On Sun, Aug 9, 2015 at 11:46 AM, Matthew Jordan mjor...@digium.com wrote:
On Sat, Aug 8, 2015 at 8:26 AM, Administrator TOOTAI ad...@tootai.net
wrote:
Le 07/08/2015 23:54, Asterisk Development Team a écrit :
The Asterisk Development Team has announced the release of Asterisk
11.19.0.
On Fri, Aug 7, 2015 at 11:20 AM, Shahid H shah...@gmail.com wrote:
Thank you. I will test it today.
Is it possible to build a list of agent-id in MySQL Database rather than
agent.conf?
Look at extconfig.conf.sample for using a database to hold the contents of
an agents.conf file.
I am
On Fri, Aug 7, 2015 at 10:06 AM, Shahid H shah...@gmail.com wrote:
Hi,
If agents is already logged in via AgentLogin() and users dialled
extension 300 which will be placed in Queue(support-queue).
How to find out which agent is available I can put their Agent id
in AgentRequest() ?
If
On Thu, Aug 6, 2015 at 12:33 PM, Murthy Gandikota murth...@hotmail.com
wrote:
Date: Thu, 6 Aug 2015 12:07:35 -0500
From: rmudg...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk uses Anonymous, but why?
snip
On Thu, Aug 6, 2015 at 11:56 AM, Murthy Gandikota murth...@hotmail.com
wrote:
Tested with X-Lite and it worked fiine. Is there some way to replace
Anonymous with a config parameter?
Thanks for your kind help
From: murth...@hotmail.com
To:
On Thu, Aug 6, 2015 at 1:25 PM, Murthy Gandikota murth...@hotmail.com
wrote:
Date: Thu, 6 Aug 2015 12:55:28 -0500
From: rmudg...@digium.com
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asterisk uses Anonymous, but why?
On
On Wed, Jul 8, 2015 at 8:14 AM, Administrator TOOTAI ad...@tootai.net
wrote:
Hi list,
we wanted to patch our servers with 11.18.0 patch against 11.17.0 actual
running version. Patch failed with
zone-s:/usr/src/asterisk-11.18.0# patch --dry-run -p0
../asterisk-11.18.0-patch
can't find
On Fri, Jun 19, 2015 at 2:14 PM, asterisk aster...@solutionengineers.com
wrote:
Hi,
Long story short - I have an ancient Britsh Telecom phone attached to my
Asterisk PBX via Dahdi. It works beautifully, receiving calls, and the call
quality is excellent. However, dialling out is impossible,
On Fri, Jun 12, 2015 at 3:10 PM, D'Arcy J.M. Cain da...@vex.net wrote:
On Fri, 12 Jun 2015 11:49:05 -0700
John Kiniston johnkinis...@gmail.com wrote:
Try this for CHAN_SIP:
same = n,Set(Peer=${SIPCHANINFO(peername)}) ; Get the peer
same =
On Tue, Jun 2, 2015 at 8:55 AM, Joshua Colp jc...@digium.com wrote:
sean darcy wrote:
I usually lurk on the asterisk devel list to see what's going on.
No posts for a week or two. Has the list moved ?
Nope - it's just been a quiet time.
Most of the activity on the dev-list was code
On Thu, Apr 30, 2015 at 4:50 AM, Olivier oza.4...@gmail.com wrote:
Hello,
I recently gave CONNECTEDLINE function a try on an Asterisk 11 setup with
a couple of SIP phones.
When a SIP phone dials an other one, with a CONNECTEDLINE statement in its
dialplan, I noticed that Asterisk update
On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere j...@jeff.net wrote:
Hello,
I am wondering if asterisk does anything at all to RTP packets passed from
channel to channel if no transcoding is involved? Can I assume that the
packet that left phone A, arrived at the asterisk server, was
On Tue, Mar 24, 2015 at 4:59 PM, Jeff LaCoursiere j...@jeff.net wrote:
On 03/24/2015 04:28 PM, Richard Mudgett wrote:
On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere j...@jeff.net wrote:
Hello,
I am wondering if asterisk does anything at all to RTP packets passed
from channel
On Thu, Mar 12, 2015 at 5:14 PM, Leandro Dardini ldard...@gmail.com wrote:
Followme is perfect to handle FMFM and it is now also realtime, but it
seems impossible to assign some value to a variable, from within the
followme to store info for example about the tenant the followme is running
On Tue, Mar 3, 2015 at 3:35 PM, Leandro Dardini ldard...@gmail.com wrote:
I'd like to dial two extensions (or external number) and ask for
confirmation to accept the call.
Dialing an extension, asking for confirmation and then dialing a second
extension if the call has not been accepted is
On Wed, Jan 28, 2015 at 8:27 PM, Charles Wang lazy.char...@gmail.com
wrote:
Hi all,
I want to test the Native Bridge mode of DAHDI (FXS/FXO). I use asterisk
11.14.2 and DAHDI 2.8.0.
I try to set callwaiting = no AND callwaitingcallerid = no in
chan_dahdi.conf.
But I can't find native
On Fri, Dec 19, 2014 at 9:01 AM, Rui Mota ruim...@gmail.com wrote:
Hi. I am replacing a legacy Alcatel PBX for asterisk and now i'm facing a
request i can't answer yet: I need to replicate the Call Pickup function
from Alcatel, mostly when used by a secretary picking up a call from his
On Tue, Dec 9, 2014 at 1:35 PM, Patrick Beaumont
p.beaum...@hatsoffsoftware.co.uk wrote:
Hi Everyone.
I was referred here by malcolmd of the Asterisk forums. What follows is
a copy of this question:
http://forums.asterisk.org/viewtopic.php?f=1t=92007
I've recently upgraded from
On Tue, Dec 9, 2014 at 2:58 PM, Patrick Beaumont
p.beaum...@hatsoffsoftware.co.uk wrote:
Thanks Richard. This is exactly the answer I was looking for.
I'm now assuming that Asterisk 11 was using it's equivalent
bridge_simple but I was getting confused because the only bridge module I
saw
On Fri, Oct 24, 2014 at 1:19 PM, Murthy Gandikota mgandik...@nts.net
wrote:
In
https://wiki.asterisk.org/wiki/display/AST/Introduction+to+ARI+and+Chann
els
it is stated:
channel-dump.js in action
Here's sample output from channel-dump.js. When it first connects there
are no channels in
On Thu, Oct 23, 2014 at 10:07 AM, Jared Terrell jared.terr...@mcc.edu
wrote:
with the below defined in logger.conf on 11.6 cert 6
I am not getting any log message other than notice and warning in any files
when doing module reload logger - queue log is the only one that says it
restarts
In case it wasn't obvious in the DAHDI release announcement.
Richard
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
On Fri, Aug 22, 2014 at 4:55 PM, Mitch Claborn mitch...@claborn.net wrote:
Asterisk 12.5
I have a reproducible segfault using the MeetMe application. How do I
gather the necessary information (backtrace, core dump...) to submit a bug
report?
See
On Fri, Aug 22, 2014 at 5:08 PM, Richard Mudgett rmudg...@digium.com
wrote:
On Fri, Aug 22, 2014 at 4:55 PM, Mitch Claborn mitch...@claborn.net
wrote:
Asterisk 12.5
I have a reproducible segfault using the MeetMe application. How do I
gather the necessary information (backtrace, core
On Fri, Aug 22, 2014 at 5:09 PM, Richard Mudgett rmudg...@digium.com
wrote:
On Fri, Aug 22, 2014 at 5:08 PM, Richard Mudgett rmudg...@digium.com
wrote:
On Fri, Aug 22, 2014 at 4:55 PM, Mitch Claborn mitch...@claborn.net
wrote:
Asterisk 12.5
I have a reproducible segfault using
On Tue, Aug 12, 2014 at 1:33 AM, Deepak Rawat deepaksingh.ra...@gmail.com
wrote:
Hi,
I am upgrading from Asterisk 1.4 to 12.4. I am able to authenticate the
user and call AgentLogin. But after that when I call AgentRequest I keep
getting Agent '1234' is busy.
If I put a delay of 5 second or
On Tue, Aug 12, 2014 at 11:24 AM, Deepak Rawat deepaksingh.ra...@gmail.com
wrote:
Thank you for the response Richard and Matthew! It's good to hear that you
are working on fixing the 5s delay. I was really puzzled by it and found
the idle time by trial and error. Is there any documentation of
On Sat, Aug 9, 2014 at 5:08 PM, Joseph Towery tech...@bellsouth.net wrote:
Hello,
I have Asterisk version: Asterisk SVN-branch-11-r420435
I have the following code:
exten = 303,1,NoOp(Dialing ${EXTEN})
same = n,NoOp(DBKey = ${DBKey})
same = n,DB_DELETE(office/${DBKey})
On Fri, Aug 1, 2014 at 12:03 PM, Roberto Fichera ker...@tekno-soft.it
wrote:
Hi All,
I've a BT Versatility PBX that I want to connect to my asterisk 11.9.0 box
via BRI port in NT
mode but actually I wasn't able to get it working. I've another standalone
PBX, it's a
Panasonic model, which
On Fri, Aug 1, 2014 at 1:26 PM, ker...@tekno-soft.it wrote:
On Fri, 1 Aug 2014 12:39:18 -0500, Richard Mudgett wrote:
On Fri, Aug 1, 2014 at 12:03 PM, Roberto Fichera wrote:
snip
Does anyone know how can I solve this problem?
You might want to try bri_net_ptmp since bri_net is for point
On Tue, Jul 22, 2014 at 12:45 PM, Eric Wieling ewiel...@nyigc.com wrote:
Making LinkedID available in the dialplan would also be useful.
LinkedID is already available in the dialplan: CHANNEL(linkedid)
Richard
--
_
--
On Fri, Jul 11, 2014 at 11:19 AM, Ethy H. Brito ethy.br...@inexo.com.br
wrote:
Hi All
I installing a new S.O. (Ubuntu 14.04) and upgrading my asterisk 1.8. to
Asterisk 11
Asterisk 11 is not in production right now but is scheduled to be soon.
From its logs (Asterisk 11) I read a lots of:
On Wed, Jul 2, 2014 at 4:39 AM, Jonas Kellens jonas.kell...@telenet.be
wrote:
Hello,
I am trying to create a dynamic call parking lot using
https://wiki.asterisk.org/wiki/display/AST/Application_Park
But this manual is not enough to fix my problem : Asterisk keeps trying to
park the call
On Fri, Jun 27, 2014 at 1:30 PM, Tiago Geada tiago.ge...@gmail.com wrote:
Is there something I can do regarding this issue?
Have you looked at these wiki pages?
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Call+Files
The setvar parameter may help here.
On Mon, Jun 16, 2014 at 4:05 AM, babak bk1...@yahoo.com wrote:
Hi
I have done everything richard told to enable ECT .
below is my trace, anyone can help ?
-- DAHDI/i1/09123278669-4 answered DAHDI/i1/88050048-3
-- Native bridging DAHDI/i1/88050048-3 and DAHDI/i1/09123278669-4
PRI
On Wed, Jun 11, 2014 at 1:35 PM, David Huebner
david.hueb...@bolderthinking.com wrote:
I'm trying to capture when a call is placed on and removed from being on
hold through the AMI in Asterisk 12.3. In previous versions, the Hold
event contained a 'Status' field which indicated if the call
On Wed, Jun 4, 2014 at 10:14 AM, Mojtaba mes...@gmail.com wrote:
Hello Experts.
Im working with Asterisk PBXand freeswitch PBX.
I have a challenge with FXO card in Asterisk and i could not solve it yet.
I hope you could guide me in this regards.
When i want route the call to FXO channels,
On Mon, May 26, 2014 at 3:12 PM, Bart Remmerie remme...@gmail.com wrote:
Hi,
I guess something's wrong with my chan_dahdi configuration, ... but I
can't seem to get it.
When I test incoming calls on a DAHDI-channel (incoming from pstn),
asterisk seems to interpret it as a caller hangup
On Fri, May 23, 2014 at 7:05 PM, Armen K armen...@hotmail.com wrote:
Hi everyone,
I was referred to this mailing list by Richard Mudgett regarding the
following thread on Issue Tracker as it's a feature request and not a bug:
https://issues.asterisk.org/jira/browse/PRI-170
We've got
On Mon, May 12, 2014 at 12:25 PM, Nick Olsen n...@flhsi.com wrote:
Hello All, Looking for a little guidance on Real Time Pattern Matching.
We are attempting to block outbound 411 via when someone dials
NXX-555-, The must common being NXX-555-1212. However, We have some
outbound providers
On Fri, May 9, 2014 at 9:52 AM, Pawel Pastuszak pawelpastus...@gmail.comwrote:
I am trying to make a data channel using ISDN and i need to set the caller
id num field.
Can any body tell me how i can set the caller id field since i notice in
chan_dahdi.conf callerid field doesn't work with
On Tue, May 6, 2014 at 1:01 PM, Richard Kenner ken...@gnat.com wrote:
That is definitely a leak and the fix looks good.
Thanks.
That leak is most likely the one biting you.
It definitely is.
Committed the fix for this leak on Asterisk v12 branch in -r413454.
There is another leak
On Wed, May 7, 2014 at 4:43 PM, Richard Kenner ken...@gnat.com wrote:
Committed the fix for this leak on Asterisk v12 branch in -r413452.
This leak also applied to Asterisk v11.
Thanks.
Is this for both the one in the talking callback or the one in
handle_cli_confbridge_kick or both (the
On Tue, May 6, 2014 at 5:45 AM, Richard Kenner ken...@gnat.com wrote:
Really, I think we're pretty positive there's a ref leak (since
otherwise, the CBAnn channel would be long gone). If you can get a
ref debug log and the standard Asterisk DEBUG log showing the
problem, that would help a
On Wed, Apr 30, 2014 at 4:33 PM, Igor Dvorzhak idm...@gmail.com wrote:
Hi all,
I need a command to originate a new channel from dialplan. I should be
able to continue execution of the current context after this command.
How to do this?
Look at this application:
*CLI core show application
On Tue, Apr 29, 2014 at 5:10 PM, Richard Kenner ken...@gnat.com wrote:
After an upgrade to Asterisk 12, I'm collecting channels. When I enter
and then exit a conference room, I see:
-- CBAnn/207-067f;1 Playing 'confbridge-leave.slin' (language
'en')
-- Channel
On Fri, Mar 28, 2014 at 4:01 AM, Administrator TOOTAI ad...@tootai.netwrote:
Hello,
I would like to use AMD on outgoing calls using analog line. I tested with
SPA3102 and cisco2811 as gw and asterisk 1.8.26.1 as well as 11.8.1 Other
end is analog number behind another cisco/asterisk, also
On Thu, Mar 13, 2014 at 1:42 PM, jg webaccou...@jgoettgens.de wrote:
When I set CONNECTEDLINE() info for an incoming ISDN call, the calling
party sees only CONNECTEDLINE(num) and the name does not get displayed.
