Hi Alexander
Le 06/05/2021 à 17:15, Alexander Perkins a écrit :
Hi All. We've put in a check for Do Not Call before a call goes out.
However, we have noticed that we cannot seem to pass a 'hangup reason'
for a call. For example, I'd like to know that this number is on the
DNC so our system
I was able to get on the UI of the Yealink T32G and fiddle with the
setting. Here's the setting for TLS transport in
/etc/asterisk/extensions.conf:
[transport-tls]
type = transport
protocol = tls
bind = 0.0.0.0:5061
; ca_list_file = /etc/asterisk/keys/ca.crt
; cert_file =
Thanks Joshua for the tip re using hostname rather than IP address when
configuring the phone. It worked nicely on the linphone on my macbookpro
at home. Dialplans are followed faithfully w/o the problems I experienced
earlier. I'll test using the hostname on the Yealink phone next time I'm
in
On Thu, Feb 11, 2021 at 9:01 PM Ruisheng Peng wrote:
> Sorry, my bad. I failed to change the transport to tls on the provision
> for the hardphone, nor did change the transport on the linphone setup.
> However, after I do that, the hardphone (Yealink T32G) failed to register,
> citing:
>
> [Feb
Sorry, my bad. I failed to change the transport to tls on the provision
for the hardphone, nor did change the transport on the linphone setup.
However, after I do that, the hardphone (Yealink T32G) failed to register,
citing:
[Feb 11 14:16:03] WARNING[24936]: pjproject: :SSL
On Mon, Feb 8, 2021 at 6:14 PM Ruisheng Peng wrote:
> Thanks Jashua for the suggestion. To find out if the issue was only
> limited to the softphone that was using tls transport (SOFTPHONE_B on ext
> 103, a linphone running off my MBP), I also turned one of the hard phone
> (f30A0A01 on ext
Thanks Jashua for the suggestion. To find out if the issue was only
limited to the softphone that was using tls transport (SOFTPHONE_B on ext
103, a linphone running off my MBP), I also turned one of the hard phone
(f30A0A01 on ext 100, a Yealink T32G) into using tls transport. It
behaves
On Wed, Feb 3, 2021 at 11:02 PM Ruisheng Peng wrote:
When using handsets with udp or tcp transports to dial ext 100, it'd hangup
> after the no-one-arround message. However, when using the handset with tls
> transport, it doesn't hang up on its own if ext 100 is not answered. I
> have to
Found a workaround... In case anyone else runs into something similar:
Setting congestion=yes in cdr.conf changes the writing behavior, and
instead of having one CDR with disposition=FAILED, I have all the CDRs with
disposition=CONGESTION, and as I can link them together with the
linkedid or the
It might work for you to branch on ${DIALSTRING} just after your Dial
command, if you want to handle a BUSY, NOANSWER, or other result. But if
the peer of that Dial hungup, then based on what Joshua said, it seems
there's no recovery.
--
On Wed, Feb 5, 2020 at 12:34 PM Farkas Levente wrote:
> hi,
> I hope someone can help me:-)
> we’ve got a freepbx server. there are 2 special extensions (2001, 2002).
> if someone calls this extensions (or a call is forwarded to these
> extensions) and these extension hangup (not the caller
On 6/1/19 9:18 AM, Harley Peters wrote:
I am receiving the following errors on any hangup handler subroutines.
[2019-05-31 18:22:13.958] VERBOSE[23943][C-0009] app_stack.c:
PJSIP/104090401-000a Internal Gosub(PreventForwardingLoop,s,1)) start
[2019-05-31 18:22:13.958]
On 9/12/18 1:32 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:25 PM, sean darcy wrote:
On 9/12/18 1:22 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:
I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.
SIP/mycall-x
On Wed, Sep 12, 2018, at 2:25 PM, sean darcy wrote:
> On 9/12/18 1:22 PM, Joshua Colp wrote:
> > On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:
> >> I understand that HangUp() hangs up the calling channel. I want to
> >> hangup the called channel.
> >>
> >> SIP/mycall-x calls and bridges
On 9/12/18 1:25 PM, sean darcy wrote:
On 9/12/18 1:22 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:
I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.
SIP/mycall-x calls and bridges with DAHDI/1-1.
I send SIP/
On 9/12/18 1:22 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:
I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.
