Displaying 20 results from an estimated 1000 matches similar to: "CISCO CP-7902G and chan_skinny."
2004 Dec 06
0
TDM OnHook/OffHook
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 --
2005 Sep 23
1
FW: channel offhook state
> -----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
2004 Dec 07
2
TDM400P FXO channel remains Offhook after outoing or incoming call / line is parallel with other telephone equipment
>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
2005 May 16
2
Telephony keypad
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
2008 Nov 11
1
What makes TDM400 FXS Connection to TELCO go into Off Hook State?
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
2005 Oct 02
1
analog phone connects to zaptel fxoks is beeping
Hi,
I have a analog phone connect to a WCTDM card.
It used to work fine. Now recently, after several conf change and power
restart,
it stops working.
Whenever I pickup the phone, instead of hearing the dial tone, I hear a
busing beeping
tone, like a machine gun is firing. :) However, from asterisk console, I do
see a a
OffHook/OnHook message, but whatever I dial in the phone keypad seems not
2014 May 28
1
'restart when convenient'
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
2004 Dec 29
1
Can I tell if it hung up due to busydetect or disconnect supervision?
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
2007 Jan 26
1
Analog FXO status checking
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
2005 Aug 12
3
TDM400P FXO channel hookstate always "Offhook" & outbound digits sent before provider dialtone
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
2006 Nov 15
1
Monitor Zap Status - Full E-mail...
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:
2005 Jul 13
5
chan_sccp new release
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
2004 Jul 11
1
Echo issues (again...)
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:
2005 Jun 12
1
Not answering inbound a line used for outboun
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,
2008 Oct 20
1
Zaptel FXO offhook when connected to PSTN
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*
2007 Apr 26
1
Cisco 7920 sccp
I am trying to register cisco 7920 to asterisk using sccp since to sip
firmware upgrade to it ,but its ends with failed registration.Can you
please send me a sample for sccp.conf configuring cisco 7902.
Thanks
-- SCCP: Accepted connection from 192.168.5.163
-- SCCP: Using ip 192.168.5.228
-- SCCP: Accepted connection from 192.168.5.163
-- SCCP: Using ip 192.168.5.228
2003 Sep 24
0
Adding a DELAY to an ADSI script
I was searching through the app_adsi.c file and found
some events and functions that are not used in the
sample ADSI scripts.
One of these functions is DELAY. I can't get this to
work. Has anybody got this to work?
I'm trying to create a HangUp soft key using the
following code:
KEY "Hangup" IS "Hang Up"
ONHOOK
DELAY
OFFHOOK
ENDKEY
The delay has no
2006 Nov 15
0
Monitor Zap Status
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:
2004 Jun 20
1
Data over Voice through Asterisk
Hi,
I'm trying to make a dialup internet connection through my asterisk
PBX. When I bipass the Asterisk box, I can get 51600bps. When I run
through the asterisk box, I'm limited to about 21600bps.
I have a TDM31B card.
Any help on speeding these connections up would be good - I was on the
understanding that if you bridged the channels, then the call should
essentially flow straight
2010 Jun 17
1
applicationmap and ChannelRedirect
Hi,
I'm struggling with a feature in my home phone setup. I have several
phones using both SIP and SCCP. What I try to do is to create a dynamic
feature that works similar to the blindxfer feature built into Asterisk.
What I want is the possibility for the called part to push a number
sequence (for example *#) to redirect the callee to a fixed extension or
(for example *123#) to redirect the