search for: portillo

Hi. In creating a custom extension, and dialing SIP/222/333#444, seems the party receives only "333" What should I do to send the # symbol? or better, where can I find that syntax? Googled a lot, nothing. Thanks! -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft f?r Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gesti?n P?blica Descentralizada y Lucha

Hi. Does anyone knows how to trigger a phone call from a bash command? Thx! -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft f?r Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gesti?n P?blica Descentralizada y Lucha Contra La Pobreza - PADEP Av. S?nchez Lima 2226 La Paz, Bolivia Tel: +591 22417628 (121) Fax: +591 22417628 (126) Web: www.padep.org.bo Email:

...nt here if I said "African Brit" stinks of an accusation of "you traitor!" They're not Africans, they're Caribbeans or Nigerians, or or or. And many Africans are White South Africans, and some Caribbeans (like me) are also white. How many people here remember Michael Portillo's "Cricket Test" for whether you are English (the joke here being *he* was of Spanish descent). So please do things for sound *technical* reasons, and sod the Politically Correct. That's usually an excuse for oppression by the Morally Superior. Cheers, Wol

Hi. I have a SPA3102 updated with with Software Version: 5.1.7(GW). I have this settings on Voice/Regional: Interdigit Long Timer: 10 Interdigit Short Timer: 3 Anyway, when hooking up (without dialing anything), the timeout starts after 3 seconds. It's like the Long Timer is unused. After dialing, the Short Timer is also used to timeout. Is that normal? Am I missing something? Thanks. --

Hi, I just want to deny FOP originated calls in TRIXBOX. All remaining operations (hanging up, transferring, etc) should go. Where is that option in TRIXBOX (already googled, nothing, checked config files but cant find that option)? Thanks a lot. -- Rodolfo Alcazar Responsable red y datos Deutsche Gesellschaft f?r Technische Zusammenarbeit (GTZ) GmbH Programa de Apoyo a la Gesti?n P?blica

Im a 3-days-asterisk-newbie. In 3 weeks, I must have a PBX installed in a new office of ours: Panasonic or Asterisk. Asterisk would be, if I can emulate some Panasonic functions on Asterisk fast, to convince the executives. What I have done until now: Bought 1 Linksys pap2 (2 FXS), 1 Linksys SPA3102 (1 FXS + 1 FXO) for making asterisk tests. Configured Asterisk/Fedora 9 so I can make SIP->PSTN

client with my asterisk. If i am wrong, please let me know On Wed, Jan 7, 2009 at 4:43 PM, Rodolfo Alcazar Portillo < rodolfo.alcazar at padep.org.bo> wrote: > Missed the thread, sorry. Gizmo5.com has some blackberry SIP clients. > Could be what you want. > > Greets! > > Am Mittwoch, den 07.01.2009, 16:07 -0500 schrieb Eric Moniz: > > TianLun, > > > > I should have know...

...n the same box and in other box with the same results) If I just try liveice (without running first icecast), liveice seems to work but the icecast server doesn't report any source connection, I got in the screen this "###" running and if I do ps ax I see mpg123 -sq /mnt/windows/yss/portillo.mp3 (something that I don't see if I run first icecast and then liveice) What am I doing wrong? please help me! Thank you, Rocael. __________________________________________________ Do You Yahoo!? Make international calls for as low as $.04/minute with Yahoo! Messenger http://phonecard.yahoo...

Hi You all, Does anyone know any SIP client for BlackBerry? thank you -- TianLun Song We care your day to day business operation CCVP, CCNP, M.Eng Cell:1-647-868-2950 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090106/a278cbbd/attachment.htm

Hi, When we moved to GitHub a few months ago, we used without more consideration the "master" convention to name our development branch. On SVN it used to be just "trunk". This naming is unfortunate <https://tools.ietf.org/id/draft-knodel-terminology-00.html#rfc.section.1.1> as it can hurt some contributors

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to