SIP/mycall-x calls and bridges with DAHDI/1-1.
I send SIP/ to listen to a long, very long, file.
On 9/12/18 1:22 PM, Joshua Colp wrote:
On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:
I understand that HangUp() hangs up the calling channel. I want to
hangup the called channel.
SIP/mycall-x calls and bridges with DAHDI/1-1.
I send SIP/ to listen to a long, very long, file.
On Wed, Sep 12, 2018, at 2:19 PM, sean darcy wrote:
> I understand that HangUp() hangs up the calling channel. I want to
> hangup the called channel.
>
> SIP/mycall-x calls and bridges with DAHDI/1-1.
>
> I send SIP/ to listen to a long, very long, file.
Define "send". How are you
2015-03-05 6:11 GMT-06:00 Steve Davies davies...@gmail.com:
Looking at the pastebin, the Vega device sends a CANCEL with reason:
Reason: Q.850 ;cause=16.
Cause 16 is normal clearing and suggests that the original caller has
disconnected. I would take a look at the Vega's logs
I tried to
On Wednesday, March 4, 2015, ricky gutierrez xserverli...@gmail.com wrote:
I'm having some problems with a vega sangoma, if a call comes into my
ivr and hangs up, the call continues to ring and leaves hanging the
channel, I have to restart Asterisk and everything works Ok
my sangoma is a
Looking at the pastebin, the Vega device sends a CANCEL with reason:
Reason: Q.850 ;cause=16.
Cause 16 is normal clearing and suggests that the original caller has
disconnected. I would take a look at the Vega's logs
Regards,
Steve
On Thu, 5 Mar 2015 at 11:41 ricky gutierrez
On 04/11/14 15:11, Pat Collins wrote:
Hello group and thank you for the attention.
I'm using Asterisk 11.12 running on Ubuntu Server 12.04
We have an issue with channels remaining open after a SIP peer
unregisters.
It seems that if the peer goes away before manually hanging up a call,
the
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gareth Blades
Sent: Wednesday, November 05, 2014 4:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hangup Chanel when a peer unregisters
On 06.06.2013, at 15:05, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello,
when picking up an incoming call from one ip phone on another ip phone, the
call terminates after about 5 to 10 seconds.
When reading out the hangup cause variable in the h-extention of the
dialplan, the
On Tuesday 18 September 2012, Satria Anamarta wrote:
Hi,
I just realize in these few days there are many calls that already hangup
but not detected by Asterisk.
Those calls occupy PSTN lines and need to be manually terminated through
Flash Operation Panel or phycally disconnect the PSTN
Hi AJS,
Thank you for your reply , I am using this in IRAN so please guide me
what to do and and explain me more.
Look forward to hearing from your side.
Regards,
Mehdi
On Tue, Sep 18, 2012 at 11:28 AM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
On Tuesday 18 September 2012, Satria
Hello
In indications.com are the tones for several countries
On Sep 18, 2012 4:34 AM, Mehdi Rahimi mrm.ci...@gmail.com wrote:
Hi AJS,
Thank you for your reply , I am using this in IRAN so please guide me
what to do and and explain me more.
Look forward to hearing from your side.
Regards,
On Tuesday 18 September 2012, Mehdi Rahimi wrote:
Hi AJS,
Thank you for your reply , I am using this in IRAN so please guide me
what to do and and explain me more.
Look forward to hearing from your side.
Regards,
Mehdi
Unfortunately I am not familiar with the Iranian telephone system.
...@allamericanasphalt.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tiago Geada
Sent: Thursday, May 24, 2012 4:41 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup
Subject: Re: [asterisk-users] hangup not detected?
Okay, the next time it gets in this state I'll gather that information.
Justin Killen
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Sent
] *On Behalf Of *Justin Killen
*Sent:* Tuesday, May 22, 2012 8:53 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] hangup not detected?
** **
Okay, the next time it gets in this state I’ll gather that information.***
*
** **
Justin
On Wed, May 02, 2012 at 11:18:54AM -0500, Stephen J Alexander wrote:
Hello all,
I'm trying to solve a problem on a T1 span setup wherein calls are
apparently not hanging up properly.
CAS or PRI?
The system in question is using a Xorcom Astribank with 1 full and 1
partial T1 span, and
Tzafrir,
Thanks for your response. I'll check into those items.
