similar to: Frequent disconnects when relaying

Hi, I'm using SIP users of the form 'ab_12345' (two letters, underscore, 5 digits). ChanSpy is working fine for listening in to conversations initiated by these channels, and I can use '*' to randomly switch channels. However, is there any way in this scenario to be able to switch ChanSpy to a specific channel by typing in a ...# key sequence during a spy session?

Hi, I'm using IAX2 between our SIP and PSTN servers, both running Asterisk 1.6.2. Users connect to the SIP server and dial; the SIP server forwards the call to the PSTN server over IAX2, which then dials out over the connected PRI. Since users need detailed call progress feedback, the first action in the dialplan on the PSTN server side is Answer(). In this scenario it's easy for

Hi, Is doing this as simple as just creating a queue in queue.conf? I have the following setup: 1. Single hunting 1-800 number mapped to multiple numbers in a hunt group by the telco. 2. All calls land up at Asterisk over SIP. Now I need: 3. A hunt group of operator extensions mapped to the incoming lines. 4. Music-on-hold if all the operator extensions are busy. Queues seem to handle this

hits=-2.5 tests=BAYES_00,FORGED_RCVD_HELO X-USF-Spam-Flag: NO Hi, I have a requirement to scan Apache logs and discover ``exceptions''. Exceptions can be of two types: 1. A single IP generating a large amount of traffic within a given time frame (for definable values of ``large'' and ``time frame''). 2. A single IP hitting a wide set of URLs on the server (indicates

Facing a strange problem with Samba 3.0.0 in a Windows 2003 domain. The net ads join works just fine, but a testjoin shows that the domain hasn't been joined and winbind user/group enumeration services, etc don't work after that. Kerberos is working fine too independently. Time has been sync'ed. Tried with both server signing auto and server signing off. The error message is:

Hi, I''m trying to route multicast between 2 LANs using a GRE tunnel. The setup is: B to C is a GRE tunnel running over PPPoE. B is the multicast router. A is on the same LAN as B. The same ethernet adapter is used for both the LAN and PPPoE connections. I''ve setup smcroute since I only need static routing. Multicasts sent from C (the remote machine) to B over the

Hi, We're using Samba 3.0.0 beta2 with a Windows 2003 domain controller at a client's site. OS: Vanilla Red Hat Linux 8.0 on i686 (no patches or updates). Samba compiled from source into RPM (makerpm.sh) and upgraded from version 2. --with-acl-support enabled. Filesystem: SGI XFS (kernel 2.4.21) from CVS. Using Winbind to provide ID mapping from Windows users and groups to Unix users

Hi, After spectacularly failing to get HTB to proportionately distribute excess bandwidth into multiple classes based on the rate, I decided to hack my requirements. Now I''m creating multiple classes with the sum of the bandwidths of the classes being equal to (OK, slightly less than) the link bandwidth. Scenario: Linux Router +------+ 256K

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I''ve written a minimal sort of Perl module that dynamically creates and destroys traffic control rules for specific IPs. I''m currently using it for a user bandwidth control application at a client site. The module essentially gets Ethernet device(s), IP address and in/out speeds as input and dynamically creates classes, queues

Hi, I have a client connected to the ''net through 3 ISP''s. Have set up a Linux box to do routing and load sharing for the 3 connections. A fourth interface is connected to the LAN with private IP addresses. Am using iptables to SNAT traffic to the appropriate IP depending on the interface the packet gets routed onto. The setup looks something like this: Interface IP

Hi, Treated myself to some new hardware as advance Eid/Diwali/Christmas/New Year gift, and was trying to install Xen and get virtual machines running on that. The path was long and arduous, and is still not complete, but some of my experiences on the way are documented in: http://wiki.kandalaya.org/cgi%2dbin/twiki/view/Main/DebianXenMigrateVirtualBox Covered are: - Basic Xen setup -

Hi, Continuing with the saga of Digium vs MTNL Mumbai, looking for suggestions on handling incoming Caller-ID issues. The card manages to grab a couple of (random) digits of the incoming CID, but they're more or less useless. Is there any way to fix this? Asterisk 1.8.13, Dahdi 2.5.0.1 on Debian Testing (Wheezy), MTNL Mumbai. Digium, Inc. Wildcard AEX410 4-port analog card (PCI-Express)

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body bgcolor="#ffffff" text="#000000"> <font face="sans-serif">Hello all,<br> <br> I've googled

Hi, Would you recommend some standalone SIP phones that work well with Asterisk? Personal experience preferred. Thanks, -- Raj

Hi everybody! I''m trying to implement the load balancing for a LAN with two ISPs. I''ve installed a Suse Linux Enterpise Server 9 with iproute2 for that porpouse. The server have two NICs, one of them is for both the LAN and ISP 1. I''ve setup both NICs with YAST (if I use ip for this, then the whole thing doesn''t work!) and execute the following commands to

Just a nifty thing to have for icecast server access logs... diff against CVS HEAD is attached. <p> Index: configure.in =================================================================== RCS file: /usr/local/cvsroot/libshout/configure.in,v retrieving revision 1.65 diff -u -r1.65 configure.in --- configure.in 13 Jul 2003 16:36:37 -0000 1.65 +++ configure.in 21 Jul 2003 04:20:22 -0000 @@

Current (and even icecast 1.3) icecast strftime() format option is to specify "%z" for gathering the timezone value for use in the logs. This isn't POSIX. The true token for standard strftime() timezone value is %Z. This patch fixes that. Without it, your log entries *may* look like this: 66.93.53.158 - - [21/Jul/2003:00:36:57 %z] "GET /full-on HTTP/1.1" 200 31554203

Dear Tariq; About elastix.org, this can be use with Asterisk or it is coming as a complete IP Telephony, Call Center, IVR and Reporting? Because, I do not need to install another IP Telephony on the server which already has asterisk which is an IP Telephony, this will cause a problem in the service (for example, when listening for SIP port of 5060).

Hi, I'm seeing the following message in the full log: WARNING[478] asterisk.c: poll returned < 0: Bad file descriptor it's repeated a number of times then I'm disconnected from the running asterisk instance. What's the best way to correctly report this?

Hi, I have a linux box which balances load between two interfaces ( say WAN1 and WAN2). I have masquerading on for any request coming from LAN to the outside world. The setup is in such a way that WAN1 drops packets with source ip belonging to WAN2''s network and viceversa. For some strange reason, I find that packet coming out from the WAN interface has source address of WAN2 and