similar to: No dial tone after installation

Oh geez nobody respond, I guess I have to be more descriptive. I'm install Asterisk and FXO FXS Lite dev kit from Digium everything has installed and I am getting an error message that my sound card is not enabled. I'm using a Dell Optix (whatever). Everything seems to be in order. I connect an analogue telephone to the FXO port? and try to dial. Dial tones work however not dial tone.

I currently working on a project using Locustworld's wireless mesh network and running SIP through the network. I want to install Asterisk as server running sip and Digium FXO FXS dev lite card. Unfortunately I having the time of my life to install Asterisk. The astinstaller does everything perfectly but when I try to configure an analogue telephone to setup voice mail I do not get any dial

I need to find some basic configuration files. Is there a place I can check out how to set up an office using sip telephone and Digium FXO and FXS ports? Don Moskaluk don@moskaluk.com www.moskaluk.com 416 737-8230 Cell 416 614-8230 Home --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.732 / Virus Database: 486 - Release Date:

Yeah, I'm a newbie and am having problems with missing .conf files in /etc/asterisk/ I get notices when I try to run asterisk like: parking.conf is depreciated in favour of features.conf Please rename it. (I'm not getting rename parking .conf to features.conf ???) I am also missing: mgcp.conf enum.conf cdr.conf I using dev-lite Digium board that has 1 fxo and 1fxs port. So far I was

I have installed samba on a linux OS suse 8.1. I have a printer installed locally on the machine and it prints ok from linux. I can see the printer in SWAT. The share print$ path = /srv/printers guest ok = yes browsable = yes read only = yes write list = @ntadmin,mm On the linux machine, I have the following path: /srv/printers The following folders are located at this path W32MIPS/ W32PPC/

Hi, I seem to have a problem with chanisavail and the call limits on sip phones(incoming and outgoing) The problem seems to be that chanisavail when trying create to create channels and hanging them up afterwards screw up the current usage limit on the phones. Example with chanisavail: Phone A calls voicemail (usage now 1) Phone B tries to call Phone A and uses ChanIsAvail in the dialplan.

Hi there, I was wondering how I can use setgroup and checkgroup for perfom incoming and outgoing limitation checks. I've have some users that doesn't what to be able to recieve more than 1 call at a time, and I also want to limit a users outgoing call abilities. Any help would be greatly appreciated. Kind regards Cf --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus

That is correct jim. We will setup a vpn, we also know the ports to open but the real question is if there is any parameter to increase the timeout period when the windows machines decide that there is no "share" at the other end.... Since the remote workers connected to the internet are using slow links, we are having problems with timeouts.... Thanks, -----Original Message-----

Hello List, I just started messing with the settings on a SPA-2000, and it has a really nasty alternative dial tone that I want to make go away. I'm not too hip on how the two numbers interact, so my results haven't been good. (I'm in the US, so I'm bias'ed towards US tones.) Default(I'm ok with this one): 350@-19,440@-19;10(*/0/1+2) OutsideLineTone(Bad, really bad. No,

FROM NT \\firewall.surson runs perfectly WHY??? root@firewall init.d]# ./smb status smbd (pid 12183) est? rodando... nmbd (pid 1397) est? rodando... [root@firewall init.d]# smbclient //firewall.surson/profile -Ucatena%motpock added interface ip=192.168.1.1 bcast=192.168.1.255 nmask=255.255.255.0 Domain=[SURSON] OS=[Unix] Server=[Samba 2.2.3a] tree connect failed: NT_STATUS_BAD_NETWORK_NAME

Sorry some email problem, i have change it for more reliable one. I have try this morning to netmasq 192.168.11.0 (eth1) to 192.168.1.0 (eth0), but it is a mistake. Yes thank you for answering so fast ! I have corrected it, here the new diagram and the new routing table. But it still doesn''t work. From the router i can access to 192.168.11.254 I have add the rules : DNAT loc

Have a problem which can't find solution to on WIKI.. Trying to get * to use UK based indication tones. i.e. british ring, dial tone, busy signal. Have changed the indications.conf file to default to UK. However this seems to have no affect. What am i missing. Am using 1.0.3 stable. Many thanks Andrew. ---------------------- indications.conf [general] country=uk [uk] description =

I looked at the error.log file for icecast and here's what I found [time] INFO connection/_handle_source_request: Source (/) attempted to login with invalid or missing password It looks like a combination of two things, it's login into an invalid mount point and the wrong password is being sent. I still can't figure out how though since I have the correct password in the config file.

you can not ping that address because ICMP is turned off. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com]On Behalf Of Deon Rodden Sent: Monday, July 26, 2004 2:22 PM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Broadvoice problems again Attn: James Greetings, C:\>ping 147.135.8.129 Pinging

I know you're looking to do this with asterisk, however, if you have trouble and want an outsourced solution, try www.openfax.com. You can do the IVR with Asterisk and then write a script that calls Openfax and they send out the fax for you. krw -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Isamar Maia

Hi all, I was thinking about integrating an old PBX with Asterisk and I was wondering some possible configurations. The question is: which is the best way to let the 2 systems interact ? I can imagine some possible scenarios: - scenario 1: I want to use other then old PBX terminations (ie I have to link the 2 systems with some internal number line) In this scenario I could think to give each user

I'm currently testing these phones and so far I'm very pleased with them. I'm using the SIP image (Version: 1.0.0.23 Release: 0032-01-A4). There are a few issues with the speaker not working properly when you dial out or pick a call but this is a known issue and will be hashed out in the next release. I've been working with the developers at Sayson and they have been very

Guillaume Found this in the Network Admin Guide - http://www.oreilly.com/catalog/linag2/book/ch05.html#X-087-2-IFACE.IFCONFIG "Receiver overruns usually occur when packets come in faster than the kernel can service the last interrupt." I think you definitely have a problem with your NIC on the Samba machine. I would swap it out and/or change the cable and see if that cures things.

I ran into a similar problem with one of my machines. I simply couldn't quite understand why it happened. I did destroy and recreate the machine account and it still persisted. I ended up changing the machine name and creating a new machine name for the system. That is the only fix that I am aware of at this time. Regards, Robert Adkins II IT Manager/Buyer Impel Industries, Inc. Ph.

During your planning and migration, DNS character set limitations = require special attention. NetBIOS names used in previous versions of Windows = may contain additional characters that are illegal in DNS (namely, ~!@#$%^&'.(){}, and not supported by any DNS server implementation) or characters (e.g. Unicode characters) that are supported by only a few = DNS server implementations (e.g.