Some time ago I called a number, where I did get back a name and a number
and
On Wed, Feb 5, 2014 at 2:46 PM, Leandro Dardini ldard...@gmail.com wrote:
Hello,
I am migrating my dialplan from asterisk 11 to asterisk 12 and it seems
the ${CDR(start)} is not returning any data. Other functions, like
${CDR(duration)} or ${CDR(src)} or ${CDR(accountcode)} are returning
On Tue, Feb 4, 2014 at 10:56 AM, Olivier oza.4...@gmail.com wrote:
Hello,
On a Asterisk 1.6.1 powered system, I've just discovered that using
Busy() application in dialplan was no enough to send a Busy signal on
incoming Dahdi channel.
On this specific install, adding an Answer()) and a
On Thu, Jan 30, 2014 at 5:45 PM, Justin Killen
jkil...@allamericanasphalt.com wrote:
Using Dahdi/PRI, I end up with channel names like
'DAHDI/i8/9995551212-4d6B', but when I do a 'core show channels' it cuts
off those names to only 'DAHDI/i8/9995551212-'. This is the same for AMI.
Is
On Mon, Jan 20, 2014 at 5:16 PM, Dale Noll dn...@wi.rr.com wrote:
We fairly recently switched service providers for our 4 PRI circuits.
Since that time, we started to notice some failed inbound calls. These
calls terminate with an ISDN cause code 47 'resource unavailble'. Most of
the time I
On Tue, Jan 21, 2014 at 3:39 PM, Michelle Dupuis mdup...@ocg.ca wrote:
When I issue a 'core show channels' command I notice that long usernames
(and channel number) are truncated. For example, if the username is
FONEMITEL1234567890 for a trunk, then it will show
SIP
Privilege: Command
On Fri, Jan 17, 2014 at 3:42 AM, Olivier oza.4...@gmail.com wrote:
Hi,
I've installed a brand new Asterisk 12.0.0 system in which I can see, with
make menuselect, in Test Modules tab, that each test entry such as test_acl
can't be installed due a to missing TEST_FRAMEWORK(E) dependency.
On Thu, Jan 16, 2014 at 3:17 PM, Andres and...@telesip.net wrote:
On 1/16/14, 2:23 PM, Michael L. Young wrote:
- Original Message -
From: Andres and...@telesip.net
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January
On Tue, Jan 7, 2014 at 4:32 PM, fred.robin...@sipfusion.com wrote:
I'm running Asterisk 1.6.2.10, and I'm having the issue described here;
https://issues.asterisk.org/jira/browse/ASTERISK-15721
The last note says Patch heap-fix.rev2.diff was uploaded - was this
Patch released, I don't see a
On Tue, Dec 17, 2013 at 12:19 PM, jg webaccou...@jgoettgens.de wrote:
Is there a recommended way to find out the cause of DIALSTATUS =
CONGESTION for PRI/BRI channels? Currently I am evaluating the DIALSTATUS
variable and I also count the active ISDN channels for the ISDN trunk in
question.
On Mon, Nov 18, 2013 at 1:21 AM, Dmitry Melekhov d...@belkam.com wrote:
Hello!
I have following connections over isdn pri:
avaya definity---pri--asterisk--pri-panasonic 500
Just because panasonic 500 can't send user's names.
I also want to have reverse callerid for avaya users.
But if
On Wed, Oct 30, 2013 at 8:40 AM, Greg Woods g...@gregandeva.net wrote:
I've got a Digium Wildcard TDM410P with one POTS line and three
extensions. One of the extensions is connected to a fax modem. This kind
of works, but there's a gotcha. If I set faxdetect=incoming in
chan_dahdi.conf, then
On Wed, Oct 23, 2013 at 6:22 PM, troxlinux xserverli...@gmail.com wrote:
Hi, I recently changed my version of asterisk to 11.XX, and I see a waning
with h323, with version 1.8 did not have these warning
I have connected one avaya ip office 500 h323 with asterisk and the 1.8
version did not
On Tue, Oct 8, 2013 at 10:06 AM, Doug Lytle supp...@drdos.info wrote:
Last month I moved a 1.4.x Asterisk install to Asterisk 11.5.1.
