similar to: TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment

I have a TDM400p with 3 FXS mods and 1 FXO mod. I have all set up with what seems to be correct settings (according to digium and asterisk wiki). As soon as I plug in my POTS line into FXO mod the line goes into offhook state (whether I have power to the card or not). Should this happen? When I try to call * box all I get is busy signal. I've installed stable version, cvs version, change

I have an Asterisk@Home 1.3 server (Asterisk 1.0.9) and recently added a TDM400P with (1) FXO card on port 4. Inbound calls are always successful but outbound calls fail 75% of the time with intercept messages from my dial tone provider that include "we're sorry, your call did not go through", and "we're sorry, when placing a local call it is now necessary to dial an area

I installed Trixbox and a TDM400P with 2 FXO and 2 FXS ports and am having an annoying issue with the FXO ports. As soon as I plug either one into the phone line it's as though the line is disconnected i.e. get disconnected tone when trying to dial out, line is busy when dialling in. The CLI shows the following: trixbox1*CLI> zap show channel 4 Channel: 4 File Descriptor: 18 Span: 11*

My TDM400P w/ 4 FXO cards seems to have trouble with onhook/offhook switching. It dials perfectly, but does not seem to be changing the onhook/offhook state appropriately. It changes sometimes, but it's not really reliable. For example: When I booted the machine, it started as onhook. It remained "onhook" through the entire first call (which was silent on both ends --

> -----Original Message----- > From: Jacqueline Lee [mailto:jlee@isdomaininc.com] > Sent: Friday, September 23, 2005 11:46 AM > To: asterisk-users@lists.digium.com > Subject: channel offhook state > > > We are using a digium card (TDM400) with asterisk for our access to the > PSTN. Initially when the server starts, all the zap channels on the card > are in the

Hello, Caller -> PRI -> SIP -> application. If application goes offhook right away, dtmf/audio works fine in both directions. If application, before going offhook (sending OK) plays a message and wants dtmf/voice from the caller, then caller can hear this message but his dtmf/voice don't reach application. Any way to configure it? Thank you. Alex Zarubin -------------- next

Hi, We are having a problem with asterisk detecting that an analog ext has been put down. This seems only to happen after a number of calls have been made. We have an FXO port (TDM400P with FXO module) connected to our PBX and are using this to test asterisk prior to rolling our for our small office. What happens is that we make a number of calls to this ext which 1st rings a phone (FXS)

Hi, We have two of these 4 port FXO cards. However, we are having some problems with incoming/outgoing calls. The latest version of Asterisk/zaptel from CVS is being used. Voicemail, internal SIP <-> SIP calls between Pingtel xpressa hard phones work terrific, echotest is fine, and so on. The zaptel and wcfxs modules load fine, and show all 8 FXO interfaces in dmesg:

Happens quite often. X100P FXO card puts the PSTN line offhook, so that no calls go out or come in. The outside callers get a busy siganl while inside callers cant dial PSTN. Its a DELL optiplex P3 128MB ram 500MHz processor. Here is some more info: (see the zapata.conf in the end) Please direct me where to look for problem. Thanks!!! ======================================== pbx1*CLI> zap

Hello I need to write a script that will dial a list of customers and play a message. I couldn't find a way to tell Asterisk/Zaptel to wait until the callee has actually picked up the phone before proceeding with Playback(): ============ ;call made through Dial(): Doesn't proceed after off-hook/hangup [internal] exten => 8888,1,Dial(Zap/1/${IPPI}) exten => 8888,n,NoOp(We never

I am trying to get Asterisk 1.6.2.5 working with a Grandstream HT-503 ATA. The FXO part is giving me fits. Every call I try to make to the FXO port outbound I get 403 Forbidden coming back. I've been through every configuration setting I can see, and Uncle Google is not helping me much. I updated the firmware to the current version, and that didn't help. If anyone has this working, I

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

Does anyone know if it is possible to use the following disconnect tone setting with an x100p card? Disconnect Tone: 350@-19,440@-19;4(.25/.25/1+2) This tone was written for a Sipura SPA-3000 for a Panasonic KX-TD1232. The Panasonic does not support disconnect supervision, so this tone is the only thing that will detect a disconnect. It is not a standard fast busy or offhook tone.

Evening all. I'm looking at building an Asterisk system for one of the projects of the Charity where I'm the SysAdmin. The project has two analogue phone lines - BT Featureline Compact, we're in the UK - that I'd like Asterisk to handle. My current quandary is which FXO interface to use. I've been looking at the Digium TDM400P and Sangoma A200, which are similar prices for

Hi all, I would like to make a script/program that would be able to show lots of status information from my analog FXO lines (and FXS lines in the near future). Example of interesting status information: - Hook status: is there a call being made with that zap? - Voltage status: cable connected, voltage values (if possible), line ringing? - RX/TX Volume status I'm using a TDM400 card with

Outoing calls I make using Motif Google Voice Calls continue ringing even after the other end picks up. I have to restart Asterisk to resolve the issue. I don't see any errors. It's not recognizing that the other party picked up the phone and restarting Asterisk fixes it only for a day. -- Co-op Vacation Rentals www.coopvr.com 15218 Summit Ave Suite #300-354 Fontana, CA 92336

Hi, I have to develop a phone application using asterisk's chan_oss. When the phone is idle, i.e. the last command was a hangup, one hears a "toot, toot, toot, ..." But unforuntaly its use is in Germany, where one expects a continous "toooooooooooooooooooooooooooooooooo ..." before dialing. Is there anything to define the tone indicating "ready to dial"?

Hi, I'm having an interesting problem with my card. It seems to work fine, for the most part. When I first load the module and asterisk, it detects the line in the on-hook state. However, after the first phone call, zap show channel 1 lists it as being off-hook. During subsequent calls, the card is listed as being on-hook, and when it's not used -- off-hook. There are also some weird

Greetings All! The Asterisk Call Manager works great. But I have one question for anyone who has used it. I cannot get the system to send some DTMF tones down the channel once the call has been made. Below is the script I am using to make the call, and start recording the channel. I am starting to make a system the will use asterisk to become an automatic random quality monitoring system

All, Experiencing some issues on my FXO lines. If a call comes in on an FXO and then get transferred to another FXO (say to call someones cell phone), those two lines will stay tied together indefinitely. This happens to us when we transfer an incoming call to our on call guys after hours and on weekends. We have installed 3 other * boxes and they do the same thing. We use a Adit Channel bank