If the function ChanIsAvail does not work to check if a SIP peer is
registered or not, what function should I use then ??
Jonas.
On 04-10-12 17:05, Jonas Kellens wrote:
On 04-10-12 16:59, Danny Nicholas wrote:
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun
On 05-10-12 14:45, Joshua Colp wrote:
Jonas Kellens wrote:
Hello,
I notice that the function ChanIsAvail always returns result : 0
It does not matter if the realtime SIP peer is registered or not.
How come ??
My dialplan :
exten = s,n,ChanIsAvail(SIP/${SIPPEERNAME})
exten = s,n,NoOp
: Re: [asterisk-users] AVAILSTATUS always 0
Jonas Kellens wrote:
Hello,
I do not want to know if the remote side may or may not decline the
call, I just want to know if the SIP peer is registered or not. That
is information that Asterisk has without placing a call. Placing a
call to an unregistered
On 05-10-12 15:27, Joshua Colp wrote:
Jonas Kellens wrote:
Using this will make Asterisk hang. Done that in the past and result was
that Asterisk hung after a certain amount of asterisk -rx command. So
my experience is that this is not the correct solution.
If only ChanIsAvail could return
Hello,
I notice that the function ChanIsAvail always returns result : 0
It does not matter if the realtime SIP peer is registered or not.
How come ??
My dialplan :
exten = s,n,ChanIsAvail(SIP/${SIPPEERNAME})
exten = s,n,NoOp(availstatus = ${AVAILSTATUS})
${SIPPEERNAME} = sip username from
On 04-10-12 16:59, Danny Nicholas wrote:
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, October 04, 2012 9:48 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk
On 04-10-12 19:50, Chad Wallace wrote:
On Fri, 28 Sep 2012 11:03:05 +0200
Jonas Kellens jonas.kell...@telenet.be wrote:
On 28-09-12 10:57, Administrator TOOTAI wrote:
Le 28/09/2012 10:40, Jonas Kellens a écrit :
Maybe I need to explain a bit further : the call is send to the
IP-phone
Hello,
are there any known reasons why Asterisk would disconnect random calls ?
My server uses 1,5 GB out of 8 GB RAM
My server uses up to 35% CPU at peak
There are about 40 concurrent calls.
I have 300 RTP-ports available.
I just see the call ending, as if one of the connected parties hung
On 28-09-12 10:31, Administrator TOOTAI wrote:
Le 28/09/2012 10:22, Jonas Kellens a écrit :
Hello,
Hi
are there any known reasons why Asterisk would disconnect random calls ?
My server uses 1,5 GB out of 8 GB RAM
My server uses up to 35% CPU at peak
There are about 40 concurrent calls.
I
On 28-09-12 10:57, Administrator TOOTAI wrote:
Le 28/09/2012 10:40, Jonas Kellens a écrit :
[...]
are there any known reasons why Asterisk would disconnect random
calls ?
My server uses 1,5 GB out of 8 GB RAM
My server uses up to 35% CPU at peak
There are about 40 concurrent calls.
I have
Hello,
this might seem a stupid question but I really don't see the solution to
the problem.
Using Asterisk 1.8.12.2
In extconfig.conf I have :
voicemail = mysql,AsteriskHosted,voicemail_users
sipusers = mysql,AsteriskHosted,sip_buddies
sippeers = mysql,AsteriskHosted,sip_buddies
queues =
On 27-09-12 11:27, Thorsten Göllner wrote:
Maybe a stupid answer ;-)
Did you make a reload?
Yes, I reloaded and restarted several times.
Did you try from shell:
mysql -u myuser -pmysecret AsteriskHosted
Yes, works perfect to connect via commandline.
Only Asterisk does not see the
On 27-09-12 11:54, Ishfaq Malik wrote:
On Thu, 2012-09-27 at 11:00 +0200, Jonas Kellens wrote:
Hello,
this might seem a stupid question but I really don't see the solution
to the problem.
Using Asterisk 1.8.12.2
In extconfig.conf I have :
voicemail = mysql,AsteriskHosted,voicemail_users
On 27-09-12 11:54, Ishfaq Malik wrote:
On Thu, 2012-09-27 at 11:00 +0200, Jonas Kellens wrote:
Hello,
this might seem a stupid question but I really don't see the solution
to the problem.
