similar to: call limit

Hi I have tried to setup freebsd (with samba3 ports), backend is openldap with smbldap-tools. After a windows client joined the domain, I tried to right click a folder to set Security/permission. The windows client has a popup windows which display: The program cannot open the required dialog box because it cannot determine whether the computer named "DOMAIN" is joined to a domain I

Hi, I'm wondering if someone can share their thoughts on how to implement a system that periodically checks active channels which have been up for more than X minutes and plays/injects a sound file. The idea is to simply warn users that they've been on the phone for quite a while and maybe they should consider hanging up. If the call stays up for more than Y minutes, it is dropped

I am trying to find out the best way to replace one of our hardware PBXs. It currently has 200+ analog phones connected to it. The idea is to take advantage of the already installed phone cables (big building) so I'm trying to avoid the use of ethernet adapters (if possible). However, I'm realizing that it's an expensive setup and will definitely require two or more cooperating

Hi List; I compiled Zaptel 1.4 and Asterisk 1.4 after downloading them using svn, but when I checked the file zaptel.conf under etc/asterisk, I did not find this file. Any help? By the way: How can I know the asterisk and zaptel version extactly that I compiled them? In other words, asterisk 1.4.... and zaptel 1.4.... ? Regards ------------- ITS IP Telephony and Contact Center Engineer Eng.

I setup two asterisk servers with identical settings (same extensions, same queues, etc). Each one is connected to the same amount of incoming/outgoing links (1 PRI, 4 BRI, 1 IAX friend, etc, on each box). Most extensions are sip and they register via DNS SRV and other methods so that the two servers are load balanced. Incoming PSTN calls (BRI) reach 50% each server so that's load balanced

Hi, I would like to know if one can set the outgoing caller ID within Asterisk when calls are going out through: 1) an analog POTS line (I suppose not) 2) a telco BRI line (I don't think so) 3) a telco PRI line (maybe) 4) a voip provider (surely) Thanks, Vieri ____________________________________________________________________________________ Moody friends. Drama queens. Your

--- Harry Sack <tranzedude@gmail.com> wrote: > 2007/9/8, Josh Coalson <xflac@yahoo.com>: > > > > it actually is complicated. the libFLAC api is not suited to a > > multithreaded design because the i/o is stream-based, not file- > > based. flac(.exe) is the file-based wrapper around libFLAC that > > allows it to work on files. the way libFLAC buffers

I realize this may be slightly off-topic but I'm wondering if someone here can lend me a hand. One of my GXW4008 has gone "unconfigurable" via standard HTTP (refuses connection) and I can't use the built-in IVR because I had previously disabled the "keypad update" feature. So I'm stuck with just telnet, the reset button and RS-232. Telnet commands are very limited

I use Centos 4.2 with all service pack installed. I verified traffic on link WEB and I see port TCP 80 with many traffic. I accessed lod /var/log/httpd/access_log and show below. ca.com/members/index.php HTTP/1.0" 401 - "http://members.sapphicerotica.com/members/index.php" "Mozilla/5.0 ( compatible; MSIE 5.01; Windows XP; NetCaptor )" 68.119.110.138 - -

it actually is complicated. the libFLAC api is not suited to a multithreaded design because the i/o is stream-based, not file- based. flac(.exe) is the file-based wrapper around libFLAC that allows it to work on files. the way libFLAC buffers data is also impossible to parallelize without significantly changing the api. it would take a specialty file-based encoder using an independent frame

Hi, I have a Linux server at ip address 10.215.144.7 running DHCP, TFTP and syslinux. DHCP config contains the following: next-server 10.215.144.7; filename "/pxe/syslinux/pxelinux.0"; and the 'default' pxelinux.cfg contains: LABEL altiris ??? MENU LABEL ^7. Altiris ??? COM32 pxechn.c32 ??? APPEND 10.215.144.60::/BStrap/x86pc/BStrap.0 When a PXE client boots in my network

Hi, I'm probably missing something trivial but I don't understand what. Asterisk is loading fine but when I connect to the console (asterisk -vr) and type "stop" I get a no such command reply: *CLI> help (...) skinny show lines Show defined Skinny lines per device soft hangup Request a hangup on a given channel unload Unload a

Hi, Suppose you have 2 identical Asterisk servers and 1 alias IP address that you assign to either one, according to system failures, etc. Also suppose that all SIP clients register requests go to the alias IP address. Imagine server1 fails and server2 gets the alias IP address. Correct me if I'm wrong but I would have to wait at least 60 seconds before most SIP clients re-register to

Sorry for top-posting but my webmail forces me to. I added -W to the APPEND line as suggested but I'm still getting the same result: Booting... Altiris, inc. X86PC PreBoot, PXE-2.x Enhanced Build ID=402 PXEPreZero: Invalid PXE Server list format. and the client PC freezes right there. Here's the full content of my dhcp.conf: max-lease-time 86400; ddns-update-style interim;

Hi List; If I set nat=yes, then asterisk will send the packets to the public IP address or to the private IP address (which will be for the endpoint that is behind the nating)? And by setting the nat=yes, then what exactly will be ignored at asterisk side when reading the registeration messages from the endpoint? Any help. Regards Bilal

Hi there! I remember reading that you cannot reload new zone files on the fly and require a full restart of the nsd daemon? We are evaluating multiple DNS servers that have better performance comparing to bind, but will require quite heavy zone reload (new and existing) every 10 minutes or so. Downtime (even 1-3 secs) is not the option. Thanks!

Hi, I'm experiencing "frequent" kernel panics on a system with Asterisk 1.4.26.3. There is no core dump, "just" a kernel panic. This is the only data I could copy from the screen: EIP: 0060: [<f8e248b4>] Tainted: P VLI EFLAGS: 00210297 (2.6.23-gentoo-r8 #1) eax: 00000130 ebx: 00000000 ecx: 00220028 edx: 00000978 esi: 346e5802 edi: 00000000 ebp: c3b45500 esp:

Hi, Is Asterisk "fully QSIG-compliant"? I currently have an Alcatel 4400 connected to Asterisk 1.2 and 1.4. Zaptel versions are 1.2.26 and 1.4.11. I am using switchtype=euroisdn and all works fine. However, it seems that Alcatel's latest firmware has dropped support for euroisdn which is really despicable. So now I need to see if I can migrate to QSIG which is supported by

I'm trying to understand why vendors keep making 100Mbps integrated 1-port switches in their hardware SIP phones. Even the recently-announced D40 and D50 Digium phones are limited to 100Mbps. Only the more expensive models (like the D70) can run at 1000Mbps. However, you can't expect a firm with hundreds of extensions to buy the most expensive model... And gigabit speed is important when

Hi List; If I am configuring Diguim Analoge card, then I need to run #modprobe wctdm, but the question why I need to run #modprobe zaptel also? What #modprobe zaptel does a things that #modprobe wctdm does not do? Any help? Regards Bilal ____________________________________________________________________________________ Looking for a deal? Find great prices on flights and hotels