Everything is working well, until I noticed that Caller ID between
facilities are showing properly, on the phone display, until the handset is
picked up, then
On Tue, Oct 8, 2013 at 1:50 PM, Doug Lytle supp...@drdos.info wrote:
You might want to look at this link for some hints about what may be
going on:
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information
I've been pouring over that document for hours and am not really
On Thu, Oct 3, 2013 at 5:15 PM, Leandro Dardini ldard...@gmail.com wrote:
When you set sendrpid=yes in sip.conf, a very nice feature is activated.
When dialing an extension, the callerid of the dialed extension is returned
back on the display of the calling phone. So if you call extension 100,
On Wed, Oct 2, 2013 at 1:41 AM, Nomad Esst noname.e...@yahoo.com wrote:
Hi list
What is the default value for signalling in
/usr/local/etc/asterisk/chan_dahdi.conf file?
You should always be explicit in setting that value.
Richard
--
On Thu, Sep 5, 2013 at 1:02 PM, jg webaccou...@jgoettgens.de wrote:
I have 2 ISDN BRI boxes, each with 4 spans, where the first one is
configured as CPE, the second one as NET(so I don't need real lines for
developing and testing).
Once in a while I do see the following libpri error messages
It seems that the ISDN switch you are connected to does not respond to the
RESTART message. You should investigate what the chan_dahdi.conf
resetinterval parameter is set to. See chan_dahdi.conf.sample for a
description.
Richard
--
On Fri, Aug 2, 2013 at 3:05 PM, Mitch Claborn mitch...@claborn.net wrote:
On 08/02/2013 01:28 PM, Matthew Jordan wrote:
On Fri, Aug 2, 2013 at 12:57 PM, Mitch Claborn mitch...@claborn.net
mailto:mitch...@claborn.net wrote:
Asterisk 11.1.0
I'm trying to use the b subroutine of the
On Fri, Jul 12, 2013 at 9:14 AM, Ishfaq Malik i...@pack-net.co.uk wrote:
Hi
I'm using asterisk 1.8 on CentOS 5
I'm initiating call recordings with MixMonitor and trying to pause them
with the features.conf.
Whenever I try to pause the recording the call dies. Is PauseMonitor
incompatible
On Mon, Jul 8, 2013 at 12:14 PM, Justin Killen
jkil...@allamericanasphalt.com wrote:
I have an installation that has analog phones connected via T1 channel
banks. I’m getting complaints from users that they will enter a partial
number (eg 91213), then turn away to get the next few digits,
On Wed, Jul 10, 2013 at 3:11 PM, Eric Wieling ewiel...@nyigc.com wrote:
From chan_dahdi.c, don't know if it applies to your situation or not.
/*! \brief Wait up to 16 seconds for first digit (FXO logic) */
static int firstdigittimeout = 16000;
/*! \brief How long to wait for following
On Fri, Jun 28, 2013 at 3:59 AM, Jens Bürger jbuer...@arcor.de wrote:
Hello everyone,
My setup:
Debian squeeze
Asterisk 1.8, DAHDI, libpri, compiled from source
TE110P, attached to a Deutsche Telekom Octopus E Modell 300/800
I'm trying to get MWI for Voicemail working. In the same server
On Thu, Jun 20, 2013 at 6:55 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
What happens when we increase the queue frame size in channels.c
if ((queued_frames + new_frames 128 || queued_voice_frames +
new_voice_frames 96)) {
Be default it is 128 and 96 if i increase it to 256
On Tue, Jun 11, 2013 at 9:29 AM, Jonas Kellens jonas.kell...@telenet.bewrote:
On 06/11/2013 04:12 PM, Matthew J. Roth wrote:
Jonas Kellens wrote:
I notice that it takes 4 to 6 seconds between someone pressing a cipher and
Asterisk continuing inside the dialplan. How come ???
...
Why
Sangoma's tech support is probably the better source of information.
