similar to: Hello Everyone

hi all; my first post/question is a bit vague. i'll be more specific on Asterisk@Home usage (eg: how was it?): 1. am able to make it running at home using a braodband connection of dynamic IP with no-ip, i able to SSH to the box and access to the web pages. the problem is with the maint password. how could i change it? in which mechanism that maint reside? is it on the mysql or the linux

Does anybody have (or know of) a command line application that would: ) Eliminate pops and other random loud noises. ) Trim leading and trailing silence. ) Trim pauses exceeding x milliseconds to y milliseconds. ) Normalize what's left. I know about normalize and have figured out how to trim leading and trailing silence in sox, but I'm looking for more :) Thanks in advance,

Ok, I've been away from the list for sometime now and feel as though I'm going to ask something that's been asked many times before, yet I find nothing in the wiki, so here goes: I notice that allison has done recordings for weather forecasts, yet I find no agi's that parse forecasts and use this recordings to piece together the forecast. I find plenty of weather agi's, but

We are doing this with the latest spandsp, iaxmodem and hylafax. Seems to work very well for us so far. -Jonathan > -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michael Gaudette > Sent: Tuesday, March 21, 2006 3:34 PM > To: 'Asterisk Users Mailing List - Non-Commercial

Hello Everyone, My names is Earl Ramirez, I am from the little islands of Trinidad and Tobago.I will like to be part of this wonderful community I'm new to CentOS and I will like to contribute in any way that I can, I have a few certificates which include Security+ and Linux+ from Comptia. I'm currently preparing for RHEL exam so I think that I might be able to contribute

Firts of all Greetings From Cuba "Saludos" to all of the member of this list. My name it's Orestes Leal, at your service. Well, The problem it's the following: I have one ATI RADEON 200 Series Integrated into the mainboard (ATI Technologies Inc RC410 [Radeon Xpress 200]), the chip has 64 megs of Video and 64 shared from memory to a total amount of 128 megs, the Xorg says me

Hi guys, I am planning to connect two Asterisk boxes that are currently running in two different countries, using IAX. I was wondering if anyone could provide me with some links or suggestion regarding best practices in connecting two Asterisk in such way. I guess many of you have already tried this, and already have some know-how (what I should be careful about, what to avoid, etc...)?

Hello, I am just asking this because I am note sure if the problem is on my side or not, I saw some comments on SIP realtime today so I was wondering, has anybody has SIP realtime working with a softfone ? If yes, please confirm, that would give me a light. My previous message to the list is below. Thanks. Frederic ----- Original Message ----- From: Frederic Jean To:

Hi All, I am trying to compile meetme2 in my asterisk box and getting the following compilaton error. Please help me to sort it out. cc -fPIC -c -o app_dial.o app_dial.c In file included from app_dial.c:14: /usr/include/asterisk/lock.h: In function `ast_mutex_init': /usr/include/asterisk/lock.h:317: `PTHREAD_MUTEX_RECURSIVE' undeclared (first use in this function)

Okay, So remember that question I had last week about taking a bunch of relay servers and greping the status2.xsl file for the number of listeners? I coded it. I have to apologize right now, as its krufty, nasty, braindead code. But it works, and I'm rather proud that after being up as long as I have been up and drinking as much coffee, it works. It should be attached to this e-mail. I

I have added the three lines to the sip.conf file based on the latest changes from broadvoice. I can receive incoming calls but cannot place any outgoing calls. The error I get is: *CLI> -- Registered to '69.73.19.178', who sees us as IPADDRESS:4569 -- Attempting call on SIP/Broadvoice/5068012 for application Playback(demo-congrats) (Retry 1) Mar 8 08:35:21 NOTICE[29290]:

Hello I'm working on a fix-point decoder for AOS/Bluebottle - the proprietary OS of ETH Zurich. I've some problems with Floor0-curve-computation: I don't know what the little omega [cos(w)] stands for, line 3 of the official Vorbis documentation. Does anybody knows what this value means? I've tried to get the answer from the jOrbis and Tremor, but both seem to do - among

Can anybody help me and explain me the following area codes: 1-340 US-USVI 1-670 US-CNMI 1-710 US-Governement 1-787 US-Poerto Rico 1-802 ~ 1-808 ??? 1-939 US-Poerto Rico 1-600 Canada Are the above codes are USA Continental tarrif (NuFone / Broadvoice ... )???? What are the codes for mobile phones in USA? bye Ronald

I'm using a Snom 320 in a CAP position and the receptionist wants to do blind transfers. OK, no problem so far. Now she has asked me how to UN-transfer a call, as in, she transfers a call and wants to hook the call back before it connects (she wanted to tell the caller additional information for example) I don't think that this is possible as once my dialplan starts using Dial()

Hello. I would like to know if somebody did a wireles voip with Asterisk PBX. I think to deploy a wireless for about 500 potential customers, it's a 3 km radius maximum coverage with houses without phone lines, I work for public places telephony small enterprises ( a common bussines in Spain) so I can get good rates from 4 telcos and do LCR at my asterisk PBX. Is anybody did this before

I havent got any mails since 2:42 this morning..usually i get at least the normal 10-15 a hour, if someone gets this can they reply? Thanks! Ron -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060308/c04e3fc9/attachment.htm

Does anybody know of a way to test whether a mySQL connection invoked from the dialplan is current or not? For example: extensions.conf =============== [context] exten => _X.,1,MYSQL(Connect connid localhost user pass db) exten => _X.,n,MYSQL(Query resultid ${connid} SELECT `something` FROM `table` WHERE `number` = ${EXTEN}) exten => _X.,n,MYSQL(Fetch foundRow ${resultid} something)

I have ASTCC installed, and compare it with NuFone, however, I find that the billing of NuFone is always a few secondes more (6 to 24 seconds) Does anybody has an explanation / solution for it? bye Ronald

All - I've been muddling around with this for a few days now.. and I'm trying to figure out why I am not receiving more than one phone call on each polycom 501 phone. I can make more than one phone call out, but not receive another one in, while on a call. Has anybody seen this behaivior before, or is there something simple in the config i'm missing, like.. maxcalls.. or

starting a 1.1 cent per min, rates may be better depending volume technical support we support all codecs using SIP / IAX2 predictive dialers, call centers and telemarketers are allowed free test account. if you have any question just contact us support at cdsportal.net -------------- next part -------------- An HTML attachment was scrubbed... URL: