On Wed, 16 Apr 2008, lordfuknowsyou wrote:
My thoughts now are to actually do a hangup at the end of the RxFAX and
rely on a 'h' extension to pick it up and carry on with the 2nd half
(which is PDFing and emailling the fax), but I'm concerned I'm going to
lose the channel variables as it
Heres something that's making me scratch my head... I'm using RxFAX on
ISDN lines and in-general it's going well.
However, there seems to be a case when the fax doesn't get delivered, but
looking through the CDRs it seems that the call happened, RxFAX was
executed .. time passed (1-2+
Gordon Henderson wrote:
Heres something that's making me scratch my head... I'm using RxFAX on
ISDN lines and in-general it's going well.
However, there seems to be a case when the fax doesn't get delivered, but
looking through the CDRs it seems that the call happened, RxFAX was
executed
hi all
I am using asterisk 1.4.15 I have a problem in conference .The conference
room is not getting hangup after disconnecting tha call also.It shows
disconnection on the x lite phone but when i run show channels on asterisk
cli it showr meetme room is reserved.
thanks
Rahul
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
Hi,
I'm using Asterisk 1.2.18 on a Debian Etch box.
I have two phones a Gigaset C450IP and a Snom 360. Suppose someone is
calling the Gigaset phone and a second call comes and is redirected to
the voicemail: if the new caller hangs up during voicemail announcement,
Asterisk drops the first
Hello,
Is there a way to find out which party hanged up the call. Generally
this is reported as Local disconnet/Remote disconnect in callcenter
environments.
Thanks.
Idris
Information and Communication Technologies Manager
Vodatech
___
On Tue, 12 Dec 2006 15:27:06 +0200
Idris AVCI [EMAIL PROTECTED] wrote:
Hello,
Is there a way to find out which party hanged up the call. Generally
this is reported as Local disconnet/Remote disconnect in callcenter
environments.
This is already written to the queue_log e.g.
: [asterisk-users] Hangup Party
On Tue, 12 Dec 2006 15:27:06 +0200
Idris AVCI [EMAIL PROTECTED] wrote:
Hello,
Is there a way to find out which party hanged up the call. Generally
this is reported as Local disconnet/Remote disconnect in callcenter
environments.
This is already written
Hi all!!,
I haven't the 'r' options in the dial command. I also try to turn off
busydetect and callprocess obtaining the same result..
If I turn off polarityswitch, I get hangup instead busy...
The peer isn't busy because I'm trying with my movil phone, and whit
known
Hi all..
I have a problem with my asterisk installation. I'm using a Wilcard
X100P clone in Spain. Incoming calls work fine, but when I make a
outgoing call, a hear the ringing, and the peer phone ring, when the
peer answer, asterisk hangup the call, or say busy.
This is my conf:
If your Dial() cmd has an 'r' in the options, could it be that the
ringing you're hearing is asterisk-generated, and the remote side
actually is busy? Have you tried turning busydetect=no in zapata.conf?
Moj
Eloy Gomez wrote:
Hi all..
I have a problem with my asterisk
I have an Asterisk box connected with a Panasonic KX-TEM824 and can not detect
HANGUP from this. Can anyone help me to get it work. Thanks!
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No way if you are using fxs on panasonic and fxo on *.
jorge
[EMAIL PROTECTED] wrote:
I have an Asterisk box connected with a Panasonic KX-TEM824 and can not detect
HANGUP from this. Can anyone help me to get it work. Thanks!
___
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Hi,
sipphone -- Asterisk PBX -- PSTN -- Cell Phone
sipphone sets up a call to Cell Phone. When Cell Phone hangs up,
it takes about 1 minute for Asterisk to show the Hangup Zap 1/1 message,
after which sipphone hangs up. During the time before Asterisk shows the Hangup
message, Busy Tone can be
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
Well I've found out what was causing my duplicate logging: it was
entirely a NAT issue. Found out it was only happening on some remote
endpoints (and not all of them), and that different routers proved to
not have duplicate logging.
