similar to: Group/Broadcast Voicemail

We are using DISA with local SIP users. The user enters in a 2 digit code then they get a dialtone and the phone dials out. The problem is that the calls waits 10 seconds after the outgoing number is dialed, no matter what I put for the timeout values. Anyone else using DISA that has run into this? exten => _2X,1,Answer exten => _2X,2,DigitTimeout(2) exten =>

Sorry, i also tried this: exten => 6000,1,Answer exten => 6000,2,MusicOnHold(default) and i got this result: *CLI> -- Executing Answer("SIP/2391-8cdd", "") in new stack -- Executing MusicOnHold("SIP/2391-8cdd", "default") in new stack Jun 29 19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class

This is my musiconhold.conf and my folder: root@voip:/etc/asterisk# less musiconhold.conf [classes] default => quietmp3:/var/lib/asterisk/mohmp3 ;loud => mp3:/var/lib/asterisk/mohmp3 ;random => mp3:/var/lib/asterisk/mohmp3,-z ;unbuffered => mp3nb:/var/lib/asterisk/mohmp3 ;quietunbuf => quietmp3nb:/var/lib/asterisk/mohmp3 ; Note that the custom mode cannot handle escaped parameters

Hi, I'm using (or trying to at least) a combination of PAM/passwd and MySQL authentication: auth vmail { mechanisms = plain userdb = sql /etc/dovecot/dovecot-sql.conf passdb = sql /etc/dovecot/dovecot-sql.conf user = root } auth system { mechanisms = plain userdb = passwd passdb = pam imap user = root } MySQL

i dont know how to edit the the time for ringing "30000ms" to "40000ms",please help me. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Hello, Is there or can anyone provide a comprehensive guide (designed for Linux/Asterisk novices) to installing/setting up Asterisk in order to support Fax2Mail service? In my case, I would like Asterisk to receive fax calls to predefined numbers (ranges) and to associate each of these numbers to email addresses. Thank you in advance. David --------------------------------- Yahoo!

Hi, If there are 5 asterisk servers on the local net and each server runs meetme, eg. 3311,3321,3331,3341,3351 respectively. Can I connect these 5 meetme conferences to one meetme using IAX2? Regards, Zen

Sorry, for disturbing the list again. > Also I got one suggestion of using ORDPACK at http://www.netlib.org/. It's ODRPACK at http://www.netlib.org/, not ORDPACK. Best, -- Etsushi Kato ekato at ees.hokudai.ac.jp -.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.-.- r-help mailing list -- Read http://www.ci.tuwien.ac.at/~hornik/R/R-FAQ.html Send

Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be

ei tuka imam edin pf conf obache pravi mnogo nomera, kato se pusne parvoto koeto e dropva paketi, timeoutva po serverite i t.n.. i speed-a e mnogo baven, vijte ako nqkoi moje da otkrie generalna greshka da reply :) vapreki che ne e freebsd-specific :P ne sym go pisal az a i ne sam mnogo mnogo zapoznat s pf zatova ako nqkoi moje da pomogne e dobre doshyl :) btw moje i neshto ot tia opcii kato set

Hi, i have just started to configure access to the * over SIP-Phones. Therefore I have defined this SIP-Phone in sip.conf: [tobias] type=friend username=tobias secret=tobias auth=md5 host=dynamic reinvite=no dtmfmode=inband callerid="Tobias" <1087006> allow=all context=javaAgi dtmfmode=rfc2833 As you can see i am directing calls from this user to the context [javaAgi] which

Dear r-help, I want to conduct nonlinear fitting to a data frame having x and y variables. Because both x and y have measurement error, I want to include error term of x variable in the model. I'm not sure but I think ordinary nls model only consider error term of y variable. How can I do this kind of nonlinear fitting in R. Is there any examples in nls package? Thanks in advance, --

We ultimately found this to be a voicemail issue. The voicemail is held in MYSQL as well (via ODBC). And we found when attempting to playback a customers voicemail unavail greeting is when the deadlock would occur (Immediately, every time. Throwing the same "task processors" errors, And making pjsip completely unresponsive). We had imported a number of greetings from a legacy asterisk

I posted this yesterday to r-help and Ben Bolker suggested reposting it here... Dickison, Daniel <ddickison <at> carnegielearning.com> writes: > > The documentation for agrep says it uses the Levenshtein edit distance, > but it seems to get this wrong in certain cases when there is a > combination of deletions and substitutions. For example: > > >

I'm sure asterisk would make a great stand alone voice mail server. Basically I want to get rid of our voice mail system and replace it with *, but the problem is we use a cisco cluster with skinny clients. So I was thinking the way to contact a * server, would be through our 3640. But so far any attempt has failed. I am wondering if anyone has done something similar. Just want to verify the

> -----Original Message----- > From: Robert Goodyear [mailto:me@jrob.net] > Sent: Tuesday, March 22, 2005 1:21 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] Is there a way to get inserted into an LEC's > CLIDB? > > > Does anyone know if there's a service out there to -- for a fee -- > inject our DID into the

/bin/sh ../libtool --mode=link gcc -I/usr/include/libxml2 -pthread -g -O2 -o ices input.o cfgparse.o stream.o ices.o signals.o im_playlist.o reencode.o encode.o playlist_basic.o im_stdinpcm.o stream_shared.o metadata.o playlist_script.o audio.o resample.o im_oss.o log/libicelog.la timing/libicetiming.la thread/libicethread.la avl/libiceavl.la -lvorbisenc -L/usr/local/lib -lshout -lvorbis

Dear Sebastian and all, This is a Japanese to live in Japan. I have a question. Ogg's window functon is f(x)=sin(1/2*pi*sin^2((x+0.5)/n*pi)) for 0 <= x < n But I cannot understand why this window function satisfys the equation f^2(x)+f^2(x+n/2)=2 which is for MDCT's perfect signal reconstruction. reference: Analysis/Synthesis Filter Bank Design Based on Time Domain Aliasing

http://bugzilla.mindrot.org/show_bug.cgi?id=1340 Summary: Support for Camellia block cipher to OpenSSH-portable. Product: Portable OpenSSH Version: -current Platform: All OS/Version: All Status: NEW Severity: enhancement Priority: P2 Component: Miscellaneous AssignedTo: bitbucket at mindrot.org

I have on ubuntu a digital library application Kete which gives in the log/production.log: Please help, thnx. /!\ FAILSAFE /!\ Fri Feb 05 19:05:08 +0200 2010 Status: 500 Internal Server Error No connection to server (localhost:11211 DEAD (Errno::ECONNREFUSED: Connection refused - connect(2)), will retry at Fri Feb 05 19:05:38 +0200 2010)