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...20]: loader.c:242 ast_load_resource: /usr/lib/asterisk/modules/chan_oh323.so: undefined symbol: __use_ast_pthread_create_instead__ Aug 15 12:40:00 WARNING[1076245120]: loader.c:423 load_modules: Loading module chan_oh323.so failed! Can anyone tell me how to fix this, or what that mean? Regards Krystian

...lling is more frequent and takes longer. I isolated the problem to be with HTB (CBQ works fine). The script I use now worked for a year without any problems, but since fedora changed to 2.6 the problems started. The problem is with the stock kernel and with a self-compiled one. -- Miłego Dnia Krystian Antoni _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc

Hi All. I'm using RedHat 9 I configured the chan_h323 and asterisk from CVS. This is the scenario SJ_lab_phone(sip) ---------------> Asterisk -------------> H323 GK --------------> PSTN I have tried all codec's and always the same result, the called phone will ring without dropping for how ever I allow it to but as soon as it is answered it immediately gets disconnected.

Hi All I'm trying with no luck to connected the Quintum D series Gateway with the new SIP release to asterisk. Have anyone done this? If yes then how should I configure the sip.conf to accept the registration? maybe a sample config? Thanks /Krystian

Colleagues- I am working on the design of a fairly large samba deployment, and I am looking for feedback on some of my design ideas. I have 10 buildings spread out in and around a city, all interconnected via 1.5Mb leased lines. There are samba servers in each building. I have some users that move from building to building. We are using primarily windows 98 desktops, with a few 2K and XPP

Hello all, Does anyone know of a methodology to build a route based on the inbound and outbound interfaces as opposed to ip addresses? We are essentially trying to forward packets from one interface to another without looking at the ip address. Bridging (brctl, br2684ctl) will not work in this case as the interfaces use different layer 2 encapsulation (e.g., atm0<=>eth0, or

Is any way to detect changed MAC adresses? Someone taught change MACs peoples in my network and I have problems. E.g. Two computers working on one MAC, and one IP (static ARP and DHCP). WinXP is screaming some message... that two computers or more have the same IP. How can I find out who''s changed MAC?

Hi Is there a program which allow me to see how "my" traffic goes through my iptables rules? Which accept it, which deny? Right now my router has a little bit of traffic and its hard to see only mine traffic. -- Miłego Dnia Krystian Antoni _______________________________________________ LARTC mailing list LARTC@mailman.ds9a.nl http://mailman.ds9a.nl/cgi-bin/mailman/listinfo/lartc

Hi, How can I set equal bandwidth of 512kbit downlink and 256kbit uplink for every single IP address of 254 IP addresses I have in my LAN? Regards, ro0ot

...Platform: Other URL: http://samba.anu.edu.au/ftp/rsync/rsyncd.conf.html OS/Version: All Status: NEW Severity: normal Priority: P3 Component: core AssignedTo: wayned at samba.org ReportedBy: bugzilla.samba.org at krystian.net QAContact: rsync-qa at samba.org In 'man rsyncd.conf', precisely the same version as reported online here: http://samba.anu.edu.au/ftp/rsync/rsyncd.conf.html In section: MODULE PARAMETERS -> auth users The section states: "See also the "CONNECTING TO AN RSYNC...

Hi all I have asked this question before but have not got any helping input. I'm really new to this and need some explanation about ASTCC. So here is the question again. In the ASTCC web admin there are Trunks, Routes, IAXFriends, SIPFriends, Brands, Cards. As I understand Brands is not used, Cards just makes the cards. Routed in the dialplan and pricelist, Trunks is for ASTCC to

Folks, I''m so surprised what happened to my box just in the few weeks lately. Here is my setup: INTERNET <----------> [eth0] SHAPER-BOX [eth1] <----------> USER-FARM both eth0 & eth1 got public ips (202.x.x.x) Why traffic monitored at eth0 is bigger than eth1 ? eth1 shaped just exactly the same as rate i defined in HTB. I just have 1024Kbps from my ISP and i defined the

...alling the CVS asterisk it was missing all modules like SIP etc...etc...or config files so I newer continued Does anyone know how to get this to work or have a idiot-proof guide how to set up asterisk on linux with voicemail, SIP, H323 and other useful modules as I'm not that good at Linux /Krystian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040806/9c41a563/attachment.htm

Hi, I am trying to control bandwidths of the connections by using TC with HTB. But I want to implement Leaky Bucket on this structure so as to have a straight flow. Due to being new in this area, could anyone help me how I could do this? Does TC support leaky bucket? Best regards, Fatih Duzova

Hi all, i need advice how can i limit ip_conntrack per IP. clients of network that i support often uses torrent , DC++ , eMule clients and i have lost packages because they open too many ports. i have traffic control limits but this obviously isn''t enough Any advance how to prevent server from this kind problems will be welcome. Best regards Emil

I am looking for a simple way to guarantee to each flow going through my traffic control point 1/numflows of bandwidth. I thought using SFQ would do this effectively but it appears to be quite unfair: a very high speed download that fills the pipe easily starves smaller flows to the point where it becomes unusable (especially if they are at all interactive) Because numflows is dynamic,

I get an echo when going from a SIP phone to a PRI trunk. I hear the echo on the SIP phone. From reading some other post I think that I need to tell me phone company to turn on echo canceling. If the echo was on the other end than it would be my problem? Is this right? What exactly should I say to my phone company so they know exactly what I'm talking about? -- Michael Sampson

...39;SIP/sj1-4ff7' The first Dial is normal but the 2nd Dial "Executing Dial("SIP/sj1-4ff7", "H323/h") in new stack" Where do that come from? In the h323.conf I only have exten => _.,1,Dial(H323/${EXTEN}) Anyone, please help I'm totally stuck!! Thanks Krystian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040812/84fd95db/attachment.htm

Hi All I seem to have a small problem with the music on hold button on SJPhone. I have 2 asterisk installations one from the Rapid distribution and one from the latest CVS. On the rapid dist when I press the music on hold button on my SJPhone I get music on hold. When I do the same I get no music on hold just silence. I create extension like this exten => 1111,1,MusicOnHold(Default),

Hello Everybody, I have configured a Linux box that does traffic shaping. Its working wonderfully fine, just as expected...Now i want to measure the bandwidth consumed by each of my hosts....But I dont want SNMP to run on all the hosts[as required by MRTG] I used iptraf on my linux box, but it only measures the bandwidth on interface basis only.... I tried installing traffic-vis, its not working