search for: messina

...is a stupid question, but would it be possible (probably alot of work if possible) to have a bandwith sharing video stream using something similar to bittorrent? Basically, bittorrent but playing the video as it's downloading. Is this possible/feasible or is it just wishful thinking? Paul Messina

25.12.2014 15:46, Anthony Messina ?????: > On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: >> I want to change call files, which has caller id in them, to call >> originate from dial plan. >> But I don't see such parameter here >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+1...

...hat starts with at least one number, but includes a literal "*" followed by 1 or more numbers would match. This is not the case, and it matches any extension that starts with a number. Thank you in advance for your assistance. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http://lists.digium.com/pipermail/asterisk...

...]# cat /proc/zaptel/1 Span 1: WCTDM/0 "Wildcard TDM400P REV I Board 1" (MASTER) 1 WCTDM/0/0 FXOKS (In use) 2 WCTDM/0/1 3 WCTDM/0/2 FXSKS (In use) 4 WCTDM/0/3 Where do I go with this? -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http://lists.digium.com/pipermail/asterisk...

Hello! I want to change call files, which has caller id in them, to call originate from dial plan. But I don't see such parameter here https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Originate How can I pass callerid to following: exten => 6003,n,Originate(SIP/6003 at asterisk,app,meetme,"6003,x") Thank you!

...gium.com You can reach the person managing the list at asterisk-users-owner at lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-users digest..." Today's Topics: 1. Re: Any 1.6 SendFAX example ? (Anthony Messina) 2. Re: force channel hangup (Anthony Messina) 3. Re: force channel hangup (Danny Nicholas) 4. Re: force channel hangup (Alex Balashov) 5. Re: force channel hangup (Danny Nicholas) 6. Re: force channel hangup (Tzafrir Cohen) 7. Re: force channel hangup (Tzafrir Cohen) 8. receiv...

...at www.voip-info.org. Do the order of the columns matter, etc.? Where do I go to find the latest Realtime table information for a specific Asterisk version? (The same questions apply to IAX Realtime.) Thank you in advance for your help. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http://lists.digium.com/pipermail/asterisk...

On Tue, Sep 16, 2008 at 2:21 AM, Michiel van Baak <michiel at vanbaak.info> wrote: > On 22:46, Mon 15 Sep 08, hugolivude wrote: >> I have two Asterisk servers running on the same LAN. One starts fine, >> but when I start the other I get: >> >> pbx_dundi.c:4582 load_module: Unable to bind to 0.0.0.0 port 4520: >> Address already in use >> >>

On Thursday, December 25, 2014 03:53:44 PM Dmitry Melekhov wrote: > 25.12.2014 15:46, Anthony Messina ?????: > On Thursday, December 25, 2014 11:48:12 AM Dmitry Melekhov wrote: > I want to change call files, which has caller id in them, to call > originate from dial plan. > But I don't see such parameter here > https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Or...

...e In Cooperation with SWIM, AAAI, ACM SIGAI, ACM SIGCHI, ACM SIGMIS. With the presence of internationally distinguished keynote speakers: Alexander Brodsky, George Mason University, United States Plamen Angelov, Lancaster University, United Kingdom Salvatore Distefano, Universit? degli Studi di Messina, Italy David Aveiro, University of Madeira / Madeira-ITI, Portugal A short list of presented papers will be selected so that revised and extended versions of these papers will be published by Springer. All papers presented at the congress venue will also be available at the SCITEPRESS Digital L...

...e In Cooperation with SWIM, AAAI, ACM SIGAI, ACM SIGCHI, ACM SIGMIS. With the presence of internationally distinguished keynote speakers: Alexander Brodsky, George Mason University, United States Plamen Angelov, Lancaster University, United Kingdom Salvatore Distefano, Universit? degli Studi di Messina, Italy David Aveiro, University of Madeira / Madeira-ITI, Portugal A short list of presented papers will be selected so that revised and extended versions of these papers will be published by Springer. All papers presented at the congress venue will also be available at the SCITEPRESS Digital L...

Hi I'm currently running asterisk with an fxo X100P and aTDM one port card in a small not world connected subnet, I've sucessfully setup two sip phone and one analog extension everything works fine with the analog phone but when you talk to someone on the sip phone the person at the sip phone can be heard with inteference. I've looked aroud the archives and found nothing specific.

...and more people will start to complain about = this. Regards, Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -----Original Message----- From: asterisk-users-bounces at lists.digium.com = [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Anthony = Messina Sent: Sunday, December 30, 2007 12:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Realtime logic in Asterisk 1.4.16.1 On Wednesday 19 December 2007 05:48:01 pm Mindaugas Kezys wrote: > -----Original Message----- > From: asterisk-users-bounces...

...able for the filename that is created by the FollowMe application when "a" is specified as an option to record the caller's name? I'd like to clean up the recorded name files after the call is complete. Thanks -Anthony -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http://lists.digium.com/pipermail/asterisk...

...ield to value defined in Dial() command exten => s,1,Set(CDR(userfield)=${ARG1}) exten => s,n,MacroExit() This always worked for me with 1.4.14 and I'm not sure how to begin to get this fixed. Any help is appreciated. Thanks. -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http://lists.digium.com/pipermail/asterisk...

...d, it is possible to do something like: Variable: CDR(accountcode)123456 Or must the variable names be "var[n]" where n is a number? I'd like to set the accountcode for a Local channel that originates a call. Thanks. -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http://lists.digium.com/pipermail/asterisk...

Hi, To check telco billing, I'm usinfg Asterisk-Stats from http://www.areski.net/asterisk-stat-v2/about.php . How can you tweak this application to display graphics and data that use Billsec instead of Duration ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

tried to get the answer on irc #samba, but people must be having a busy weekend... 1. when you do, "enable privileges = yes" and set some privileges on the pdc, are those privileges effective throughout the entire domain? 2. and then does "enable privileges = yes" need to be set on each samba member server of a domain? 3. would i need to grant the same privileges on every

I will soon be implementing a Fedora Core 4 (or 5), 2.8MHz Xeon, 1GB RAM, 2.5TB storage, gigabit ethernet server which will hold backup copies of about 200 DVDs. These DVDs will be played through a gigabit LAN to another Fedora Core 4 (or 5) workstation using xine which will then output video to the projector in my living room. I am not asking to start a competition or war on the list. My

...'/usr/lib/hotplug/firmware': No such file or directory make[1]: *** [hotplug-install] Error 1 make[1]: Leaving directory `/builddir/build/BUILD/dahdi- linux-2.9.1/drivers/dahdi/firmware' make: *** [install-firmware] Error 2 -A -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 198 bytes Desc: This is a digitally signed message part. URL: <http://lists.digium.com/pipermail/aster...