DAHDI: obviously DAHDI channel
i: incoming call
The 'i' is for ISDN not incoming call since it will be this way for outgoing
calls as well.
3: span 3 (not the port)
211123456: CLID, probably subject to filtering (see
On 05/16/2013 10:07 AM, sean darcy wrote:
On 05/16/2013 09:41 AM, sean darcy wrote:
I have a call on gv over motif. I try to bridge it to another call
over
motif, but a different gv account, and I get congestion.
motif only handles one 1 channel at a time??
sean
More:
Hey, all. I've got an office set up with Asterisk, and forwarding's
got
a bit of a glitch:
When they forward, they listen for the remote phone to ring, then
hang
up. If the remote phone doesn't connect, it goes to the original
phone's VM. Is this Polycom's fault, or Asterisk's? I've
Hello to all,
I have a problem with an asterisk qsig .
I have three machines :
Nortel CS1000 --- Card Sangoma PRI --- Asterisk QSIG --- SIP Trunk
--- Asterisk
I use Snom phones on Asterisk .
If I call from Asterisk to Nortel , Nortel reminds me of the name of
the person i'm calling
In case anyone else sees this discussion in the future, the
Set(__DYNAMIC_FEATURES) line can't be over a certain length or it
stops parsing anything after that.
Thanks for the tips, Kevin.
On Thu, May 2, 2013 at 3:37 PM, Carlos Alvarez car...@televolve.com
wrote:
Good point, and that
Hi all,
I have console debugging enabled in logger.conf:
console = notice,warning,error,debug
Then a issue de command:
core set debug 100 manager.c
To see only debugging messages from AMI.
But It shows nothing!!!
And then if I do:
core set debug 1
Then I can see managar.c
- Original Message -
CLIchannel request hangup DAHDI/1-1
Would work.
But 'dahdi destroy channel 1' shouldn't segfault asterisk.
The dahdi destroy channel command is *only* for use when you know
what your doing. Even then I would not recommend ever using that
command. The CLI help
- Original Message -
hi,
strange behaviour while trying to use pri debugging on asterisk 11.x
...
please take a look:
bas1104*CLI pri show version
libpri version: 1.4.13
bas1104*CLI dahdi show version
DAHDI Version: 2.6.1 Echo Canceller: HWEC
bas1104*CLI help pri
pri
I'm reading this in my log files:
[Mar 25 12:01:23] WARNING[1593] sig_pri.c: Span 1: Got SETUP with
duplicate call ptr (0x8e3b998). Dropping call.
[Mar 25 13:21:40] WARNING[1593] sig_pri.c: Span 1: Got SETUP with
duplicate call ptr (0x8e3b998). Dropping call.
[Mar 26 10:20:54] WARNING[2643]
On 03/25/2013 05:17 PM, Olivier wrote:
Hello,
I'm giving hangup-handlers a try on a new Asterisk 11.2.1 setup.
My plan is to use this handler to update my CDRs with values such
as
Asterish and Tech cause (see function HANGUP_CAUSE).
I want to have my custom hangup-handler be run
On Thu, 21 Mar 2013, Administrator TOOTAI wrote:
I have a variable created like
... Set(__myVar=${ARG1})
... Set(__${myVar}STATUS=)
If ARG1 is abcd, variable is abcdSTATUS and should be empty. This
is OK.
Now I would like to get the value of abcdSTATUS. How to do it?
14.03.2013 17:53, Gianluca Merlo wrote:
Hello Grigoriy,
i think that you can access the information you need by using the
dialplan function CALLERID(num-plan). It should contain the lower 7
bits of the Q.931 type-of-number/numbering-plan-identification
octet.
Best regards
I need to get type of called number (TON), which is displayed in pri
debug messages:
Called Party Number (len=13) [ Ext: 1 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) 'xx' ]
Does anyone know how to do it?
