similar to: help with syslinux config

I've just gotten pxelinux working (very cool) to boot a Dell PowerEdge. Everything is great, except for one bit. I turned on the IPAPPEND option, and when the machine boots (configured for nfs-root, IP kernel-level autoconf with no options) it says: "ip-config: incomplete network configuration information". According to the syslinux.doc file, the ipappend option appends a line to

Hi, Rsync is a fantastic program, and it does everything we need in terms of file transfer and syncing source and destination file directories. We do have a situation now where we need rsync to transfer the files once only from the source directory. Please let me briefly explain using a simple generic exmple: 1. At t0, we transfer f0 location A to location B 2. At t1, 10 new

Hello everyone, I have a problem regarding gre tunneling, I have two linux box both of them has a private network and the linux A is connected to the internet via wireless radio and the other linux B is connected to the internet via lease line. Here is the setup of my two linux box Linux A eth0 = 203.189.x.1 (internet) eth1 = 192.168.1.1 (going to hub private network) Linux B eth0 = 205.198.x.1

I've got IAX2 setup between two servers with this config: I have two servers on a switch: asteriskm is 192.168.0.160 and asterisk1 is 192.168.0.161 asteriskm has a Sangoma T1 card in it. I want to route calls from asteriskm to asterisk1 which will run an AGI IVR for the call. Config is below, but my problem is that 90-95% of the time when I start asterisk on the two servers I get the

When attempting to join my domain, the NT 4 Workstation and XP Pro clients cannot contact the domain controller. The Samba server is running normally, and can be connected to via IP address, but not by name. Additionally, when I set up a DNS, it still could not contact the Samba server. The clients and server are on the same subnet. I have read as much as I could find on configuring Samba as a

Hi, I have installed a CCNP test application that came with a CCNP book but the actual text of the questions is missing. The wine version is 1.2.2, the OS is xubuntu In the quiz preferences the two font options are arial and times new roman I am guessing that it just can't find these fonts, but i am not exactly sure where i should be putting them. I have installed MStruetype fonts but

Hi all: I am pretty new to xen, but ,thanks to you, I am learning fast :) I have a Dell Poweredge 2950 server running Debian Lenny as Dom0 named Asgard I have configured three DomU (Loki, Hermod and Thor) with DHCP support. All the machines (Dom0 and DomU''s) ask for an IP address to a DHCP server in the Office LAN. The DHCP server is configured to bind certain IP addresses to the Dom0

client with my asterisk. If i am wrong, please let me know On Wed, Jan 7, 2009 at 4:43 PM, Rodolfo Alcazar Portillo < rodolfo.alcazar at padep.org.bo> wrote: > Missed the thread, sorry. Gizmo5.com has some blackberry SIP clients. > Could be what you want. > > Greets! > > Am Mittwoch, den 07.01.2009, 16:07 -0500 schrieb Eric Moniz: > > TianLun, > > > > I

I need to accept an incoming call, make a series of outgoing calls, and once I find someone willing to accept the call, bridge the original incoming call to the outgoing call. Using Dial from an AGI script isn't enough because once the Dial'ed number connects, the call is immediately bridged and I need to ask the called party if they will accept the call. I can see a couple of

We have a server with a TE120 on a partial PRI trunk that several times a day declares the PRI trunk down and stops handling calls until the asterisk is stopped, the zaptel/te120 modules reloaded, and asterisk started. Just before things go down, the log shows the following error: ERROR[9424] chan_zap.c: Write to 28 failed: Unknown error 500 at which point a "show pri spans"

I have been struggling over central provisioning for quite some time. I have eagerly watched each post with like problems but have yet to find a reliable answer. I have a Polycom 501 and I am trying to provision from an FTP server, and just to take any routing out of the issue it is on the same subnet. I am running the 2.0.1.0291 firmware and 3.2.2 bootrom. I set the IP info on the phone and

Hi. I've spent the past four hours on this... if it's a FAQ, I apologize. I am setting up a system with Asterisk 1.4, a TDM400 (3 FXO, 1 FXS) and 3 Aastra 480i CT phones. (1.4.1 firmware.) I have the system mostly working, but am still having trouble with a couple features. This email will deal with parking. First, I think I should point out that transferring calls seems to work

From c|net News: "On Monday,Microsoft and nine leading phone manufacturers--Asustek Computer, GN, LG-Nortel, NEC, Plantronics, Plycom, Samsung, Tatung, and Vitelix--announced the public beta program for Microsoft Office Communications Server 2007 and Microsoft Office Communicator 2007." http://news.com.com/8301-10784_3-9719931-7.html?part=rss&subj=news&tag=2547-1_3-0-20 --

We have a customer with an obsolete Centigram voicemail system who would like to replace it with Asterisk. Any one with experience doing this or information on the signalling and trunking used to connect the Mitel SX-2000 to the Centigram server? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.digium.ca

Attempts to perform a "make all" in /usr/src/asterisk/channels/h323 fails with countless errors of the form: /usr/src/pwlib/include/ptlib/ptime.h:152: macro or `#include' recursion too deep In file included A "make all" using the stable branch builds with the same pwlib code but of course the h323 code in the stable branch doesn't work. So it seems those of us who

We're having a problem with an E1 trunk in Mexico into an IVR server and would appreciate any suggestions. Hardware: Digium TE110P jumpered for E1 zaptel.conf: span=1,1,0,ccs,hdb3 # clear=1-30 bchan=1-15 bchan=17-31 dchan=16 loadzone = us defaultzone=us Circuit status is fine: Status: Provisioned, Up, Active Calls are accepted by Asterisk without any

We're trying to use some AudioCodes MP104 FXO units as gateways to Asterisk but cannot get them to reliably detect DTMF. Some landline calls get most digits but some are duplicated. Some cell phone calls get 0% DTMF recognition. Anyone with experience with these units have any suggestions? ABP Technical Support has been unable to diagnose the problem and is now sending random guesses and

Trying out SVN-oej-t38passthrough-r12677 on a server that also needs to pass some calls to another using IAX and attempts to use the Dial command results in multiple messages "Out of idle IAX2 threads for I/O, pausing!". Since this server needs to support IAX I'll have to back out this version and find another idle server to use to play with the T.38 code. g. -- George

Please tell me the obvious mistake I'm making here. (And yes, I well know about NAT and one-way audio problems in general.) I want to try the new T.38 passthrough stuff, downloaded it, built it, tested it with an SPA-2100 and can hear announcements fine but echo test shows no audio outbound (i.e. SPA to Asterisk). Registered the second SPA-2100 channel with an Asterisk 1.0.10 server in

I wrote earlier: > Please tell me the obvious mistake I'm making here.... The problem was a lack of sleep. Sorry to have troubled the list. -- George Pajari, netVOICE communications 604 484 VOIP (484 8647 x102) Open Source VoIP/Telephony Specialists 1 877 NET VOIP (638 8647 x102) www.netvoice.ca www.ip-centrex.ca www.digium.ca www.grandstream.ca