Hi
Try going into vi /etc/profile insert the lines in brackets.
USER=`id -un`
LOGNAME=$USER
MAIL=/var/spool/mail/$USER
MONITOR_EXEC=/usr/bin/soxmix
VPB_TONE=BUSY,P,400,100,500(insert the following line)
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
I as a similar problem with this:
ignorepat = 9
exten = 9,1,Dial,Zap/g2
exten = 9,2,Congestion
What if I pressed 9, called a number, and hanged up before someone replies..
It happened with me more than once that the line is left open, waiting for
the other side to hangup (what if there is no
Hi,
this call is from? Zap channel, Capi channel or other channel? It is
possible that you don't detect well hangup from incoming channel.
Regards.
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Altus Snyman
Enviado el: lunes, 22 de noviembre
Good day all
I want to tell asterisk that it should hangup a channel in a certain step
For example: exten = s,5,Dial(SIP/302,25) exten = s,6,Hangup exten =
s,7,Hangup(SIP/302)
What happens is that if someone calls into the pbx and hangs up before
it gets answered it still keeps on
Altus Snyman wrote:
Good day all
I want to tell asterisk that it should hangup a channel in a certain step
For example:
exten = s,5,Dial(SIP/302,25)
exten = s,6,Hangup
exten = s,7,Hangup(SIP/302)
What happens is that if someone calls into the pbx and hangs up before
it gets answered it still
On Wed, 8 Sep 2004, JP Hindin wrote:
I have a bit of a conundrum, and I can't tell if Asterisk is doing
something daft, or whether I'm clean missing out why it's doing what it's
doing. So, I have a dialplan that looks a little like this:
[start]
include = dids
include
You just described an supervised transfer. You need to be pressing # then
dialing bob and hanging up.
I suspect your sip devices don't support supervised transfers properly.
bkw
- Original Message -
From: Chad Scott [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 01, 2004
Thanks for the pointer... I don't think the device is the issue because
these types of transfers work in any other case.
For instance, say I call Bob from my SIP phone. Bob answers and decides
he's going to transfer me to Mike. He hits 'transfer,' dials Mike,
talks for a bit, then hits
I think I've narrowed this down via experimentation...
This seems to happen *only* when a call is sent to a phone out of the
queue and you attempt to put it back in the queue.
All the other use cases I've said seem to work during testing (from the
queue to party A, transferred to party B).
So,
I do those types of transfers all day long without issues on a 7960
bkw
- Original Message -
From: Chad Scott [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 01, 2004 7:29 PM
Subject: Re: [Asterisk-Users] Hangup on transfer...
Thanks for the pointer... I don't think
This is a known issue with SIP - look at bug 207 in the bug tracker
Andy
*** REPLY SEPARATOR ***
On 08/04/2004 at 12:37 Scott Laird wrote:
I've noticed a little problem with my setup. I've been using a flaky
version of X-Lite for testing, and it tends to crash every few
What version of *? I'm using 0.7.1 and it still has occasional problems
detecting call hangup.
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, March 31, 2004 8:54 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk
I've using CVS-03/30/04-14:38:02
Not sure where else to get the version number.
-Original Message-
From: John Vogel [mailto:[EMAIL PROTECTED]
Sent: 01 April 2004 16:45
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Hangup not detected on X100P
What version of *? I'm using
Matt Bridges wrote:
I've configured my [*] to dial the pstn which is working like a charm.
I've also configured an extension to ring when the PSTN line is ringing
which is also working brilliantly, but, sometimes it doesn't detect that the
call has been hungup.
I've had a look on voip-info and
On Wed, 2004-03-31 at 10:14, Matt Bridges wrote:
I've configured my [*] to dial the pstn which is working like a charm.
I've also configured an extension to ring when the PSTN line is ringing
which is also working brilliantly, but, sometimes it doesn't detect that the
call has been hungup.
What sort of phone line are you using? Connecting an X100P to a PBX line
or ISDN TA can cause the problems you mention.
Iain
--On Wednesday, March 17, 2004 7:37 am -0600 [EMAIL PROTECTED] wrote:
Hullo!
It appears that the X100P (FXO) does somehow not passes the
'hangup' signaling *.
Sample
Ahaa!
I am using a line coming out of an ISDN breakout box ..
I'll try it with a regular analog line next.
I'll let you all know what happens.
Thanks for the hint,
Willy
- Original Message Follows -
What sort of phone line are you using? Connecting an
X100P to a PBX line or ISDN TA
[EMAIL PROTECTED] wrote:
It appears that the X100P (FXO) does somehow not passes the
'hangup' signaling *.
snipped your scenarios
I am having the same issue on a normal analog POTS line (but in France
so you never know what other signalling anomalies there may be.)
The h signal never happens on
NOOOP!!
Unfortunately, a simple POTS line (AllTel Communications)
does not resolve the issue. It appears the problem is
somehow related to the digium card, or the drivers or what
not.
Anyone from digium monitoring this list? Is this a bug
thing?
FYI here's my zapata.conf
It's probably because you're using loop start lines, which don't offer proper
hangup detection. Switch to ground start and the X100P will work for you.
One way around the voicemail issue is to turn up the silence detection and set
a short timeout.
Search the list archives for lots more
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!
What is the relationship between when CDR recording occurs and the
hangup extension is executed. Normally CDR happens before the h
extension is executed.
In short: Do not rely on h for CDR purposes.
I use the h extension to clean up for routines, but sometimes it gets
called to
On Fri, 30 Jan 2004 19:22:21 -0500
Andres [EMAIL PROTECTED] wrote:
Eduardo Goncalves wrote:
Hi list,
I'm with a little problem on my E1 (EM signaling) link. Every
call a
make hangs up after 2 or 3 seconds of conversation. I got the fowling
messages from cli:
:
Zap/1-1
Eduardo Goncalves wrote:
Hi list,
I'm with a little problem on my E1 (EM signaling) link. Every call a
make hangs up after 2 or 3 seconds of conversation. I got the fowling
messages from cli:
:
Zap/1-1 answered SIP/atapd-238e
Urgent handler
Urgent handler
-- Hungup 'Zap/1-1'
Urgent handler
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
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
John Todd wrote:
At 6:32 PM -0600 12/20/03, Brian West wrote:
On a side note.. you can't use exten = h, if you have any hope of
getting
accurate billing info. Its wise to call ResetCDR(w) in your exten = h,
or not use it at all.
Care to expound a bit on that topic for the wiki, with some
What about playing a warning beep or IVR one minute before the call
hangs up, can this be done?
Thanks,
Lei
On Sat, Aug 02, 2003 at 09:39:38PM -0500, Martin Pycko wrote:
Typically you use AbsoluteTimeout app.
Martin
On Sat, 2 Aug 2003 [EMAIL PROTECTED] wrote:
hi everybody,
can
Typically you use AbsoluteTimeout app.
Martin
On Sat, 2 Aug 2003 [EMAIL PROTECTED] wrote:
hi everybody,
can anybody pls tell me a way to hangup an answered Zap/SIP/IAX call, after a
specified time period
expires, like after 10, 15 minutes.
Surajee
--This mail sent
in a year or so
anyway, so its no big loss.
Thanks again for your help!
Jim
- Original Message -
From: Martin Pycko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 24, 2003 2:42 PM
Subject: Re: [Asterisk-Users] Hangup problems...
It should detect if the party that called
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