similar to: analog phone connects to zaptel fxoks is beeping

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

> -----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

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*

Does anybody know if there are any external telephone-keypads for sale anywhere? (containing the keys 0-9, *, # and onhook/offhook would do) I am looking for a keypad to control a softphone and would prefer the controls to be in the physical world instead of as a window. Sincerely, Markus Hakansson

Hello list, I have reason to believe that ADSI can be spoken to phones even when they're onhook. Is this true? Does anyone know? Right now, I'm having trouble figuring out how to do anything to an onhook channel other than ring it. Does this require a magic application and some serious voodoo? Any pointers? I got onto this idea because I noticed that whenever I got voice mail on a

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 --

Hi, I want to do a scripted 'restart when convenient' on a daily basis. This used to work, but since i've upgraded to Asterisk 11.7 it seems it's never convenient to restart the server. My question: how can i tell *why* it's not convenient to restart the server? It used to be some colleague left the receiver OffHook or something like that, but even when i'm fairly

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

I've installed Grandstream GPX-2000 phones and have successfully enabled one of my buttons to use Asterisk BLF for an extension. I can tell when this extension is available, is being rung, or is on the line. I'd like to do the same for my Zaptel channels, to be able to see when a line is onhook, ringing or offhook. I tried the following but alas, it doesn't seem to be working:

I've been having trouble with making outbound calls to my TELCO from a TDM400 card (FXS KS signalling) after upgrading from 1.6-beta9 to 1.6.0. The problem is completely intermittent. When it fails, I get this message: [Nov 9 19:12:26] WARNING[18916] app_dial.c: Unable to create channel of type 'DAHDI' (cause 0 - Unknown) At some point, it starts working, but I don't know what

OK... so I'm not sure what I'm looking at. I've had the good old echo problems on my Rev C FXO again this morning, so I thought I'd attempt some debugging, though I'm not sure what I'm looking at. This call has echo. Channel: 2 File Descriptor: 20 Span: 1I> Extension: Dialing: no Context: incoming Caller ID string: "External Call" <99999999> Destroy:

http://chan-sccp.berlios.de/ 20050713 ftp://ftp.berlios.de/pub/chan-sccp/chan_sccp-20050713.tar.bz2 I didn't have a spare 7960 to use this week, so maybe some line issue is still present. - fixed a memory leak on database updates (dnd, cfwd*) - fixed old memory leak on unload (now unload chan_sccp.so and load chan_sccp.so work. It does reload the config when asterisk is running) - socket

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

Thanks Zeeshan, SAngoma told me that the asterisk problem is unrelated to wanpipe drivers, they told me to reinstall asterisk again But, i still having doubts about the problem :( Thanks in advance > > Message: 10 > Date: Thu, 18 Mar 2010 00:21:06 -0400 > From: Zeeshan Zakaria <zishanov at gmail.com> > Subject: Re: [asterisk-users] Asterisk DIES with no trace. PLEASE >

Hello, In my quest to figure out the source of the random echo on our shiny new asterisk install, I have been using ztmonitor on the TDM400p channels for the good part of today. I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to them (last 2 channels are unused but configured in zaptel). Even when the lines are onhook, the Tx values settle down to 0 but the Rx values still

I'm receiving the following error over and over, adnauseam: Oct 1 23:59:53 NOTICE[3194]: chan_sip.c:5890 check_auth: stale nonce received from 'CNAME-CID <sip:5551212@192.168.1.X>' Does anyone know what "stale nonce" is? Thanks! Paul Conn -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, On Sun Jun 12 09:11:13 CDT 2005, Rich Adamson wrote: > > > exten => s,1,Wait(1) > > exten => s,2,GoTo(s,1) > > > > If I'm on the console when a call comes in, it loops through this bit of > > code a bunch of times. I'm guessing I could lengthen the "Wait(1)" time, > > but is there any other way to do this? > > Sure there is,

Hi, the new Grandstream release for the ATAs allows the setting of the FXS impedence, the Onhook Voltage and the Polarity Reversal. Anyone know how these should be set in Germany? -- Best regards Peer Oliver Schmidt

We are having an issue with the SPA962/932 combo where the phone and the sidecar will reboot unexpectedly ? could be onhook, could be on a call, doesn?t seem to matter. I read that certain early firmware revisions could cause this so I?m running what was a week ago the newest available (5.1.18). A call to Linksys support suggested that I ensure that the phones are using a recent firmware version

>Asterisk and it works fine untill the following >situation: > >- one of the telco lines occasionally becomes mute after call is completed, would not provide dial tone, (not sure about ringing on that >line) - both via old and new PBX. >- zap show channel <n> would show that line as 'Offhook', though no telephone is off hook. > >If physical line would be