similar to: Change IP info.

I have a Smoothwall server on my network and am running three network interfaces off it. 1) local LAN 192.168.0.0 with PCs and an internal dovecot server on 192.168.0.154. 2) internet interface 3) DMZ 192.168.2.0 which has a linux web server 192.168.2.1 on which I want to install a webmail so I can access my email remotely. Originally 192.168.2.1 couldn't see the 192.168.0.0 network but with

Hi all, Is it possible and easy to make a CTI server with Asterisk? Florent,

Hello everyone, I'm using a TE410P, no irq sharing, and all extraneous devices disabled, such as USB, Parallel etc. I'm getting a few IRQ misses according to ZTTOOL. We're running a standard PRI_CPE interface and seem to be getting dropped calls, and errors on the D-CHANNEL occasionally. The circuit itself is very solid, it was in use on our old PBX just a few weeks ago, never

What is everyone using for a firewall? I'm currently using www.astaro.com but their recent releases have soured me on ASL as a practical solution on my hardware (1.2MHz Athlon, 30G, and 256M). I only have 4 computers going through the firewall wall but it's consistently at 50% cpu load. There is very little network traffic (<10k bits per second on the wan connection, < 40k

I'm trying to config a couple of things on Asterisk CVS-03/22/04-16:41:51. The number shows up, but I can't get the "words" to show on a local bell line. The text always comes up as "unavailable". In sip.conf for each extension, I've tried: callerid="VERTEX" <2142618000> callerid=VERTEX <2142618000> Neither one works. Suggestions? On the

Hello! Asterisk box receiving calls. Is there some way to get information about current calls from external or AGI application? I'm interested in: - duration, how long calls already in the system (billing and actual time); - source/destination phone numbers; - etc. In other words can I receive information which we are usually getting in CDRs during the time when the call is still active?

Is it possible for asterisk to do an sql query in order to get the member list of a call queue? thanks micko

Playing with time ranges - using the examples found in one of the asterisk cook books... (pdf - page 17) ; After Hours include => night_menu|00:00-08:00|Tue-Fri|*|* include => night_menu|17:00-24:00|Mon-Thu|*|* this gives... ... pbx.c:2962 get_timerange: 24:00 isn't a valid end time.... -- Including context 'night_menu|17:00-24:00|Mon-Thu|*|*' in context 'default'

I was trying to use TDMoE and I lasted with two problems. First of all I can't configure the dynamic span to use CAS signalling but documentation (by Mark) says that you can use any type of signalling (and this includes CAS I guess). My second problem is related that my Linux system crashes frequently due to ztdynamic and friends. I'm using a 2.4.26 version kernel and zaptel

Are there any variables or structure elements unique to a call that stay till the end of a call -- including when caller enters a queue and then bridged with agent. I am trying to get some variables about the caller in an AGI script when the agent's phone is ringing, and I'm finding not even the queue name the caller just came from can be found. Using the callers wait time in queue would

Hello list, I wonder if this is possible with Asterisk: - While talking through Asterisk, I would like a client to start recording a call by typing, say, #99# I know it is possible to do it using an external monitoring application, but I want to know if it's possible to have Asterisk silently monitoring an on-going call and responding to DTMF tones within it. How do things like call

Hello, I am having an issue with making two simultaneous outbound calls. When I dial, both phones try to take the same channel and it causes an error. Anyone have any suggestions. My set up is as follows: CO - PRI - ASTERISK - VODAVI(pbx). Thanks, Dave *CLI> -- Starting simple switch on 'Zap/69-1' -- Executing Wait("Zap/69-1", ".1") in new

Are there any debugging tools for the digium zaptel cards that would report the activity on the line, such as DTMF and/or connection protocol? I'm looking to debug the connection with a T100P, & I don't have $2000 for a T1 test set. Thanks, Glen

Hello all: We have installed the latest cvx version of asterisk. We have a FXO card on the server. When some one dial that DID we need to forward the call from asterisk to a sip address, example 2222@sip.sip.com. In that sip address an IVR will handle the call. Also is some one dial any phone number from any asterisk extension the call should be sent to the same sip address. We

Hi All, Could anyone tell me which is the recommended hardware to a system running voicemail and conference, attending four E1 trunks and, another, attending only one E1? Can I use a PIII 850Mhz? Thanks in advance. Robert Almeida -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anybody have a good source for hold music? I can see a number of companies on the web that sell royalty-free MOH, but they don't all provide samples. The customer service desk has requested "calming, not sleeping, but calming" and "this is a high-tech company, so make it 'techie' [sic]". Thanks, --Ernest

Im trying to come up with a cost effective way to unite two PBX using VOIP. My idea is that since most companys here (Argentina) are not ready cough up the money to go to full-fledged VOIP, they might be willing to pay for a hybrid-solution: a kind of "point-to-point" line using VOIP, which let's them dial an extension on the other PBX. What i want to accomplish:

I've noticed that when i pass a wait in an exten => that it doesn't allow for dtmf tone input. Also on another note i've noticed that when using gotoif it will also cut the dtmf tones and drop the first part if the gotoif is hit in the middle of input. Anybody else seen this or have this problem?

- - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. "time is the fire in which we burn," Tollian Soran. "grudges aren't worth holding--One who holds them shows his self-weakness." Contact info: hank@hanksmith.net Email: Same as MSN. ----- Original Message ----- From: "hank"

Hi, I am new in asterisk and i've bought a X100p and a TDM400... First of all, how can i verify my config files ? Secondly, when i'm trying to pass a call to the outside, i ve a Notice about appdial.c (l 554) telling me: unable to create channel of type Zap ...and i don't understand... Finally, when i plug my analog phones in RJ45 of my TDM400, there is no tonality ( i'm not