-w64-mingw32/8.1.0/../../../../x86_64-w64-mingw32/bin/ld.exe: Error: export ordinal too large: 283482 The command I used is: cmake \ -DLLVM_TARGETS_TO_BUILD=X86 \ -DCMAKE_BUILD_TYPE=Debug \

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