similar to: TLS connections between Samba&OpenLDAP

I think I guess what the error is. I've configured slapd to require a valid certificate for all TLS incoming sessions. However, I didn't create a ertificate for OpenLDAP client. I am going to do so. --- Bahya NASSR EDDINE <bahya_nassr@yahoo.fr> a ?crit : > Date: Wed, 27 Jul 2005 11:46:50 +0200 (CEST) > De: Bahya NASSR EDDINE <bahya_nassr@yahoo.fr> > Objet: RE: RE

Hi, Is there a module within the Asterisk standard distribution that provides SNMP features? Is there any third party software for that purpose? Regards, Stojan Sljivic -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/918b5ebf/attachment.htm

Can someone who is knowledgable in the traditional telco space please give me a layman's explanation (or point me to an appropriate url) of LBO as per the zaptel configuration file? # The line build-out (or LBO) is an integer, from the following table: # 0: 0 db (CSU) / 0-133 feet (DSX-1) # 1: 133-266 feet (DSX-1) # 2: 266-399 feet (DSX-1) # 3: 399-533 feet (DSX-1) # 4: 533-655 feet (DSX-1)

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 try set Ua---SER----Asterisk (voicemail/ARA) | Ua ser stable asterisk cvs head I read http://mail.iptel.org/pipermail/serusers/2005-February/015997.html to forward unavailable or busy sip agents to asterisk voicemail in failure route. How may I configure extensions.conf and ser.cfg ? I have been trying without success! Regards Harry

Hi, Please, what does this message mean: "getpeername failed. Error was Transport endpoint is not connected"? I get the following in my log.smbd file each time I try to log to my samba domain from a Windows XP client: [2005/08/24 09:49:09, 0] lib/util_sock.c:get_peer_addr(1150) getpeername failed. Error was Transport endpoint is not connected [2005/08/24 09:49:09, 0]

What's wrong with ARA (asterisk realtime architecture) from voip-info: Asterisk, SER and MWI http://mail.iptel.org/pipermail/serusers/2004-December/013727.html Actually I wrote a patch for this and it supports ast_data too. What you do is tell asterisk that all of your phones IP addresses are your SER machine. Then when a message gets left Asterisk sends the NOTIFY to username at

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, 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

Hi all, I've just updated to current CVS, and have 2 polycom IP phones, one is a IP600 and the other is a IP300. The IP600 shows the status of the IP300 and a ZAP line quite nicely, but the IP300 won't show the status of the IP600.... Is there any additional debug apart from "show hints" to see why this might not be working ?? -= Registered Asterisk Dial Plan Hints =-

Hi, I'm setting up Asterisk as a voicemail with SER. My problem is, when a caller that is not registered with asterisk (no username and password in sip.conf) it prompts "403, Forbidden" . I need all calls from outside of my network to reach asterisk for my users' voicemails, because anonymous users will surely reach voicemail of my users to leave messages. What do I

Dear Asterisk community, sorry if I'm so stupid, but I couldn't register myself with Asterisk. I created the [sip-incoming] context in the sip.conf: [sip-incoming] type = peer username = elzhov port = 5062 ; my kphone listens port 5062 host = 127.0.0.1 Then run Asterisk, and checked peers that are known for Asterisk: *CLI> sip show peers Name/username

Note: forwarded message attached. __________________________________ Discover Yahoo! Have fun online with music videos, cool games, IM and more. Check it out! http://discover.yahoo.com/online.html -------------- next part -------------- An embedded message was scrubbed... From: Khubeka JM <jmkhubeka@yahoo.com> Subject: JE TROUVE QUE VOUS N'ETES PAS HONETE! Date: Sun, 26 Jun 2005

Greetings, Recently there has been talk of the presence/buddy feature with asterisk and Polycom phones. I have it setup, and working as expected, however I can only get 7 buddies to appear on the screen at any given time. Has anyone gotten more than 7 buddies to appear? I'm just trying to find out if this is some polycom limitation, bug, or my error. Thanks, Matt -- Matt Gibson VOIP

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 ;

Hi All, My family and I are doing well. Thank you all for your prayers. We are using this as an opportunity to rebuild. I didn't think I really needed to but God knows best and we will obey. My family and I will temporarily be in Lafayette, Louisiana for a while but will probably relocate to Houston, TX in the future. We already have my Daughter registered in school here. Lafayette is my

Hello, I've been planning to replace my aging CENTREX switch with a new PBX and am seriously considering Asterisk as my solution. I work at a college, and we currently support just under 300 regular analog lines to the offices and whatnot. I was wondering.. Is asterisk ready for such a job? Would I be making a mistake to deploy this across the campus at this time? It seems that

Hello I searched on the net and found nothing about it : when I would like to create a symbolic link on a smbfs partion, I get an error : > ln -s ../foo bar ln: creating symbolic link `bar' to `../foo': Permission denied > ln -s etc/foo bar ln: creating symbolic link `bar' to `etc/foo': No such file or directory But : > ln -s foo bar > ls -l bar lrwxrwxrwx 1 XXX

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 >

Hi, I using openldap 2.0.23 and samba 2.2.4 on a Redhat 7.2 Linux distrib. I've compiled with ldap support dans It works fine in clear mode. I've configured unix auth. in order to use ldap on TLS mode, and it works also. When I try to use TLS more (or SSL on 636), it doesn't work. LDAP doesn't seem to have an error (see logs below), but samba tells "Failed to issue the