Using Asterisk 1.8.12.2
In extconfig.conf I have :
voicemail = mysql,AsteriskHosted,voicemail_users
Hello,
using asterisk 1.8.11.1
using realtime queues
When trying to remove a queue member, I get the following :
-- Executing [122@from-TESTCORP:2]
RemoveQueueMember(SIP/testcorp5-000c, testcorpq1,SIP/testcorp7)
in new stack
WARNING[18788]: app_queue.c:5653 rqm_exec: Unable to remove
On 19-08-12 21:58, Alec Davis wrote:
So I'm just looking on how to make a BLF-button blink or turn
red, to show to my customer that there are still calls inside
the queue waiting.
Can I only apply on Asterisk 1.8.5 ? Or can I apply to my Asterisk
1.8.11 also ?
It's 4 lines, plus 2 debug
On 08/19/2012 06:38 AM, Alec Davis wrote:
Do you also know why it hasn't been accepted ? Seems like this
functionality is asked for on different forums. Wanting
to watch a
queue for calls is not that strange.
Not sure why?
Maybe I didn't promote it enough.
Maybe my examples aren't simple
On 18-08-12 03:03, Alec Davis wrote:
I'm trying to monitor a Call Queue with BLF-button to see if there
are calls inside the call queue.
Currently asterisk doesn't support hint's on queues, unless done externally.
See my review 'Support a hint on a queue'
On 18-08-12 12:42, Alec Davis wrote:
-Original Message-
From: Alec Davis [mailto:siva...@paradise.net.nz]
Sent: Saturday, 18 August 2012 10:36 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] BLF and Call Queues
I've seen this post.
Hello,
I'm trying to monitor a Call Queue with BLF-button to see if there are
calls inside the call queue.
This I have :
extensions.conf
exten = 566,hint,Queue:voipq1
On the CLI I see :
566@908001-blf : Queue:voipq1State:Unavailable Watchers 1
But when a call enters my queue
Hello,
is there a way to order call queue members in the database table?
When defining the table for realtime queue_members, I notice there is no
ID-column.
Can I add an ID-column, or will this fail realtime Queues ?
Kind regards,
Jonas.
--
Hello,
when a call comes in and is answered by colleague A, this colleague A
sees the CallerID of the external calling number.
When colleague A transfers the call to colleague B, attended or
unattended, then colleague B sees the number of colleague A on his
screen while talking to the
On 05/08/2012 04:24 PM, Kevin P. Fleming wrote:
On 05/08/2012 08:50 AM, Jonathan Rose wrote:
- Original Message -
From: Jonas Kellensjonas.kell...@telenet.be
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Sent: Tuesday, May 8, 2012 7:13:30
On 05/08/2012 04:32 PM, Karsten Wemheuer wrote:
Hi,
Am Dienstag, den 08.05.2012, 14:13 +0200 schrieb Jonas Kellens:
Hello,
when a call comes in and is answered by colleague A, this colleague A
sees the CallerID of the external calling number.
When colleague A transfers the call to colleague
@sub:15] NoOp(SIP/kal3-024f,
end) in new stack
There are currently only 8 calls going on...
Kind regards,
Jonas.
Original Message
Subject:Re: [asterisk-users] Mysql identifier not found
Date: Sat, 05 May 2012 12:06:38 +0200
From: Jonas Kellens jonas.kell
Hello,
I notice when upgrading from 1.6.2 to 1.8 that in the menuselect
app_mysql is indicated as deprecated.
If one wants to use the MySQL-command in the dialplan, how to do so if
app_mysql is deprecated ??
Kind regards,
Jonas.
--
Will ODBC become the default then ?
I see no ODBC-command to use in the dialplan.
Jonas.
On 05/05/2012 11:12 AM, Leandro Dardini wrote:
Use ODBC. Check the func_odbc.conf configuration file.
Leandro
2012/5/5 Jonas Kellens jonas.kell...@telenet.be
mailto:jonas.kell...@telenet.be
Hello,
notice in the console output beneath that there is a resultid 6 but it
can not be cleared :
[May 5 11:46:27] -- Executing [s@sub:3] MYSQL(SIP/vart-0336,
Connect connid localhost dialplan host Asterisk) in new stack
[May 5 11:46:27] -- Executing [s@sub:4]
%';
+---+---+
| Variable_name | Value |
+---+---+
| Connections | 2946 |
+---+---+
1 row in set (0.00 sec)
Jonas.
On 05/05/2012 11:53 AM, Jonas Kellens wrote:
Hello,
notice in the console output beneath that there is a resultid 6 but it
can
Hello,
what does it mean when you read in the backtrace file :
Reading symbols from /lib64/libgcc_s.so.1...(no debugging symbols
found)...done.