According to documentation it is
I installed
DAHDI Version - 2.6.1
DAHDI Tools Version - 2.6.1
libss7-trunk
Asterisk 11.0.1
from source on Fedora 12 x86_64.
Now i`m unable to load chan_dahdi and libss7:
myserver*CLI module load chan_dahdi.so
ERROR[10124]: chan_dahdi.c:17842 process_dahdi: Unknown signalling
method
has anybody ever encountered this ERROR before? It happens frequently
on
my debian6-based pbx. I'm using Asterisk 1.8.11 with
dahdi-linux-2.4.1
and a quadBRI card.
ERROR: chan_dahdi.c: PRI Span: 3PROBLEM: General: Badly
Structured
Component
I tried to google but without success.
Is it working for anyone?
I have tried with
trustrpid=yes
sendrpid=yes/pai
but can not get it working, Asterisk cli shows prevented message like
this.
Connected line update to SIP/1231-0200 prevented
This message shows up because you are using the Dial, Queue, or
I've always used dahdi-genconf to just create the dahdi-channels.conf
and since our PRI is fairly simple (just dump all the channels into
one
group) it works with dialing with dahdi/g1/(number). I'm trying to
understand the file though for my own reference.
It seems the file looks like
What you say...Richard Mudgett (rmudg...@digium.com):
I've always used dahdi-genconf to just create the
dahdi-channels.conf
and since our PRI is fairly simple (just dump all the channels
into
one
group) it works with dialing with dahdi/g1/(number). I'm trying
- Original Message -
I am pulling my hairs out here. This is my dialplan.
exten = 100,1,Set(AGISIGHUP=no)
exten = 100,n,AGI(a2billing.php,4,callingcard)
exten = 100,n,Set(__APP_MSG_IND=${APP_MSG_IND})
exten = 100,n,Set(__APP_MESSAGE=${APP_MESSAGE})
exten = 100,n,Hangup()
exten =
I have just upgraded to asterisk 11 from 1.8
I have noticed that my Page command:
exten = 1,1,Page(SIP/101,diqA(local/intercom))
does not play the local/intercom sound to the conference.
according to the doc at
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Page
,
- Original Message -
Hello out there,
I'm running an Asterisk 1.8.15-cert1 with DAHDI.
Today I noticed that Asterisk is signalling to the calling party the
current internal CallerID whenever I put a call to another internal
phone.
Example:
Customer calls 020212345-555
- IVR
Ok so I was not able to get the actual line in use from core show
channels anymore.
So I thought I would bit the bullet and just monitor events since
that
seemed like the thing
to do. After doing that I mad a call.
Event: Newchannel
Privilege: call,all
Channel: DAHDI/i4/317XXX
This was a change in v1.8 and is documented in the v1.8 UPGRADE.txt
file:
* The PRI channels in chan_dahdi can no longer change the channel
name if a
different B channel is selected during call negotiation. To
prevent using
the channel name to infer what B channel a call is using
It is
Action: ExtensionState
Exten: 5551212
Context: fubar
This will return the status of the dialplan exten hint.
and Action: Command Command: ChanIsAvail Parameters: DAHDI/1
says Error No such command ChanIsAvail ChanIsAvail is a
dialplan application not a CLI command. It
You should just cache the AMI DAHDIChannel event information in your
program.
If you really must you could use the CLI command pri show channels.
However, it is not intended to be repeatedly run for performance
reasons. It blocks processing of ISDN messages while it is running.
I am not
On Fri, 2012-12-14 at 15:16 +, Ishfaq Malik wrote:
Hi
Can someone else please check the following:
We have installed asterisk 1.8.18.0 onto our development and test
servers. They were previously on 1.8.7.0
When an inbound call executes a queue, I can see in the logs that
the
I’ve built a custom application for our call center and am having one
problem. Unfortunately certain things happen whilst the agent has
the customer on hold which I’d like to work around. But I can’t work
out how to catch the actual hold event so I can do something about
it. From the console
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