Regards,
Stephen J Alexander
MPBX, LLC
http://mpbx.com
832-713-6729
On Thu, May 3, 2012 at 4:39 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Wed, May 02, 2012 at 11:18:54AM -0500, Stephen J Alexander wrote:
Hello all,
: [asterisk-users] Hangup Cause and SIP Response Code
On 04/25/2012 04:45 PM, brya...@zktech.com wrote:
Kevin
I am using 1.8.x 10.x
Then you have SIP_CAUSE available, although you'll have to enable it because it
is off by default due to performance concerns
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
track the actual SIP response code as well. How do I get access to it
durring the hangup?
It's rather hard to answer that question without at least knowing what
version of
Kevin
I am using 1.8.x 10.x
Bryant Zimmerman (ZK Tech Inc./interNetGR)
(616) 855-1030 Ext. 2003
On Apr 25, 2012, at 5:00 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 04/25/2012 07:08 AM, Bryant Zimmerman wrote:
I can log the ISDN cause code using ${HANGUPCAUSE} but I also need to
On 04/25/2012 04:45 PM, brya...@zktech.com wrote:
Kevin
I am using 1.8.x 10.x
Then you have SIP_CAUSE available, although you'll have to enable it
because it is off by default due to performance concerns.
Bryant Zimmerman (ZK Tech Inc./interNetGR)
(616) 855-1030 Ext. 2003
On Apr 25,
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming
Sent: Wednesday, April 25, 2012 6:25 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup Cause and SIP Response Code
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like
that when creating the originate command?
I don't know if it works, but it is worth a shot.
--
_
-- Bandwidth and Colocation Provided by
On Thu, 22 Apr 2010 15:58:34 -0400
Ryan Bullock rrb3...@gmail.com wrote:
Have you tried setting 'Variable: TIMEOUT(absolute)*=*60' or something like
that when creating the originate command?
I don't know if it works, but it is worth a shot.
Hi Ryan, thanks for your comment.
Unfortunately
One way to do what you want is to create an extension and then in your
originate action use a local change with that extension.
Action: Originate
Channel: Local/allow_caller_id:415111:541222:3...@context
Exten: do_echo
Context: cfmc_cdi_private
Priority: 1
Variable:
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
mancyb...@gmail.com
Sent: Thursday, April 22, 2010 3:32 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup after n seconds using originate ?
On Thu, 22 Apr 2010 15:58:34 -0400
Ryan Bullock rrb3...@gmail.com wrote:
Have
Thanks for the comments, this did the trick :)
On Thu, 22 Apr 2010 13:51:35 -0700
Jim Dickenson dicken...@cfmc.com wrote:
One way to do what you want is to create an extension and then in your
originate action use a local change with that extension.
Action: Originate
Channel:
Anahi Ludueña schrieb:
is it possible to hangup a channel from another channel?
I want to finish a call from another channel, but if I put
exten = h,n,HangUp(channelname)
it doesn't hangup... Is that correct?
You need to use the SoftHangup() application.
core show application SoftHangup
Thanks Phillipp!, it works!
Anahi Ludueña
Date: Tue, 10 Nov 2009 14:44:09 +0100
From: philipp.kemp...@amooma.de
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup, SoftHangup
Anahi Ludueña schrieb:
is it possible to hangup a channel from another channel
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
Sent: Friday, October 23, 2009 1:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup from which side
if you are debugging visually
, October 27, 2009 8:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup from which side
no, I meant this
s,1,Set(H=us)
s,n,Dial(,,g)
s,n,Set(H=them)
h,1,Noop(${H} hanged up)
That might or may not work ... since I didn't actually check it
Martin
)
- exten = h,2,hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
Sent: Friday, October 23, 2009 1:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup
B.Masoud @ SH schrieb:
When Asterisk establish a call through an outbound trunk, Is there any
way I can know who hang up the call first? The caller or the party called?
you could use the 'g' option of the Dial command together with some
logic in the hangup extensions
regards
klaus
Sent: Friday, October 23, 2009 9:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] hangup from which side
B.Masoud @ SH schrieb:
When Asterisk establish a call through an outbound trunk, Is there any
way I can know who hang up the call first
if you are debugging visually then look at SIP BYE message ... who sent it first
and on PRI who sent the DISCONNECT message first.
