search for: systinet

...+ available updates - computer is a domain member controlled by another samba (2.2.8a) with ldap backend) - problematic computer is a file server under medium/high load (about 50 clients read and write to it gigabytes of data) /etc/samba/smb.conf # Global parameters [global] workgroup = SYSTINET server string = Buildsrv security = DOMAIN password server = 10.0.0.3 encrypted passwords = yes log level = 10 log file = /var/log/samba/%m.log max log size = 5000 wins server = 10.0.0.3 smb ports = 139 445 name resol...

...l 19 17:15:56 ustredna ztcfg: Channel 12: D-channel (Default) (Slaves: 12) Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna ztcfg: 12 channels configured. Jul 19 17:15:56 ustredna ztcfg: Jul 19 17:15:56 ustredna zaptel: Running ztcfg: succeeded Thank you, -- - David Hajek IT/IS Manager Systinet Corporation Phone: +420 2 7201 9526 Cell: +420 604 352 968 hajek@systinet.com http://www.systinet.com

Hello, is it possible Asterisk's ChangeLog will contain a reference to appropriate bug number in bugzilla? This can be very handy. Thanks, David

Hi, what new Dell servers are compatible and KNOWN to work with Digium TDM cards? I've looked at Digium's compatibility list at http://www.digium.com/index.php?menu=compatibility. Does this mean that other Dell servers like SC1420, SC1425, 800, 1800 are working just fine with TDM cards? Can someone clarify this? Thanks -David

Does anyone test Asterisk 1.2.X + bristuff-0.3.X and TDM card? We can't get it to work. -David -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Julian J. M. Sent: Friday, March 31, 2006 1:44 AM To: Chris Earle; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Re: BRI

Hi all, I have a box running asterisk with T410 connected to a Nortel DMS 100 switch and another box running SER with grandstream phones on it So if there is a call from the pstn it goes from the Nortel to the asterisk and then to the SER box and finally to the phones.if the phone is busy or the number is invalid the * box will first send an ALERT message to the Nortel and say the call is going on

Thanks. I think our problem ca be similar. Have you tried to call from analog phone #1 to another analog phone #2? It works. But when you try to call vice versa from #2 to #1 it does not work. When you restart asterisk it works again - but only one direction. -David ________________________________ From: asterisk-users-bounces@lists.digium.com

Hello, I'm running Asterisk and using VoicePulse for IAX termination. I would like to have toll-free number assigned to my asterisk, any hints where I can get this number? VoicePulse does not offer toll-free numbers. Thanks, David

Hello, I understand there's no possibility to have asterisk configuration (sipusers, extensions, voicemail) in LDAP right now. I'm thinking about put the (sipusers, extensions, voicemail) info in LDAP and then run some synchronization script on the asterisk server which will build up appropriate configuration files and reload asterisk. I'm sure this script is already around. Can some

Hello, I'm experiencing choppy sound first 2-3 secs of every call. Looks like there is some more handshaking between Asterisk and other party. After that voice quality is pretty good. Using IAX2. Any hints? Thanks. David

Hi, I'm experiencing some voice delay (2-3 sec) after outgoing call is setup. It means during the first 2-3 secs, audio is very choppy or nothing. So usually I can't hear the 'Hello". I use IAX2 for my ougoing calls with Grandstream phone as a client. Any hints to prevent this? Thanks, David

Whats up to VoipJet.com? Their DNS servers are not reachable. Both primary and secondary are on the same subnet - weird setup. Thanks, David

Hello, I'm doing lot of international calls via Sixtel and VoipJet. And there are some calls which do not go through - Asterisk immediatelly returns with NOANSWER. And it is not because the dialed party does not pickup the phone, it is because the call does not go through the provider. I've written a dial macro which route the call via second provider if the first returns

Hi, can you recommend cheap GSM Gateway which works with Asterisk? VoiceBlue solution is quite expensive. Thanks, -- - David

Hi, what is the status of chan_cornet? Does someone here use it in production? I can't find enough info about it. Some URLs will be great. Thank you, -David

Hi, I'm running Asterisk 1.0.7 and would like to add Speex support. I downloaded Speex 1.0.5, installed and recompile Asterisk again. Now trying from X-Lite to connect using Speex but getting lot of weird erros on Asterisk console: Sep 7 15:03:25 WARNING[28605]: codec_speex.c:166 speextolin_framein: Out of buffer space I was trying to setup Speex on my second Asterisk server and wanted to

Hallo, i searched archive of list, google.com and found nothing suitable :( My problem is : i would like to rsync only directory /lib/modules/2.4.18-5 and nothing else. As a second condition i'd like to use rsync server at source side and suck files to the destination. The server module, let's name it [test], have "exclude from =/etc/exclude" and "include from

Hi, I would like to shape incomming traffic on eth0 and ipsec0 (binded to eth0). I need to set minimal bandwidth to some packets going via ipsec0 interface. It is running fine when I simply mark the ESP (protocol 50) packets in the PREROUTING chain - means all ipsec packets are shaped. Like: iptables -t mangle -A PREROUTING -i eth0 -p 50 -j MARK --set-mark 30 iptables -t mangle -A PREROUTING -i

Hello, it is possible to get MWI working with Grandstream and Asterisk? Thanks. -David

Hi, I'm trying to configure Sipura 2000 (behind NAT) which connects to Asterisk (public IP, no NAT) and having interesting results. When Sipura is behind Linux/NAT firewall it works great and no special NAT settings on Sipura are necessary. The issue I'm having is when Sipura is behind Linksys broadband NAT router. Sipura gets registered with Asterisk just fine, but I can't hear