What part of NAT could cause this? Was it really sending all
http://www.voip-info.org/wiki/index.php?page=Australia%20Asterisk%
20Details
Stumbled across this Reverse On Idle Condition (ROIC) 'feature' that
sounds very promising. Will get it enabled later today and give it a go.
On Tue, 2006-06-20 at 23:35 +1000, Carey O'Shea wrote:
Well I've found out
I'm having the exact same problem. Please any ideas? My IP phones keep ringing after PSTN hangup or PSTN answer... for about 6 or 7 seconds.On Sun, 2006-06-11 at 15:18 +1000, Carey O'Shea wrote: Does anyone have any ideas as to what can cause this large delay to stopringing?It's quite a show
Does anyone have any ideas as to what can cause this large delay to stop
ringing?
It's quite a show stopper... imagine ringing a business and being
answered by 3 different people, one after the other, all talking over
the top of each other.
On Fri, 2006-06-09 at 15:12 +1000, Carey O'Shea wrote:
I've been testing the debug version of AstTAPI, which worked for a few
calls, then a bit later in the day (and ever since), when the call is
hung up, the TAPI client doesn't get notified.
Looking at the server logs, The TAPI message that is sent upon hangup,
isn't being sent.
exten =
Thomas Kenyon wrote:
I've been testing the debug version of AstTAPI, which worked for a few
calls, then a bit later in the day (and ever since), when the call is
hung up, the TAPI client doesn't get notified.
Looking at the server logs, The TAPI message that is sent upon hangup,
isn't being
hi all (again). i have this problem. when a people call to meetme and join a conference when this people leave and hangup your phone asterisk can't detect the hangup. all people use analog lines to connect the meetme is any way to tell asterisk to hook when these people leave?
I have a TDM-400P with one FXO module. On an incoming call, I have set
Asterisk to dial my phone (exten = s,1,Dial(IAX2/carey)), which is
basically the only thing in my dialplan.
When the call is answered by the PSTN phone first, or when the ringing
call is hung up, Asterisk keeps ringing for 5+
So, your dialplan for that incoming call is just the one line?
exten =
s,1,Dial(IAX2/carey)
Nothing else? Try adding a Hangup command on the
next priority and see if that helps any.
exten = s,2,Hangup
If you
already have a Hangup command in there, then I apologize for wasting your
time. :)
Hi Undrhil,
A logical idea, but unfortunately adding it didn't change anything.
Two important points:
(1) When I test this with just IAX endpoints, no Zap, the call is hungup
immediately, (2) but the console still shows the user being called
twice.
So as a wild guess, maybe the console logging
Hi all,
I need help in disconnect supervision. Im running on AAH ver.2.5 at home
with TDM400P with 1 FXO and 1 FXS (TDM11B). I have implemented DISA on
AAH for origination (PSTN to VOIP bridging).
I'm facing problems with disconnection supervision. My calls are not
getting disconnected at times
I used quadBri Junghanns card and I config
zaptel.conf:
ZAPTEL.CONF
loadzone=it
defaultzone=it
span=1,1,3,ccs,ami
span=2,2,3,ccs,ami
span=3,0,3,ccs,ami
span=4,0,3,ccs,ami
bchan=1,2
dchan=3
bchan=4,5
dchan=6
bchan=7,8
dchan=9
bchan=10,11
dchan=12
ZAPATA.CONF
[channels]
language=it
Hello,
Load wctdm with debug=1, i.e, add this line to /etc/modprobe.conf:
options wctdm debug=1
Then watch /var/log/messages (tail -f /var/log/messages will do it),
and check when you are getting the first polarity reversal, you should
get it before the first RING. If it happens that you get
is possible to define a parameter to, hangup the line on silent? or ping
dead or something?
because all line have busy after the pc hangup :(
--
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Hi !I have some issues, i don't know exactly if it's a busy detection issue.When i dial into the Asterisk box, and if i hang up before the Asterisk answers with the IVR Welcome message, the Asterisk goes on with the call. But, if i wait for the Asterisk to answer, and if i hang up, the Asterisk
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
i updated to actual
sVN but now when i call with my phone i get a hangup when the clal should be
ringing.
with the branch all
is fine.
