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I've noticed that some callers listen to our main menu and don't press any keys. I have it set up to restart the menu a few times and eventually hang up. I'm wondering if these are wrong numbers (in that case, why don't they hang up) or they really want to speak to someone here but don't understand the menu (what's so hard about "for the operator, press

I have a Linksys PAP2. Identical setups for the two channels in both the unit and in Asterisk. In particular, both channels enable g729 and set it as the preferred codec, and have disallow=all and allow=g729 in sip.conf. If we make a call on one channel, it works (and uses g729), but if we make a call on the other channel when the first one is still connected, it fails. We have three g729

I'm not having any trouble with interrupts, but here's my /proc/interrupts on Fedora Core 2 on a hyper-threading CPU and using the SMP kernel (2.6.5-1.138). I don't think I need to worry about uhci_hcd, nothing is plugged into USB, but libata is the disk driver. How do I get libata and wctdm to use different interrupts? $ cat /proc/interrupts CPU0 CPU1 0:

I am trying to put together a list of 'departments' to request as voice prompts. I have the biggies (sales, accounting, shipping, etc...) but I want to make sure I do not miss any. If anyone anyone has some suggestions (Ha... that is like going to an NRA meeting ans asking if anybody has a gun :-) ) please forward them to me (and / or post here although, with the volume of this

When I first saw the priority numbers in extensions.conf, I thought BASIC, if a number is missing, * will fall thru to the next number. I learned that this is not so, if you have nothing between 1 and 3, you don't ever get to 3. But I'm wondering what does happen? Hangup and wait for next offhook? Undefined?

I think I've managed to figure out that there are two ways to transfer a Zap call, using hookflash (defined in zapata.conf) or the # key (the t and T options of the Dial command in the dialplan), but not why there are two ways to do this, nor what the difference is between them. Is there something that explains this? thanks -------------- next part -------------- An HTML attachment

I have: RedHat 9.0 TDM40B asterisk-0.9.0 compiled from sources zaptel-0.9.1 likewise /etc/zaptel.conf contains fxoks=1-4 loadzone = us defaultzone=us loaded modules zaptel and wcfxs /etc/askterisk/zapata.conf contains [channels] language = en signalling = fxo_ks context = phones channel => 1-4 /etc/askterisk/extensions.conf contains [general]

The FXO lines on my TDM400 are connected to PSTN lines in Israel. As far as I know, the lines around here have disconnect supervision (I've seen some other Israelis on this list, anyone know for sure?), because it's worked on Dialogic cards, which reported hangup, not busy detect (while when I connect a Dialogic card to a PBX, I have to measure the busy signal's frequency/cadence or

We are using Bellsouth 8867 phones on our TDM400B FXS lines (asterisk-1.0.3). It has a "Voicemail" light, which appears to be MWI (according to the manual it works with voicemail from the telco that sends a FSK signal). The dialtone stutters when a line has voicemail, so I know that I have the mailbox setting right in zapata.conf, but the light doesn't go on. I am also getting

I'm going to be hooking FXS lines on a TDM400 to a PBX which doesn't drop line voltage at the end of a call, so I'm going to have to use busy detection. A few questions - The tones are taken from the tones specified by the zone in zaptel.conf, right? Which tones cause hangup? The PBX may not use the national standard tones. Does anyone have any suggestion for how I can

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I connected two plain old telephones to FXS lines of a TDM400P (defined as fxoks in zaptel.conf), one of them dials the other with Dial(Zap/2), I talk to myself for a while, hang up either of the phones, and the phone that remains off-hook gets the congestion tone until it goes on-hook (at least as long as I've cared to wait). I don't have a voltmeter on the line, but if I'm hearing

I thought I had posted this, but I didn't see it in the archives, so I guess I hadn't. I've got FXS lines going to a legacy IVR. When I Dial into one of these lines and then hang up, FXS plays the Congestion tone until the IVR drops voltage. I would like the IVR to hang up sooner. I could do this by either making the IVR recognize the standard Congestion tone, or changing the

http://lists.digium.com/pipermail/asterisk-users/2004-September/061677.html says "If you don't have a soundcard then try just loading the sound module, that might just be enough." I don't know what that means. What sound module? In Asterisk? In Linux? Yes, I have mpg123 0.59r

I have more FXO ports on TDM400's than I have PSTN lines available for testing. When all the lines were used up (the FXO ports are all in zap group 2), and I did a Dial(Zap/G2/${phone},5) it thinks the Dial succeeded even though there is neither line voltage nor dial tone. Can at least the lack of voltage be detected? It would be good in case one of the phone wires fell out that it would

Do you have threewaycalling and transfer set in your zapata.conf? Here's mine (four TDM400's, seems to be working so far). I didn't do anything in my extensions.conf for any of these features (what confused me at first is the t and T options of the Dial application in extensions.conf are for transfers via the # key), when you flash you get another dialtone that works just like the

Rich Adamson wrote: > How old of code are you looking at? The wcfxs driver was renamed to wctdm > some time ago. Current cvs doesn't include it. I was using zaptel-1.0.0, but I took a look at 1.0.3 and it's still wcfxs there. I'm about to bring this online, would rather stick with releases.

I moved my TDM400B cards (first two cards are 40's, third is a 31, last is an 04) from one computer to another, copied all the config files, and now the LED on the line 11 - third line of the third card doesn't go on (it used to on the previous computer). I can get by telling * not to use this line for now but does anyone have any suggestions for getting it to work? Unseat and

Here's a strange one - when I run safe_asterisk on either of these distros, words that are colored blue or violet (but not red) turn up in Russian (and some other languages, I think). If I run asterisk with the same arguments (-vvvg -c) as safe_asterisk does, from the console, it's OK. If I run it in a Putty window it's OK. If I run asterisk -r from another console or from

Recently I put callwaiting=yes in zapata.conf because customers want to speak to the operator in person, not leave her a voicemail, when she's busy with another caller. But now she can't transfer either of the calls (which she can do when there's only a single call). The operator has an analog phone connected to a TDM400B FXS line. The calls are coming from PSTN lines connected