Hi,
Although zonedata.c contains ITU E.180 recommendations for Turkey, we are
still experiencing unrecognized hangups from Turk Telekom PSTN lines when
callers hangup.
Turk Telekom does *not* provide supervised disconnects on analog PSTN, and
the tone we receive we when caller hangs up is
Is there any way, i can detect in asterisk that which party hanged up the
call either from A side or B.
Both parties are using SIP protocol. I am using Asterisk 1.4.27
Shariq Khan
0333-3501125
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-- Bandwidth and Colocation
On 24/04/2009 2:22 p.m., Saurabh Nirkhey wrote:
I have written an asterisk manager client which creates an outbound
call using Asterisk manager API's Originate action.
when the call is connected I run 3 applications on it.
1)read a dtmf digit from user
2)A customized application which I have
I have written an asterisk manager client which creates an outbound call
using Asterisk manager API's Originate action.
when the call is connected I run 3 applications on it.
1)read a dtmf digit from user
2)A customized application which I have written,(It plays something to user)
3)Hangup
If
Hi All,
I hate to post yet another bloody hangup detection issue on the list, but
I have been pulling my hair out no end of late with a hangup detection issue
on 1 system (have a few others out there with TDM400's and no issue but this
one is causing a real headache)
The scenario is - system
I'm having real problems getting my Sangoma A200 card with FXO board in to
detect hangup at all.
Basically if the remote end hangs up the call, Asterisk does not seem
to detect a
hangup.
A month or so ago I was running a system with 2 x X101P cards in, this
detected hangup fine.
Since switching
Is it possible to patch the zaptel drivers (or whatever appropriate files)
to use DTMF tone D for hangup detection? I have a Toshiba PBX which does
not provide CPC by any means other than congestion or D tone.
Thanks,
Aaron Picht
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A little background. I'm integrating asterisk as the voicemail service
for an old Meridian/Norstar pbx which has an ATA-2 connected. The ATA-2
is used to connect an analog device (such as a voice modem) to the pbx.
In the past we've used vgetty and a voice modem with varying degrees of
success.
Darrick Hartman wrote:
A little background. I'm integrating asterisk as the voicemail service
for an old Meridian/Norstar pbx which has an ATA-2 connected. The ATA-2
is used to connect an analog device (such as a voice modem) to the pbx.
In the past we've used vgetty and a voice modem with
Thanks for your
suggestion Steve.
I have done as you advised and set busypattern=300,200 to match the sample I recorded.This hasn't worked though, asterisk doesn't seem to detect the busy signal.Does asterisk require a the signal to be in a certain power range? The signal I getis very
On Tue, 10 Jan 2006, [EMAIL PROTECTED] wrote:
Thanks for your suggestion Steve.
I have done as you advised and set busypattern=300,200 to match the sample
I recorded.
This hasn't worked though, asterisk doesn't seem to detect the busy signal.
Does asterisk require a the signal to be in a
On Mon, 19 Dec 2005, [ISO-8859-1] Diego Andr?s Asenjo Gonz?lez wrote:
Hi everybody!
Jonathan wrote:
Hi,
I'm using a td400p card with an FXO port and asterisk 1.2.1 in South
Korea and asterisk isn't detecting when PSTN callers hangup.
I've gone through all the settings related
Hi,
I'm using a td400p
card with an FXO portand asterisk 1.2.1 in South Korea and asterisk isn't
detecting whenPSTN callers hangup.
I've gone through
all the settings related to hangup detection and none work. I've
tried:
hanguponpolarityswitch=yes
callprogress=yes
busydetect=yes
Hi everybody!
Jonathan wrote:
Hi,
I'm using a td400p card with an FXO port and asterisk 1.2.1 in South
Korea and asterisk isn't detecting when PSTN callers hangup.
I've gone through all the settings related to hangup detection and none
work. I've tried:
hanguponpolarityswitch=yes
Hi,
I have a long delay when detecting hangups on the TDM400P card, with 4
FXO ports,
When an incoming call dial's in, when hanging up, the asterisk will
detect the hangup only after 10 seconds, i searched around, and found
many similar problems, but no solution, i tried some options in
hmmm
di you try this ;hanguponpolarityswitch=yes
Cheerz!
On 11/17/05, Marco Supino [EMAIL PROTECTED] wrote:
Hi,
I have a long delay when detecting hangups on the TDM400P card, with 4
FXO ports,
When an incoming call dial's in, when hanging up, the asterisk will
detect the hangup only
Yes, didnt change anything
Marco.
Angelito Manansala wrote:
hmmm
di you try this ;hanguponpolarityswitch=yes
Cheerz!
On 11/17/05, Marco Supino [EMAIL PROTECTED] wrote:
Hi,
I have a long delay when detecting hangups on the TDM400P card, with 4
FXO ports,
When an incoming call dial's in,
Subject: Re: [Asterisk-Users] Hangup detection - TDM400P
Yes, didnt change anything
Marco.
Angelito Manansala wrote:
hmmm
di you try this ;hanguponpolarityswitch=yes
Cheerz!
