search for: brown75

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 So a friend of mine and I setup a static key based point to point OpenVPN connection from my box to his for the express intent of carrying IAX traffic encrypted. His network on his lan is 172.30.1.0/24 and mine is 10.0.30.0/24. His PBX is located at 172.30.1.48 and mine is at 10.0.30.2. We had an existing working IAX trunk in place prior to the VPN,

I'm having trouble with ztdummy and I can't seem to figure it out. I am running Zaptel 1.4.12.1 under Debian 4.0 with latest updates applied and I have compiled Zaptel from source along with a new kernel from Debian sources to include 1khz timer support. The modules build fine, yet when I load them I get the following output from dmesg: rtc: lost some interrupts at 1024Hz And then

Ok so this is officially driving me crazy. I have an Asterisk 1.6.1.6 install with DAHDI/DAHDI tools (latest) and an OpenVOX TDM400 card with 1 FXO port and 1 FXS port. I have a POTS line from my phone company attached to the POTS line. I have asked for "disconnect supervision" to be provisioned on my line and they claim to have added it. However, my scenario is as follows: I receive a

I''m playing around with VLAN''s and I have a VLAN capable (layer 2) smart switch. I see a steady stream of martians in the logfile if I have the routefilter option set on the loc zone interfaces in /etc/shorewall/interfaces. I have two interfaces in the loc zone, eth1 and vlan2 respectively. vlan2 is an 802.1q trunk going towards the switch. Is this the expected behavior in

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Anyone know of any Asterisk 1.8 deb's available or when they might be included in backports or (hopefully) Squeeze? I can compile from source... but would much rather have a pre-packaged binary if one exists... Thanks, Stephen -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (Darwin)

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I want to have the ability to have anonymous SIP calls hit my server but I want to run it on different ports and create an SRV record for my target domain. My understanding of SIP is limited, but in a nutshell I want to accomplish the following: - - run SIP signaling on port 6200 - - create RTP ports on 6201-62XX Do I really need 10k ports open for

I'd like to start playing with 1.8, however I don't want to potentially damage anything on my existing 1.6.2 install on my production server. I'd like to test 1.8 against my existing configs leaving my 1.6.2 install untouched. Looking at the output of ./configure --help suggests that it's possible to install Asterisk into another prefix of my choosing, but as this is

I am running Asterisk 1.6.1.6 and DAHDI/DAHDI tools 2.2.0.2 compiled from source on a Debian Lenny box, also running FreePBX 2.5.2. I also have an OpenVOX TDM400 card installed with an FXO port on port 1, and an FXS port on port 2. I have a POTS line installed and working on the FXO port. However I've encountered a weird problem that I can't seem to figure out. I have incoming POTS

I've got a piece of CPE equipment that has an FXS port that I have tied to an FXO port on a TDM400 clone card. Normally, if I go off-hook with a standard telephone connected to it, I get a dialtone. If I dial a digit, and send a hookflash, the device will provide a dialtone back for the next available channel on the device. I'm trying to recreate this same behavior with Asterisk,

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I''m running Shorewall 4.4.0/Debian Lenny and I''m trying to setup OpenVPN with a mild degree of success so far. My ultimate end goal is to basically have an extension of my home lan to my laptop as well as my wife''s when we are away from home, and have all of my normal network resources available as if I were sitting at home