if you need to know that in the dialplan ... then if the originating
channel hanged up
then the dialplan should stop executing and go straight to h,1 even if
I would try hanguponpolarityswitch=yes in my dadhi.conf.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Tuesday, May 12, 2009 3:09 PM
To: Asterisk Mailing
Subject: [asterisk-users] Hangup()-command
On Tue, 12 May 2009, jonas kellens wrote:
When I call my Asterisk-server from my cell phone on one of the
PSTN-numbers that terminate in a FXO-module on my TDM410P Digium card,
and in the dialplan the end of a context is reached and Asterisk needs
to execute the Hangup()-command, I notice the
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
directory.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent: Sunday, February 15, 2009 11:26 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup extensions via CLI?
On Mon
On Tue, Feb 17, 2009 at 08:57:51AM -0600, Danny Nicholas wrote:
Ok isn't this replacing a western hack with a bridge hack? The init
0 and init 6 probably aren't going to work anyway since (1) asterisk has
to be running as root and
I have already mentioned that this is a requirement.
(2)
-users-boun...@lists.digium.com] On Behalf Of Lenz Emilitri
Sent: Friday, February 13, 2009 3:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hangup extensions via CLI?
This version will hang up the given extension even if it has multiple channels
On Mon, Feb 16, 2009 at 12:15:08AM -0500, Alexander Lopez wrote:
This will hang-up all channels even if multiples channels are open...
Exten = _86,1,system(“init 0”)
Use with Caution…☺
Only if Asterisk is running as root. Which is not recommended, anyway.
And besides, I think you
On Wed, 11 Feb 2009 12:34:12 +0100, Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201
| awk '{ print $1 '} )
what if there're also channels
On Fri, Feb 13, 2009 at 06:08:45PM +0800, Dinesh Nair wrote:
On Wed, 11 Feb 2009 12:34:12 +0100, Lenz Emilitri wrote:
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' |
, will repost:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Helius
Ferreira
Sent: Thursday, February 12, 2009 4:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup extensions via CLI
You guys think YOU'RE overdoing it... your solution works with a single line.
My solution was some convoluted 100 line shell script!
Tim Nelson
Systems/Network Support
Rockbochs Inc.
(218)727-4332 x105
- Lenz Emilitri wrote:
I have a feeling we're overdoing it :)
l.
Asterisk 1.6 implements the hangup on the channel you just made the call
and I used it with this command (apparently)
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/7000|
awk '{ print $1 '} )
In my asterisk system:
debian*CLI core show channels
Channel
:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Helius
Ferreira
Sent: Thursday, February 12, 2009 4:42 AM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Hangup extensions via CLI?
Asterisk 1.6
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep SIP/7000
Hi,
I used this way of processing output from asterisk 1.2 and found out
that it is not 100% safe because there can appear unprintable characters
in the output. This will cause the following grep command to show
I have a feeling we're overdoing it :)
l.
2009/2/12 Lukas Rypl r...@marconi.ttc.cz
asterisk -rx soft hangup $(asterisk -rx 'core show channels' | grep
SIP/7000
Hi,
I used this way of processing output from asterisk 1.2 and found out
that it is not 100% safe because there can appear
This is a bit of trickery, but could not resist :)
This will kill a channel that is connected to SIP/201
asterisk -rx soft hangup $(asterisk -rx 'show channels' | grep SIP/201 |
awk '{ print $1 '} )
It basically calls *, gets the list of channels, filters them out to get the
channel name and
Have you looked at soft hangup
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Monday, February 09, 2009 3:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
I've tried using a SIP client and when asterisk issue the Hangup
function the SIP client indicate that the call is terminated.
Maybe a SIP parameter with the pstn gateway ?
Cyril SCETBON wrote:
Hi guys,
My asterisk server is connected to a pstn gateway using SIP. When I
receive a call and
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
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
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
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
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!
___
--Bandwidth and
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:
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
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
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
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 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
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
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
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 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
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 for interfacing with the PSTN lines.
Previously I had exactly the problem you
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
Thanks
Terry noticed [EMAIL PROTECTED] 0.7 will try version 1
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Terry H.
Gilsenan
Sent: Monday, 23 May 2005 1:35 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk
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