similar to: Retreiving the called number

Hi, Dovecot currently treats an LDAP user/password database the same way as a text or SQL based database: it just tries to retreive the (hashed) password for a given username. LDAP however has the capability to authenticate the user itself: dovecot could try to bind to LDAP with the given username and password, and if authentication succeeded, the LDAP server returns the other info (uid,

Hi all, In order to do loadbalancing between my two *, i wanted to stock all things concerning voicemail on a NFS partition... I see that the voicemail system put his files onto two differents directories : /var/spool/asterisk/voicemail/mycontext etc. and /var/lib/asterisk/voicemail/mycontext etc. I've two questions : Why ? and how can i do to centralize the destination of the messages AND

Hi, I am new to asterisk configuration. I want to get called number in features.conf. I am defining a feature in features.conf and that feature got executed on pressing a particular DTMF key sequence. As I want to execute my own application on pressing that key which will use called number. testfeature => 3,peer,AGI,StoreNumber|<CalledNumber> Here I want to use called number in place

The variable ${CONTEXT} stores the value of the current context. However if we are in a macro that will be the name of the macro. How do I access the name of the local channel's context. For example: [macro-test] exten => s,n,NoOp(Context ${CONTEXT}) CLI shows: -- Executing NoOp("Local/2592@1100006-2000-e802,2", "Context macro-test") in new stack I want to get

Hi, I am making calls from the dial plan using the dial application. Due to technical requirements I need to find out the sip call-id used in the dialog initiated by the dial application. I dont see any straight forward way of doing this so I am looking for answers. There is a sip callid session variable but the problems is that dial is a blocking call and the dialog ends when dial returns. I

If you have two asterisk systems how do you hook them up together so the users of one system can make calls onto the other system. Thanks Steve steve@17q.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040927/69058745/attachment.htm

I'm toying with adding a feature request to provide some sort of gain setting for voicemail when accessed from "certain" interfaces. Maybe something like voicemail=6.0 (db) within a specific channel section of zapata.conf corresponding to a pstn line. Situation: 1. Someone calls into asterisk and leaves a voicemail. The sound is recorded at some volume well below 0 db, and is

Is there any way to program a soft key in ADSI to put a caller on hold. Then able to retreive that caller. Example - Softkey Hold Softkey Retreive Call Softkey End Call -gcc

Hi to all, I and using asterisk with following setup. 1. TDM400p card with four FXS modules, so there are four analog phone lines on four zap channels, My setup is working fine. And version is like such Asterisk CVS-v1-0-11/27/04-20:48:45 I want your guidance for the following issue. with help of agi scripts i am able to insert caller id number in database of mysql now i want to pop it up via

I am trying to retrieve the "dialed peer number" but it seems that ${DIALEDPEERNUMBER} is "broken". Also, I know that I could extract the dialed number from the ${CHANNEL} variable but this only works for SIP and maybe IAX (untested). However, it doesn't work for ZAP. All I get when using ZAP is something like "Zap/1-1" (for SIP I would get

I'm having a big problem where digits dialed from certain cell phones are too short to be recognized by my asterisk server. I'm running AAH 2.8. Some cell phones don't allow the caller to hold down the digits and have the tones play as long as they hold them down for. They just play a short tone no matter how long you hold down the digits for. Has anyone run into this before, and if

I am using asterisk along with freepbx . When recording is enabled for a extension the call record file made in /var/spool/asterisk/monitor contains information like OUT(extension number)-(timestamp)-(uniqueid).wav . This can be a big mess if there are more than 1000-2000 files in that folder and very hard to locate a call recording based on call time and extension number who dialled. I need to

How do you retreive the caller id on incoming analog lines and display the id on the sip phones on the network? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060201/076073d1/attachment.htm

Hi, I am looking for a TFTP server that is easy like the tftpd32 for Windows that I have been using. Just want to start it with a command and my Cisco can connect and retreive the config files from it. Many thanks, Christian

Hello, I need help, after hours trying to do it myself, to do the following: I have an object, named, say, F1 from a class "fichier" that I buildt. this F1 has a slot named datas ,which is a vector, so I can access to : F1 at datas[1], F1 at datas[2], ... Now I have a string s="F1 at datas[1]" and need to retreive the real content of F1 at datas[1] using s.

Hi, I''m using the ''backup'' parameter in file resources to backup to a central filebucket on the puppetmaster. How do I restore a backup? I couldn''t find anything in the wiki, but saw some docs by running ''filebucket --help''. It says to restore a backed up file, you need to know the md5sum. Unfortunately, AFAIK the only place this

Hi all... I need to make a few queries to WINS, but am unsure how to do so. 'nmblookup' will allow me to enter an IP address and get information about the machine name, domain, and username. I want to do the opposite...I would like to have a tool that will take a username and return the IP address. I know this information is stored in WINS somehow, and I am capable of writing some Perl

Dear R users, I have now managed to fit the curve using the thin plate spline as follows: library(mgcv) b <- gam(y~s(x1,x2,k=100),data =dat) vis.gam(b) What I want now is to get the fitted data for y and copy it so that I use it for further analysis. Many thanks in advance mintewab

Hello Guys, I'm somewhat of a newbie and am desperately seeking for some help... I've managed to get asterisk up and running on my server, and signed up with a broadvoice account... I'm having no problem dialing and communicating between extensions, but whenever anyone tries to call my broadvoice account, they are greeted by no ring or anything, but rather simply a direct to

All, According to my readings CDRs are stored at the end of the call. My concerns is when asterisk goes down (I know that it's never happen but it's just in case) or when the is a power shutdown of the server. then CDRs are not stored in mysql. is there a way to store periodially CDR during a call, and set the periodical timer regarding the context. if no is there a way to retreive CDR,