Loaded symbols for /lib64/libgcc_s.so.1
Core was generated by `/usr/sbin/asterisk -f -vvvg -c'.
Program terminated with signal 6, Aborted.
#0
Hello,
is this an answer or is the most of your answer missing ?
I don't understand what you mean and I still don't understand what
happened with Asterisk.
Kind regards,
Jonas.
On 05/04/2012 09:18 PM, Steve Edwards wrote:
On Fri, 4 May 2012, Jonas Kellens wrote:
Terminated with signal
I have selected don't optimize in the menuselect for better
information in the trace and now you tell me that it's still useless ?
(off course the trace created with bt full is more verbose, but I cant'
post all that)
Kind regards,
Jonas.
On 05/04/2012 11:08 PM, Stephen J Alexander
Hello,
can someone please tell me if this is possible and how ?
Kind regards,
Jonas.
On 04/24/2012 12:59 PM, Jonas Kellens wrote:
Hello,
is there a way to put a certain SIP peer on state busy ?
I know you can do this by pressing DND on your IP-phone, but can
this state also be set
#11# ) to
put your phone on DoNotDisturb so it does not receive any calls.
I want that with Asterisk.
Jonas.
On 04/26/2012 09:53 AM, Leandro Dardini wrote:
Check the command Busy() of the dialplan, it return the busy state
at the calling party.
Leandro
2012/4/26 Jonas Kellens jonas.kell
this bit of information, maybe it'll help:
http://www.voip-info.org/wiki/view/PBX+Do+Not+Disturb
On Tue, Apr 24, 2012 at 12:59 PM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
Hello,
is there a way to put a certain SIP peer on state busy ?
I know
Hello,
is there a way to put a certain SIP peer on state busy ?
I know you can do this by pressing DND on your IP-phone, but can this
state also be set in the dialplan ?
Thanks.
Jonas.
--
_
-- Bandwidth and Colocation
Hello,
backtrace was created. Can anyone help me with understanding it and
telling me what went wrong with my Asterisk-server ? Thanks in advance !
This is Asterisk 1.6.2.22.
[root@sip1 ~]# gdb -se /usr/sbin/asterisk -ex bt full -ex thread
apply all bt --batch -c core.sip1
On 03/21/2012 08:04 PM, Jonas Kellens wrote:
Hello,
when generating backtrace I get following output :
/[root@sip ~]# gdb -se asterisk -ex bt full -ex thread apply all
bt --batch -c core.sip-2012-03-21T10\:57\:29+0100 /root/backtrace.txt
asterisk: No such file or directory./
/warning
...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Thursday, March 22, 2012 12:06 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk generating backtrace
On 03/21/2012 08:04 PM, Jonas Kellens wrote:
Hello,
when generating backtrace I get
:* Tuesday, March 20, 2012 11:08 AM
*To:* asterisk-users@lists.digium.com
*Subject:* Re: [asterisk-users] Cut off + sign in telephonenumber
2012-03-20 16:36, Jonas Kellens skrev:
Hello,
I'm trying to cut off the + sign if part of a telephone number, but
not succeeding :
exten = test,n,Set(cid
Hello,
when generating backtrace I get following output :
/[root@sip ~]# gdb -se asterisk -ex bt full -ex thread apply all
bt --batch -c core.sip-2012-03-21T10\:57\:29+0100 /root/backtrace.txt
asterisk: No such file or directory./
/warning: no loadable sections found in added symbol-file
Hello,
I notice that at the end of the day, after about 4000 calls have passed
my Asterisk-system, that the use of memory is very high and stays that
way untill a restart of Asterisk or a reboot of the server.
This is the situation at the end of the day :
[root@sp asterisk]# free -m
has
more frequent access than the one already present in ram.
If your server isn't swapping.. things are okay.
On Thu, Mar 8, 2012 at 1:15 PM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
Hello,
I notice that at the end of the day, after about 4000
Doesn't this automatically finish ?
Jonas.
On 03/07/2012 05:03 PM, equis software wrote:
Is there any way to do this?
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join
On 02/24/2012 10:51 PM, Jared Geiger wrote:
On Thu, Feb 23, 2012 at 2:48 PM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:
On 01/20/2012 08:07 AM, Jonas Kellens wrote:
Hello
Hello,
when trying to start Asterisk on a new server, I got the following problem :
[root@sip asterisk]# /etc/init.d/asterisk status
asterisk is stopped
[root@sip asterisk]# /etc/init.d/asterisk start
Starting asterisk: Cannot find specified TTY (9)
On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:
On 01/20/2012 08:07 AM, Jonas Kellens wrote:
Hello,
I want to place an Asterisk-server A in front of 2 other
Asterisk-servers (B1 B2).
