similar to: x100p card - detect dialtone?

i need to know something in the zaptel configuration as it seems i can configure detecting the busy tone and hangup after number of busy tone counts, that was great but the problem is sometimes the pstn line has no dialtone and when i try to make call it continue dialing while not having a dialtone! while it should say "all lines are busy/congested" how can i configure that?? i already

Is there any way to make the X100P to check for a dialtone before dialing? My roommate is tired of hearing random dialing in his ear when he's on the phone (same circuit as the X100P). I know one solution is to just put in a TDM400P for the house, but I'd rather get away w/o having to do that. Thanks, Jared

Hi A quicky - has anyone had success with an X100P in the UK ? I'm seeing some oddness - when I plug the wall socket into the card, I lose dialtone. If I unplug it, it comes back. Does this sound familiar to anyone ? Cheers Nigel Nigel Taylor Technology Director ITAzure Limited Dunn House Warren Park Way Enderby Leicestershire LE19 4SA 0116 286 3016

So I have an X100P, and a TDM10B both working (at least I think they are). The drivers have been loaded and ztcfg -vv shows no errors in the configuration of two channels. When I run asterisk -vvvc and pick up my phone (plugged into TDM10B), I don't gear a dialtone. in phone.conf, I have [interfaces] mode=dialtone format=slinear ... Shouldn't that produce a dialtone when I pick up the

Hi all, Really excited to be using Asterisk and learning about VOIP and PBX's. I'm a complete beginner at telephony but have built and installed Asterisk 1.4.5 and read several of the Asterisk books online and have successfully connected to FWD with IAX2 and to GIZMO using SIP. Just purchased a Motorola Wildcard X100P and installed it into a clone PC running Fedora Core 6. Analog

I have some strange behavior happening with call flow when analog line errors are encountered. This may be due to the way that the X100P detects "busy" signals, or it may be something in the software. Could someone with more in-depth knowledge make a comment on the items below? My dialing logic says "dial local area code numbers out of the analog line, and if the analog line

Hi, I have just built my home Asterisk box into a better PC that became available (still only a P2 350 but it only has to manage 1 analog line).. Anyway I have built it on Fedora Core 1.. I have an X100P and a TDM400P (1 module installed).. These cards were working fine in my older PC that was running my Asterisk at home.. The inbound calls via the X100P to my sip phones are working great..

Hello, I created a new application for myself called Dialtone() by modifing res/res_indications.c file. It can be used as such: exten => s,4,Dialtone(30|${CALLERIDNUM}) exten => s,5,Playback(time-exceeded) exten => s,6,Goto(s|1) It will stutter if you have new voicemail and you have passed the mailbox number as I did above. It will stop dialtone the moment you press a key

Hi, I suspected that I the analogue phone should have got a pass through signal when the power was off to the server, unfortunately it doesn't. I kept asking digium support about that but they didn't give me an answer. The problem is how do I identify whether the X100P is incompatibel with the network or faulty without possibly wasting another USD100??? Aaron On Sat, 2004-05-15, Eric

Slightly off topic, but maybe some can suggest something off list... Trying to upgrade a 7960 that was running skinny. I've got sip v3.2.0 installed and running, and am able to place calls via *, etc. However, when upgrading to v4.4.0 I can never get to the point of being able to place a call (eg, no dialtone, etc). I can ping the phone, look at the Network Config, etc, but I can't

I'd like to write a snippet of dialtone that plays dialtone and collects a specific number of digits into a variable. Sort of like READ but with a generated dialtone. Naturally, I want the dialtone to stop playing after the first digit. I can't find this anywhere. Only thing I can think of is a no-password DISA. Is this the correct method? Is there a better one? </edg>

PROBLEM: My x100p only does bad things. When I plug the line from the wall into the card, other phones in my house go nuts, no dialtone, crazy clicking, random tones, etc. Following the steps below, the line is screwed up any time I test after step 2. Just plugging the line from the wall into the x100p takes my other phones off line. External calls to my line hear busy. Obviously, can't

Hi, I have two asterisks connected via IAX2 trunk. The first * use dial prefix 2XX, the second one 3XX. Calls routing works OK. But I don't know how to get dialtone of remote asterisk pbx. I'd like to get dialtone of asterisk #2 after dialing 3 and dialtone of asterisk #1 after dialing 2. I know something about DISA but I'm not sure if it is a right way. Can you give me advice?

I've been looking around for a small external sip fxo gateway, sending emails to possible vendors, etc, and can not seem to come up with anything that fits. Suggestions anyone? (No channel bank & T1 card suggestions, please. I've also just completed an eval of the Mediatrix 1204 which does not support the requirements.) The market between two fxo pstn lines (pair of x100p's) and

Hi List, I cant seem to find the setting that changes this! You put a person on hold, they are on hold like normal, but after a few seconds the phone will then start having dialtone coming from the speakerphone, really weird!! Anyone know how to fix this? I see where it could be nice, but in this case, we just want them on hold is all, no dialtone! Any help would be great! Thanks! Ron

Hi, I see applications for signalling busy, congested, ringing, progress etc, which I understand can be provided either in or out of band. But all I want to do is generate a dialtone. This obviously can only be done in band. There is code for generating the tones when you have a physical line, like the alsa channel, or a zap channel. But I'm just thinking of if they've selected an option

Hello, Here's what I'd like to do: call my Asterisk box from a phone, hangup after a few rings, then Asterisk calls me back and presents a dialtone, than I can dial any valid number in the context the call originated. I've done it with CAPI (thanks to the script on http://www.junghanns.net/asterisk/page14.html), I'd like to do it with H323. Problem is, how to present a

I've configured an analog/single line off my mitel sx-2000 to a x100p card in my *, however when the remote caller drops the line I get a dialtone back from my pbx and the x100p doesn't detect the end of the call. Any way I can tell * to drop? I've tried call progress but it doesn't seem to work...

Systems: Definity G3rV6 Suse 9.1 with two X100P cards. I'm having trouble getting Asterisk to interface with my Definity. I can call out fine (Asterisk to Definity), but if I call either X100P (DID from Definity), strange things happen. I figure I should probably get dialtone, punch in an Asterisk station number and get it to ring. Asterisk picks up fine, but I can't figure out what to

Hi All... I'm currently trying to have an extension of my Lucent Partner phone system connected to Asterisk using an X100P. The issue I'm having is that the Lucent Partner analog port connection have different ring and dialtone than the one specified for US in indications.conf. Anyone have accomplished this connection before. I need to know what I should use for the different