similar to: Problem when I call someone who is busy

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, For my IVR, I use Read command. It works fine when ending bu # but I can't get anything without ending by # The wiki tell me is it possible with maxdigit option but it don't work for me. my command : exten => 3,1,Read(ILE,as/iles,1) Anybody can tell me howto do thanks Another question about read command: Howto sup file option and keep maxdigits options ? exten =>

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, 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

Hi, I am faced with this dilema of asterisk not sending an ACK after it receives 200 OK from OpenSER (which is a response to a reinvite request sent by asterisk. Here is my setup Carrier<->OpenSER<->Asterisk1<->Asterisk2 A user is connected with Asterisk1 (through the carrier and OpenSER). On certain dtmf events the call is forwarded to Asterisk2 using the Dial command.

Hello group, I wanted to try out the asterisk iax protocol between two asterisk machines but have several problems with it. My scenario looks like follows. I am using asterisk 0.9.0 on both machines. SER1 <-> asterisk1 <-> IAX <-> asterisk2 <-> SER2 Both SER and asterisk run on a machine with a public IP address. When the telephone on one side makes a call the telephone

Hi Kevin, no you didn't miss the reply and I've not resolved it yet. Have you got similar problems? Pamela Kevin Fjelsted wrote: >Pamela, >Did you resolve the problems you described? >I didn't see a reply on the list but I may have missed it. > >-Kevin > >-----Original Message----- >From: Pamela Weis [mailto:peawy@gmx.at] >Sent: Thursday, August 05, 2004

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 ?

Sorry, I am resending this, I tried earlier, but I couldn't see it appear on the archives - apologogies if it appears double! -------------------------------------------------- My Sipura 3000 ATA died on me this morning. I had a Linksys SPA 3102 available which I would like to use as a replacement. Unfortunately, the SPA3102 is not able to register with the asterisk server - I am

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I've been struggling with an ongoing problem the last month. Here is the layout of the wiring: T1 from ISP > DiTech Echo Cancel device > Voice Channel-Bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1 server zap card > fax channel bank (same) T1 from ISP > (same) DiTech Echo Cancel device > asterisk1

Hi, I recently configured Linux HA for Asterisk service (using Asterisk resource agent downloaded from link: https://github.com/ClusterLabs/resource-agents/blob/master/heartbeat/asterisk ). As per configuration it is working good but when I include "monitor_sipuri=" sip:42 at 10.3.152.103" " parameter in primitive section it is giving me an errors like listed below; root at

Hi, i. After estimating some coefficients using robust regression with rlm() or lqs(), I wonder if there exist some measures of the goodness-of-fit as those for standard linear model(R2)... or evenly if it's a statistics non-sense to look for since I do not find any mention of that in differents chapters on robust and resistant regression or in severals R documentation (Fox, Ripley and

Hello, I need help for that error message: ?chan_sip.c:14889 handle_response_invite: Failed to authenticate on INVITE to? My network is: Client1-- -----------asterisk1------asterisk2 Client2-- ? With client1, I do a call ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Asterisk1 forward the call to

Hello, My network is: Client_SS7_1-- -----------asterisk1------asterisk2 Client_SS7_2-- ? I receive a fax from Client_SS7_1 ? Asterisk1 forward the call to asterisk2 ? Asterisk2 forward the call to asterisk1 ? Then, asterisk2 forward the fax to Client_SS7_2 I want that the SIP signaling go to asterisk2, But, I need that the RTP don?t go