similar to: Sip test

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, 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, 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, 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 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

Got my first round of IP500s in today. Anybody have any example sip.cfg files they'd like to share? Tim Jackson Network Engineer Angelina County, Texas (936)639-4827 office (936)414-6723 mobile -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040928/a923e094/attachment.htm

Hello, If asterisk.org can't provide you documentations have a look here : http://www.digium.com/index.php?menu=product_detail&category=software&product=ABE I do hope some people understand my posts. Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger

hello, I try to set up two lines per ip 300 phone, registration is ok but i get Failure to authenticate 407 for subscribe. Anybody could help me to configure Asterisk in order to set instant message and presence ? I've tried with Ondo sip server it's ok ! Regards D?couvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Cr?ez votre Yahoo! Mail

Hello, What do you think of this project www.openpbx.org ? Something like ser and openser ! Kinds Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version 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 have a polycom reboot script which sends a NOTIFY with check-sync. It worked fine with 1.3.4. After I upgrade to 1.4.1, it stopped working. Anyone has the same problem? Thanks, Richard

Hello, I'm new to Asterisk and would like to know SIP support status. Are there any testing been done with some widely deployed client (Cisco SIP IP phone, ...)? I was using Vocal but I'm now interested in Asterisk as it seems to offer more features...if it supports SIP. Thanks for your help. Francois.

hello, I need help with ast_data I downloaded asterisk from cvs cvs -d :pserver:anoncvs@cvs.digium.com:/usr/cvsroot co -r HEAD asterisk and the latest ast_data. When i run ./INSTALL.txt i get : serveur1:/opt/asterisk/ast_data# ./INSTALL patching file contrib/scripts/sip-friends.sql patching file contrib/scripts/iax-friends.sql patching file apps/app_voicemail.c Hunk #1 succeeded at 27 with

Hello, I patched asterisk cvs head sources with http://juraj.bednar.sk/work/software/asterisk/messaging/ and presnce patch without success. asterisk send "405 method not allowed" to sender. I use polycom ip300. Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger

Hello, I wish to use asterisk as a voicemail server with ser . I want to use asterisk external configuration toHello, I wish to use asterisk as a voicemail server with ser . I want to use asterisk external configuration to manage users and storing voicemail messages according to ser database. Where can i find the schema of the SQL DB for voicemail accounts . for example in extconfig ;

Sorry it's asterisk-users@lists.digium.com --- harry gaillac <gaillacharry@yahoo.fr> a ?crit : > Luca, > > you may find help here: > > http://www.cs.colostate.edu/~somlo/CSU-SIP-notes/ > http://www.asteriskdocs.org/ http://www.voip-info.org/tiki-index.php?page=Asterisk+at+large > > ask for help to asterisk-users@lists.digium.org > > Regards >