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Hi
I have a Te411 PRI card connected to parlay voxtream i60. Every call
that comes on asterisk over zap channel and goes on to SIP Voice Blue
gsm gateway disconects after this timeout.
This is complete sip debug log. I also described how sip communication
is done in this matter. My configuration
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
From the list I read ..
Connecting Asterisk with Digium FXO interface card with any EPABX
will have hangup issue.
Ie: hangup from EPABX extn will not be recognised by Asterisk.
I am facing this problem with TDM FXO interface cards.
Is there any work aroud for this .Like if the User
Dear Colleagues,
I Have my * with a X100P clon card. When a call in from the PSTN and
nobody answer the call go to the voicemail, then the caller my hangup
or press #. If the caller hangup the ZAP channel never hangup, but if
the caller press # the ZAP channel hangup. Even every time the outside
Hi,
I'm trying to set up a call-back system using auto-dialout files. I
want the call to be terminated when a specific timeout (defined in the
.call file) is detected. Both parties should then be hangup.
The problem is that the timeout is never detected... How to solve this?
Thank you,
Scenario is as follows.
Caller comes in over ZAP channel connects to handset on another ZAP
channel. Call is bridged.
I'd like the callee to be able to hangup on the caller and then be
presented with a agi application. Basically the agent that answered
the call has to enter a few
I have Rhio CB24 8FXS/16 FXO which connects to Digium T100P card on [EMAIL PROTECTED] 1.3. There are 2 FXOs of the channel bank connect to the Mobile Interface which the box that insert Mobiles SIM card and it acts like a normal mobile phone. I can dial via these ports but if the
Hi,
On my FC3 box I am having *v1.0.9.
The problem is that when a user calls through POTS line and leaves a
message in voicemail, the channel doesn't detect the remote hangup.
After 10 seconds of remote hangup it plays messages like vm-thankyou,
vm-review etc as if user is still online.
i have a box running debian sarge with asterisk installed from distribution
packages:
CLI show version
Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] on a x86_64
running Linux
I have managed to configure a simple dialplan (the PBX task is quite simple as
this is a small
On Thu, Sep 08, 2005 at 02:56:28PM +0200, Marek Zachara wrote:
i have a box running debian sarge with asterisk installed from distribution
packages:
CLI show version
Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k built by [EMAIL PROTECTED] on a x86_64
running Linux
I have managed to configure a
David Sampson wrote:
Hello
My single line extension
users (connected via channel banks)
need to be able to hang up faster. If they just flash the hook it
doesnt
disconnect right away. Any ideas on how to resolve this?
Thanks,
Dave
In zapata.conf put this
Hello
My single line extension users (connected via channel banks)
need to be able to hang up faster. If they just flash the hook it doesnt
disconnect right away. Any ideas on how to resolve this?
Thanks,
Dave
___
Asterisk-Users
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
I am using asterisk as a voicemail for an old tie onyx phone system. I have
almost everything working except that the stupid phone system doesn't hang
uo when the user hangs it it only sends me a bunch of 9's. I put a 9
extension that hangs up and that works for the Background app, but once I
get
Afternoon all,
After doing some test on my asterisk box I can successfully receive
calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network
Dial out from a sip phone is also not an issue, all calls connect and
terminate normally.
If I call the Asterisk PBX say from PSTN in Zap1-1
I have a similar issue.
I have 2 pstn lines and a phone plugged into my tdm400.
If I make a call to the outside using the phone, and the pstn number is
engaged, and I hang up, the line is not freed. I have been restarting
asterisk to get my external line back.
This does not happen if I make
not reappeared.
These server are also NTP servers and DHCP servers
Regards,
T
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, 23 May 2005 12:49 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Hangup Issues
I had a similar issue both with the X100P clones and TDM400.
Both were fixed by enabling AU zone and the busydetect functions. Don't
forget a full asterisk reload needs to take place after changing Zap conf
files, not just a soft-reload. Best way is to reboot the computer.
