similar to: voicemail

Hi, I have remove the comment defor res_odbc.so and res_config_odbc.so in my modules.conf, but the module is still not loading when I do: module show like odbc I have o module returned anybody knows why? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090127/0963b5a4/attachment.htm

Hi! http://www.voip-info.org/tiki-index.php?page=Asterisk+sound+files+international#German lists a plenty of sound files for German. Can someone recommend one for Asterisk 1.4 (any maybe 1.6 soon). thanks klaus

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

Does anybody think it would make sense for /etc/init.d/asterisk to run /etc/asterisk/startup.d/*.sh on start like safe_asterisk did? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied -> http://www.amooma.de Gesch?ftsf?hrer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP

Is there any method in Asterisk to enable the updating process into another SQL database through entering IVR options during the call. Thanks a lot. _________________________________________________________________ Invite your mail contacts to join your friends list with Windows Live Spaces. It's easy!

Howdy, Working on some niche requests from one of my hotel clients. asterisk 1.4.20-1 on CentOS, Polycom 501s. The first request is for the Polycom's screen to show the CID of the inbound caller when a call pick is executed, so the picker knows if the call is internal or external. I have already "worked around" this issue by using ALERT info to give seperate ring tones for

Hi, is it possible to hangup a channel from another channel? I want to finish a call from another channel, but if I put exten => h,n,HangUp(channelname) it doesn't hangup... Is that correct? Thanks, _________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL:

I just want to confirm but it seems that if-then-else is not permitted in case structure. It was not really documented but it seems to be the case. Can anyone confirm? switch(${DIALSTATUS}) { case NOANSWER: { // if-then-else not permitted If (${ael-var} = 1) { Playback(beep);

I would like to know if I can set Call Forwarding on an extension but allow direct calls from the extension it is forwarded to. Example: Extension 100 sets call forwarding (all) to extension 101. All calls to 100 are immediately forwarded to 101 as expected. However, if 101 tries to transfer a call to 100 or tries to call 100 directly, it sounds "busy" because it obviously goes into

Hello, I'm just wondering if anyone has fixed the 'InUse&Ringing' problem. * v1.4.23.1 Ta

~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ CLI Output : ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ vicidialnow*CLI> == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Parsing '/etc/asterisk/manager.conf': Found == Manager 'sendcron' logged on from 127.0.0.1 == Manager 'sendcron' logged off from

Hi all, I have a question: how can I see hints of a remote Asterisk in IAX2 trunk?? I want to set BLF on my phones to look state of other phones also in other Asterisk server. Someone have any idea or solution? I use Asterisk 1.4.24. Thanks all Marco -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have an account with FWD and I have configured my SIP.conf with [fwd] type=friend secret=password username=901835 host=fwd.pulver.com But when I am trying to dial out my own DID , I dont see any call landing in asterisk. In extension.conf (vicidial) file I have exten => 2062036895 ,1,Ringing() exten => 2062036895 ,2,Wait(1) exten => 2062036895 ,3,Answer() exten => 2062036895

Hello all, I have being trying to replicate the following call scenario with my Asterisk box: http://www.tech-invite.com/Ti-sip-service-8.html <http://www.tech-invite.com/Ti-sip-service-8.html> I have a situation that I have to notify the calling party that the call is being forwarded to another number. So far, in the tests that I made, calling from a SIP extension to another SIP

Heelo, I currently search a program that can make a web browser Pop-up on an incoming call on a specific URL like : http://directorie.ch?CALLNUMBER:00451849799 I have found ADM, but it's a bit more complex for my purpose an it's not very stable. Do you know a simple software for that ? An other part of my project is to eneable click-to-call from a web page, do you know a kind of

On 1.6.1, I must be losing my eyesight: [internal] include => outbound-pstn ............. include => meetme ; 2663 include => setup-meetme-conf-room ; 6000xxxYYYY [setup-meetme-conf-room] exten => _6000XXXNXXX,n,Set(Time-in-secs="${STRFTIME(${EPOCH},,%s}" ) ........ CLI: -- Starting simple switch on 'DAHDI/1-1' [2009-05-17 14:54:49]

Hi All, Asterisk 1.4.12 on CentOS 5 Sorry for a question that I'm guessing is obvious to most of you. I'm trying to revamp my dialplan. When I first created it, I had something like: exten => s,1,Set(CALLERID(name)=${DB(cidname/${CALLERID(num)})}) exten => s,2,Dial(${rgMain},${RINGTIME},t) exten => s,3,VoiceMail(main at default) exten => s,103,VoiceMail(main at default)

JFYI and slightly off-topic: All of the videos we recorded at AMOOCON open-source VoIP conference (Rostock, Germany, May 4-5) are now available on the web site: http://www.amoocon.com/ All of them are available in different qualities and formats, including Quicktime 7, versions for the iPhone and iPod and h.264 which IIRC can be played in MPlayer etc. 100 GB in total. :-) Philipp Kempgen

Hi, If I have multiple kernel sources in /usr/src, e.g. linux-headers-2.6.26-1-686 linux-headers-2.6.26.custom.1 how does the Zaptel Makefile(?) know which one to pick? Is it a good approach to compile the kernel first and then compile Zaptel "manually" afterwards? Or should I rather put zaptel in /usr/src/modules and use fakeroot make-kpkg ... modules_image in the kernel

Hi, I had downlaoded iaxclient-2.0.2 and complie project *\iaxclient-2.0.2\contrib\win\vs2005* ** It gives many83 fatal and file missing error of file missing Error 1 fatal error C1083: Cannot open include file: 'portaudio.h': No such file or directory d:\mohit\asterisk\iaxclient-2.0.2\iaxclient-2.0.2\lib\portmixer\px_win_wmme\px_win_wmme.c 40 Error 2 fatal error C1083: Cannot open