similar to: Read command without #

Hi, I use asterisk for a phone quiz game. I need to store data in a database (MySql, postgres) : telephone number, name (voice), ... and of course the answers at the quetions. What's the best way to store my data ? - script with system() command ? - AGI script - CDR - others ... Thanks Jerome Vous manquez d?espace pour stocker vos mails ? Yahoo! Mail vous offre

Hello, I have 3 hdd (120 GB, 120 GB and 80 GB) mounted on /data1 , /data2 and /data3. All these drives must be shared via a public access with Samba. For the moment, I can only share the 'data1' directory. [public] path = /data1 Is there a possibility to share several disks under the same account ? By example : [public] path = /data1, /data2, /data3 Then, under Windows, I'd like

Hi all, I am using debian 3.1 and samba 3.0.7. I configured samba as a member of a w2K domain and set up a share in /tmp. Now, when I issue the command 'smbclient -L localhost -Uuser_domain%pass' I get NT_STATUS_LOGON_FAILURE but as guest it works 'smbclient -L localhost -U%'. wbinfo -u and wbinfo -g work well after joining the domain. Thank you for your help. Nirina.

Hi, I'm trying to set up a voip gateway between a classic pbx and ip network with asterisk. phones -- pbx -- * -- ip network I would like a prefix ( 0 ) for the classic calls and another prefix ( 1 ) for voip calls. The problem is that pbx can talk with asterisk only with S0 synchro (like a terminal) and succeeded not to make call with prefix in this mode. I also try to consider asterisk

Hi all, I wish to receive calls from anybody to sip:infos@neos.yi.org in order to test asterisk. Listen music and leave me a message. If you speak french send me a mail i'll give you an other sip URI to test voice quality. Sorry I don't speak english fluently. I use ddns so yours calls might failed if dns is not update or my computer is switched off . Thanks harry Vous

hi all, I need help ! What are Built-in Extension Numbers ? if i dial *69 with callreturn=yes in zapata.conf i don't get the last caller . How may i use Built-in Extension Numbers ? I should not add extensions in dial plan !? Harry from voip-info.org: There are some "extension numbers" that are built into the Zap channel module. You may override these in your Dialplan, i.e.

Hi, My setup is quite complicated. I have to Asterisk server linked via IAX. My Sip phones are connected to one and go out (PSTN) via the IAX trunk and the other server is connected to a Quintum CMS via H323. Phone---(SIP)---Asterisk1---(IAX)---Asterisk2--(H323)---CMS--> PSTN All work fine but when a call someone who is busy I didn't hear the corresponding tone and asterisk2 go to

Hi, PDC works fine, but Samba BDC doesn't make its job. In srvmgr.exe PDC, BDC appear, but when I kill smb PDC's process, normaly BDC may give a response to smb request. My problem... BDC do not respond, no PDC :: no authentification. any idea. my smb.conf : [global] # Main Config. netbios name = LYS workgroup = TNN server string = Lys (TNN's PDC) security = user domain

Hi all, Anybody would be able to call my voicemail just for test sip:infos@neos.yi.org regards harry Le nouveau Yahoo! Messenger est arriv? ! D?couvrez toutes les nouveaut?s pour dialoguer instantan?ment avec vos amis. A t?l?charger gratuitement sur http://fr.messenger.yahoo.com

Hi, Has anyone ever succedded in shaping H323 traffic ? I mean reserve a certain bandwidth for it, in order to have a comfortable Netmeeting, and not be disturbed by downloads & others. I tried with HTB but it doesn''t seem perfect... Thanks for replies, Sam Vous manquez d’espace pour stocker vos mails ? Yahoo! Mail vous offre GRATUITEMENT 100 Mo ! Créez votre Yahoo! Mail

Hello all, I have been searching thru all docs that I can find on wiki and such but can not get an answer. I am trying to collect a date from user input in the form of digits dialed from the phone to use in an agi script to do a database look up. I have tried to use "Get Data filename, timeout, maxdigits " in the agi script. In * console I get message saying playing filename but it

Hi Folks, Consider the following example (artificial, but it illustates the point): > r2<-sqrt(2) > x<-2-r2*r2 > print(c(pi,sqrt(pi)),digits=5) [1] 3.1416 1.7725 > print(c(pi,sqrt(pi),x),digits=5) [1] 3.1416e+00 1.7725e+00 -4.4409e-16 whereas I would prefer [1] 3.1416 1.7725 0.0000 which could be achieved by > print(c(pi,sqrt(pi),round(x)),digits=5)

I'd like some feedback on potentially submitting a request (and probably a patch too) to change the way the AGI command GET DATA works. Right now, # terminates the entry, which is then returned with the # stripped off the end. What I'd like is to allow user configurable termination digits, which are not stripped off the end. Reasoning: Some entries you'd like to terminate with #.

Hello everybody: This could be a stupid question, or may be not; I'm not sure 'cause I have not a very wide experience working with Asterisk, actually I just started last week. I need to make an IVR system work and I choose working with AGIs, written in Perl. The available documentation I've found show it as a very simple proccess, but it doesn't work for me... and I

Hi, I try to connect my asterisk box with a classic pbx (Siemens). I have a T100P E1 card. modprobe zaptel and wct1xxp ok ztcfg -vv ok I can run asterisk and show the channels but the led still flash red My zaptel.conf : span=1,1,0,ccs,hdb3,crc4 bchan=1-15 bchan=17-31 dchan=16 My zapata.conf : ;; E1 switchtype = euroisdn signalling = pri_net group = 1 context=e1 channel => 1-15,17-31

Can anyone point me to a sample or information on using MeetMe like this? Conference room is set up with 2 PINs, one for the moderator and one for the participants. Participants get music until the moderator joins (to avoid wild, un-moderated tangents). Call is ended and all participants are kicked out when the moderator leaves (or the moderator can kick everyone out via phone keypad).

Dear all, I am having trouble plotting a PCA result. The plot doesn't appear!!! R goes through without any errors but doesn't make a plot appear!! Could it be wrong window parameters? In this case how do I change them? I am under red hat 9 with the latest version of R! Thanks. Vous manquez d??espace pour stocker vos mails ?

Hi ; Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card... I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and vice versa ?... Thanks in advance Betul Onemli not : Bu e-mail iletisi, sadece adreste belirtilen kisi veya kurulusun kullanimini hedeflemekte olup, mesajda yer alan bilgiler kisiye ozel ve gizli

Sir, Here is the working Audiocodes MP-11X FXO configurations to work with Asterisk. ;************** ;** Ini File ** ;************** ;Board: MP-118 FXO ;Serial Number: 905371 ;Slot Number: 1 ;Software Version: 5.00A.024 ;DSP Software Version: 204IM => 209.13 ;Board IP Address: 192.168.0.195 ;Board Subnet Mask: 255.255.255.0 ;Board Default Gateway: 192.168.0.1 ;Ram size: 32M Flash size:

Hi, this is my 1.st response to this list, i hope this will work. I tend to agree with Steven since just allowing other termination digits probaly wont solve your upcoming the issues anyway. I use a wrapper around the 'get digit' which allows me to specify that the * digit repeats the menu but maxium 3 times and if the * star digit is used twice in sequence (without other digits inbetween