Mike
I have a
Discussion'
Subject: RE: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia
Hi,
I have 2 Asterisk servers in .pg and 2 in .au
In .pg I have had to configure them as if they were in .au and use LS
signaling.
I am using the latest Asterisk @ Home (1.0) and it is working well with
1 TDM400P
Subject: [Asterisk-Users] Hangup Issues on TDM40B FXO Australia
Afternoon all,
After doing some test on my asterisk box I can successfully receive
calls to my Asterisk PBX to a SIP phone from The Telstra PSTN network
Dial out from a sip phone is also not an issue, all calls connect
Folks;
I've put forth many many many hours into trying to identify this problem,
and finally I resolved to posting to the list for assistance
here.
Asterisk CVS-HEAD-04/02/05-14:31:34 built by [EMAIL PROTECTED] on a
i686 running Linux
(No, this is not the April Fools Windows Version ;-)
One
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
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 ringing on the
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
de 2004 9:48
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] hangup()???
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
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
Hi,
Asterisk does not hangup automatically after caller leave a voicemail message
and
hangup.!
Asterisk does not hangup automatically after the caller hangup in the Auto
attendant
menu system!
What variables should I change to have * automatically hangup if the caller
hangup?
Right now, I
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
Greetings folks;
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 = everythingelse
[dids]
I've noticed a problem with calls to Hangup when talking to my Norstars over
channelised T-1 EM trunk lines - it's been present since I started to
fiddle with Asterisk last December and it's still present in 'Asterisk
CVS-HEAD-08/13/04-10:37:13'.
Specifically, when a call is connected to Asterisk
Hi,
I realize that this topic has been hashed and rehashed a lot of times on
this list, but none of the information I was able to find quite fits my
case.
I seem to have disconnect supervision, namely the light on the handset
turns off for ~2 seconds when the other side hangs up, and I have
Hi,
The national PSTN is built up by a Siemens Eriksson digital PBX
which, in most cases, ends up in analogue interfaces with tone
dialing. It proved a hard job for me to find the most important tone
signals and information messages going out of the PSTN. However, I
dont know where and how to
I've got another issue I can't quite figure out and a search of the
archives and Google turn up nothing...
Say a call comes in (these are all via SIP) and is sent into a
Queue(myqueue,t,,,300). Note the t to allow whomever receives the
call to transfer it.
The call is enqueued, and the logged
6:36 PM
Subject: [Asterisk-Users] Hangup on transfer...
I've got another issue I can't quite figure out and a search of the
archives and Google turn up nothing...
Say a call comes in (these are all via SIP) and is sent into a
Queue(myqueue,t,,,300). Note the t to allow whomever receives
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
Hello
I setup asterisk with 4 ports FXS card for pstn
lines and one port T1 card for analog lines. I am facing hangup issue with one
of the pstn line while i think it happening when ever user tried to call
outside. I have following entries in the extensions.conf file:
[trunklocal];; Local
I am trying to integrate asterisk with a 3rd party PBX for voicemail
(Mitel). For the most part, things are working well. We only have one
main issue left: hangup detection. The connection between the two is:
* w/T100P - Zhone channel Bank FXO port - Mitel ONS (station) port
(Yes, overkill, but
If I call the Hangup command from AGI directly of via EXEC Hangup it
does not work. If shows on the console but it does not hangup. It
continues on to the rest of the priorities in the dialplan..
___
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[EMAIL PROTECTED]
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 phone
calls. Since I'm just using it for testing, I don't really care, but
it's exposed a problem: when the SIP client goes away, their calls are
left in limbo.
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
-Users] Hangup not detected on X100P
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
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
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 checked the conf
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.
Hullo!
It appears that the X100P (FXO) does somehow not passes the
'hangup' signaling *.
Sample Scenario 1:
I call in on external line X100P. I successfully ring an
extension. The extension answers. [we have an established
call going on now] I hangup (from the external call).
Listening to the
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
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