similar to: getpeername failed

--- Bahya NASSR EDDINE <bahya_nassr@yahoo.fr> a ?crit : > Date: Wed, 24 Aug 2005 12:14:37 +0200 (CEST) > De: Bahya NASSR EDDINE <bahya_nassr@yahoo.fr> > ?: samba@lists.samba.org > Objet: [Samba] getpeername failed > > Hello there, > > I am using a samba 3 PDC with OpenLDAP directory. I > can't log on to my samba domain any more from my > windows

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, 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]

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 >

I agree you however i solved my problem with app_voicemail.c The table scheme provide in doc/README.odbcstorage don't match to sql queries in app_voicemail.c I think developpers who has written app_voicemail.c for ARA provide a useable table ! Regards Harry --- Steve McMahon <voip@digitaldatabits.net> a ?crit : > These questions should be sent to Asterisk-Users > this is not a

I agree you however i solved my problem with app_voicemail.c The table scheme provide in doc/README.odbcstorage don't match to sql queries in app_voicemail.c I think developpers who has written app_voicemail.c for ARA provide a useable table ! Regards Harry --- Steve McMahon <voip@digitaldatabits.net> a ?crit : > These questions should be sent to Asterisk-Users > this is not a

Goos morning all, I compiled Samba 3.0.14a with OpenLDAP 2.1.22-0 directory. I then enabled TLS between Samba and OpenLDAP. The following tests succeeded: s_server to s_client --> OK slapd to s_client --> OK slapd to OPenLDAP client commands (ldapsearch..) --> OK The problem is the following: when I start Samba (service smb start), slapd output returns: TLS trace:

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

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

Hi, I''m coding a shopping cart using your effects library. There are 2 parts in my page. On the left the list of my products and on the right the shopping cart. I can sort dynamically my products (by price, title, etc...) . I have a problem when I do a sort, and I want to drag a product. My system keep the initial position of each product. The products moves correctly after the

Hi all, I configured a Samba 3 server as PDC with OpenLDAP server. I enabled TLS. When I start samba (service smb start), the slapd server output shows: TLS trace: SSL_accept:SSLv3 flush data connection_read(16): unable to get TLS client DN error=49 id=23 The "net" command as user1 failes with the following error message: Could not connect to server 127.0.0.1 The

Remarque : message transf?r? en pi?ce jointe. ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com -------------- next part -------------- An embedded message was scrubbed... From: harry gaillac <gaillacharry@yahoo.fr>

Hello, I read wiki however I can't install voicemail.conf i wish to stored voicemail conf and voice messages via odbc in mySQL db . Thanks for help Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger T?l?chargez cette version sur http://fr.messenger.yahoo.com

I don't want to offend you but you should have a look to sems . You won't find docs to help you at asterisk.org --- "lucape@inwind.it" <lucape@inwind.it> a ?crit : > hello > > help me to configure ser + asterisk > how to do the configuration? > > Luca > > > > ____________________________________________________________ > 6X

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, What's dnsmgr ? Anybody could tell mr more? cat /etc/asterisk/dnsmgr.conf [general] ;enable=yes ; enable creation of managed DNS lookups ; default is 'no' ;refreshinterval=1200 ; refresh managed DNS lookups every <n> seconds ; default is 300 (5 minutes)serveur1:~# Harry

Hello, Asterisk is said to handle call routing and codec translation. I would like to force transcoding function with asterisk but when I try to force transcoding I get the errors: codec not compatible or WARNING[4425]: app_dial.c:1024 dial_exec: Had to drop call because I couldn't make SIP/xxx compatible with SIP/yyy How exactly works asterisk, in order to transcoding? If you have any

Am i alone with this problem ? I just rewrote voicemessages table because of errors. I read app_voicemail.c to fix my problem. However app_voicemail.c support many schemes to query the tables. Harry --- "Jerris, Michael MI" <mjerris@ofllc.com> a ?crit : > > harry gaillac > > > > I agree you however i solved my problem with > app_voicemail.c > >

Am i alone with this problem ? I just rewrote voicemessages table because of errors. I read app_voicemail.c to fix my problem. However app_voicemail.c support many schemes to query the tables. Harry --- "Jerris, Michael MI" <mjerris@ofllc.com> a ?crit : > > harry gaillac > > > > I agree you however i solved my problem with > app_voicemail.c > >