On 11/17/05, Marco Supino [EMAIL PROTECTED] wrote:
Hi,
I have a long delay when detecting hangups on the TDM400P card
My telco doesn't provide Disconnect Supervision or Polarity Change. So I
figured I have to detect hangups with busydetect=yes in zapata.conf.
I tested it. When the telco sends a busy tone * detects it and hangsup.
So far so good. The problem is the telco doesn't always send a busy
after
Hi
I have a Digium TDM400 card with 4 FXO modules
connected tothe extension ports onaPanasonic KXTD816.
I'm using [EMAIL PROTECTED] v1.0, which has
Asterisk 1.07.
There's a problem that Asterisk doesn't detect when
the line is disconnected on the Panasonic. The Panasonic doesn't provide
Hilton Williams wrote:
Hi
I have a Digium TDM400 card with 4 FXO modules connected to the extension ports
on a Panasonic KXTD816. I'm using [EMAIL PROTECTED] v1.0, which has Asterisk
1.07.
There's a problem that Asterisk doesn't detect when the line is disconnected on
the Panasonic. The
PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Hangup detection with TDM400 in UK
When a caller hangs up (e.g. after leaving a voicemail), my British Telecom
exchange sends a continuous tone for about 15s and then silence. I can't get
asterisk to recognise this tone as a hangup
On Tue, Feb 08, 2005 at 04:03:50PM +0200, Doug Reid - Stormcorp wrote:
Hi
Try going into vi /etc/profile insert the lines in brackets.
USER=`id -un`
LOGNAME=$USER
Generally LOGNAME is set by login, sshd or
When a caller hangs up (e.g. after leaving a voicemail), my British Telecom
exchange sends a continuous tone for about 15s and then silence. I can't get
asterisk to recognise this tone as a hangup indication.
I have tried indications.conf with both country=uk and country=us.
My zapata.conf has
Hi ,
I need to execute a query when a user hangs up the agi application , Ive tried monitoring some return values of AGI commands
Still doesnt work .
Any ideas ?
Thanx
Ali Mughrabi
MSN 8 helps ELIMINATE E-MAIL VIRUSES. Get 2 months FREE*.
___
Ali Mughrabi wrote:
Hi ,
I need to execute a query when a user hangs up the agi application ,
Ive tried monitoring some return values of AGI commands
Still doesnt work .
Any ideas ?
Thanx
Ali Mughrabi
You will need to put another agi with you cleanup script onto the 'h'
extension.. If you
Hi,
We have a system that recorded voicemail for about an hour after the caller
hungup. I'm going to put a timeout on it but is there anything to look for
that can help prevent this? The system is running on a telenet line in
Belgium. The answer dialplan I used was:
[macro-stddial]
exten =
We have a system that recorded voicemail for about an hour after the caller
hungup. I'm going to put a timeout on it but is there anything to look for
that can help prevent this? The system is running on a telenet line in
Belgium. The answer dialplan I used was:
[macro-stddial]
exten =
Use something like the following in voicemail.conf
; How many seconds of silence before we end the recording
maxsilence=10
; Silence threshold (what we consider silence, the lower, the more sensitive)
silencethreshold=128
Rich
Ah, great. Thanks! Do you know how to find out what the
Use something like the following in voicemail.conf
; How many seconds of silence before we end the recording
maxsilence=10
; Silence threshold (what we consider silence, the lower, the more sensitive)
silencethreshold=128
Rich
Ah, great. Thanks! Do you know how to find out what
Hello all,
I just went into production on a system. 6 incomming lines 24 phones (* setting
in front of an old key system). The six lines are standard analog POTS lines
through a service called Centrex/Plexar. I am using an CAC Adit 600 channel bank
with 1 8 port fxs and 1 8 port fxo cards. I
So I made the mistake of buying a Carrier Access channel bank without
noticing the page on the wiki about the fact that they don't support
disconnect supervision (bastards!). However, apart from that, I do have
it working fine for incoming calls.
Is there some trick to get asterisk to detect
On Fri, 2004-01-02 at 12:25, Sean Adams wrote:
So I made the mistake of buying a Carrier Access channel bank without
noticing the page on the wiki about the fact that they don't support
disconnect supervision (bastards!). However, apart from that, I do have
it working fine for incoming
busydetect should help you. Set busycount=10 busydetect=yes in zapata.conf
and measure the length of the tone .. should be equal the pause too.
Then in dsp.c change the vaules BUSY_MIN and BUSY_MAX for example like
this: your result - 100, your result + 100 [ms]
regards
Martin
On Fri, 2 Jan
Okay, I'm an idiot. The tones are picked up just fine by asterisk with
no changes.
It helps if you understand the syntax of zapata.conf. I thought
busydetect=yes just had to be under the context line. I didn't realize
how the channels= is actually the delimiter that includes the stuff
above
Hello,
I've got the following configuration:
2 X101Ps
Asterisk built with BUSYDETECT_MARTIN
busydetect=yes
busycount=10
callprogress=yes
signalling = fxs_ks
With this setup, the best I can do is get voicemail with 17 to 19 seconds of
silence tacked on at the end. Ideally, I'd like at most 2-5
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