similar to: Realtime Database Lookup

Following is the last few lines of output when i try to launch Asterisk:- [app_zapscan.so] => (Scan Zap channels application) == Registered application 'ZapScan' [app_saycountpl.so] => (Say polish counting words) == Registered application 'SayCountPL' [func_cut.so] => (Cut out information from a string) == Registered custom function CUT == Registered custom

Is there a guide anywhere which runs through how to set up asterisk with mysql? I've looked and almost all the document misses out relevant information. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050920/1ce45adb/attachment.htm

Hi, I'm trying to set up remote voicemail pickup. I've created the following dialplan, but when I press *, I am not sent to voicemailmain. The unavailable message just continues to play as normal. exten => 2345551111,1,Set(MAILBOXID=1) exten => 2345551111,n,Set(MAILBOXCONTEXT=company3) exten => 2345551111,n,Voicemail(${MAILBOXID}@${MAILBOXCONTEXT},u) exten =>

I need to change the sip port from 5060 to 5061 actually we already used 5060 for proxy to sip any idea to change 5060 to 5061 so all can acces the sip using this port please help........................ On 4/8/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at

Hello everyone, I successfully set up the mail_crypt plugin using folder keys, and require user's key to be encrypted with a password using mail_crypt_require_encrypted_user_key = yes. As I'm trying to streamline the process of creating a user, and want to develop an application in PHP to help me in that process, I'm very interested in the doveadm HTTP API. Although the

Hi everybody, I'm receiving the message "res_musiconhold.c:309 monmp3thread: Request to schedule in the past?!?!" in asterisk console when I try to put a call on hold. I don't the reason and I'm sure the relative module is loaded. In musiconhold.conf I put these lines, trying something I found in some previous post: ; ; Music on hold class definitions ; [classes]

Hi, I was wondering if anyone is successfully running Asterisk in a cloud environment. If you could state which cloud you are using, I'd appreciate it. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091014/076ff188/attachment.htm

Is there a guy somewhere on how much bandwidth each codec uses, along with the advantages and disadvantages of each one? Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050923/3bee2776/attachment.htm

Hi, Does anyone have any advice on customising firmware of an SPA921 so that it can be locked to a sip provider and display logos on the config pages. Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091019/f6aa2510/attachment.htm

Hi all, I am having a problem with ChanIsAvail. It always returns the same result, regardless of whether an extension is available or not. It always returns 0 Unknown Status. This is my dialplan. exten => _2XX,1,ChanIsAvail(SIP/winsor_${EXTEN}|s) exten => _2XX,2,Verbose(0, ${AVAILSTATUS}) exten => _2XX,3,GoToIf($[${AVAILSTATUS} = "1"]?4:5) exten =>

Hi, Does anyone have an up to date guide for setting up fax 2 email with asterisk? Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150 per month, please contact Kesher

Does anyone know how to rewind the music on hold? Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060202/ab26b18f/attachment.htm

Hi, We have SPA921 handsets which apparently support Paging, however i can't find any information on configuring Asterisk to make a page call. Does anyone have any information on Paging? Many thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm working on a call recording solution. I would like recordings to either be automatically uploaded via FTP, or posted to a URL for processing by our main server. Is Asterisk capable of doing this or will I have to create a separate application that monitors a temp directory for new recordings? I ask because I don't have any experience in Linux programming, so I

Hello, I'm trying to figure out how to limit the number of concurrent calls a client can make. I have a client that has 6 SIP accounts. One for each SIP phone. I want to limit it so that they can only make 2 outgoing calls at a time so that I can bill them "per channel" rather than "per extension". A separate (but not so important) issue is that I want them to be able to

Hi, Does anyone know how to play music to a caller while you dial a second call? Once the second calls has answered, i'd like to music to stop, and the calls to be bridged. Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060130/eb3402df/attachment.htm

Hello, Does anyone know how to test the timing device? I've tried the following but with no luck. Zaptel is installed. I'm trying to use ztdummy as a timer. [root at TemplateAsteriskServer ~]# dahdi_test Unable to open dahdi interface: No such file or directory [root at TemplateAsteriskServer ~]# zttool Unable to open /dev/zap/ctl: No such file or directory Thanks Dan

Please stop emailing me personally. If no one replies to a post, it means that everyone is busy or they think you should read through the documentation before posting. If you can't figure out simple things like Music on hold from the documentation, then i dont think VOIP is for you. -----Original Message----- From: asterisk at opensourcesolution.in [mailto:asterisk at opensourcesolution.in]

> Hello > as you can read in my original post "moh reload" and "module reload res_musiconhold.so" does nothing. > Only at restart the new files are available. > Is this a bug ?? How can I get more debugging for this problem ?? I think there is currently a bug with MOH. For now, if you add a file to a moh folder, 'touch musiconhold.conf' and then reload moh.

Hello, I was wondering if anyone has any insights on the best way to automatically monitor an asterisk box to check it is constantly available and processing calls. Many thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more