This first Asterisk-server A needs to send incoming calls to one of the
2 available Asterisk-servers (B1
On 02/23/2012 08:36 PM, Jonas Kellens wrote:
Hello,
when trying to start Asterisk on a new server, I got the following
problem :
[root@sip asterisk]# /etc/init.d/asterisk status
asterisk is stopped
[root@sip asterisk]# /etc/init.d/asterisk start
Starting asterisk: Cannot find specified TTY
Hello,
with the command lsof -i I notice the following network connections of
the asterisk proces :
asterisk 23006 root 12u IPv4 1088961 UDP
*:mgcp-callagent
asterisk 23006 root 13u IPv4 1088964 TCP *:sieve
(LISTEN)
asterisk 23006
On 02/02/2012 11:24 AM, Jonas Kellens wrote:
Hello,
ChanSpy can not be used on a Channel that is being recorded with
MixMonitor.
How can I verify if a channel which I want to spy on, is currently not
being recorded ?!
Anyone with some feedback ?!
I notice that ongoing recordings
On 02/07/2012 01:07 PM, Sammy Govind wrote:
Hello,
I've been managing multiple call centres, almost all of them having
their calls recorded one way or other. Even in PBX environments with
MixMonitor and call recordings I haven't came across the situation
where I discovered that I can't
, 03 Feb 2012 16:43:35 +0100
From: Jonas Kellens jonas.kell...@telenet.be
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
On 02/03/2012 04:32 PM, Paul
On 02/06/2012 12:14 PM, Steve Davies wrote:
On 6 February 2012 10:45, Jonas Kellens jonas.kell...@telenet.be
mailto:jonas.kell...@telenet.be wrote:
Hello,
is there anyone that can give me some more information on these
deadlocks ?!
How can these deadlocks occur and what
On 02/06/2012 12:25 PM, isr...@gmail.com wrote:
Your running into a bug and the only way to solve it is to report it and debug
it and hope for a fix
There is no way someone can help without it being debugged and knowing what's
causing it to lockup
The only key to unlcock it when it gets
On 02/06/2012 03:19 PM, Paul Belanger wrote:
On 12-02-06 09:15 AM, Jonas Kellens wrote:
On 02/06/2012 12:25 PM, isr...@gmail.com wrote:
Your running into a bug and the only way to solve it is to report it
and debug it and hope for a fix
There is no way someone can help without it being
Hello list,
using asterisk 1.6.2.22
What is wrong with Asterisk when the CLI becomes unresponsive ?!
I can login to the CLI with /usr/sbin/asterisk -r but whatever I type,
nothing happens.
The only thing I can do is exit the CLI and do a /sbin/service asterisk
restart. Yes, the command
Hello Anton.
What does this deadlock exactly mean ?
How does it occur and how can I avoid it from happening again ?!
Jonas.
On 02/03/2012 11:12 AM, Anton Kvashenkin wrote:
It seems like simle deadlock.
2012/2/3 Jonas Kellens jonas.kell...@telenet.be
mailto:jonas.kell...@telenet.be
On 02/03/2012 01:05 PM, Mikhail Lischuk wrote:
Jonas Kellens писал 03.02.2012 12:09:
using asterisk 1.6.2.22
What is wrong with Asterisk when the CLI becomes unresponsive ?!
Greetings. I am using the same version, Asterisk 1.6.2.22
IDK is your problem is same as mine. My CLI also
On 02/03/2012 01:46 PM, Steve Davies wrote:
On 3 February 2012 12:12, Jonas Kellensjonas.kell...@telenet.be wrote:
On 02/03/2012 01:05 PM, Mikhail Lischuk wrote:
Jonas Kellens писал 03.02.2012 12:09:
using asterisk 1.6.2.22
What is wrong with Asterisk when the CLI becomes unresponsive
On 02/03/2012 02:52 PM, Paul Belanger wrote:
On 12-02-03 07:55 AM, Jonas Kellens wrote:
This is a production server. Will it affect theserver ?I already have
dont_optimize checked in the debug options.
Yes, reproduce the issue on your test infrastructure. then generate a
backtrace [1].
[1
On 02/03/2012 03:48 PM, Paul Belanger wrote:
On 12-02-03 09:05 AM, Jonas Kellens wrote:
On 02/03/2012 02:52 PM, Paul Belanger wrote:
On 12-02-03 07:55 AM, Jonas Kellens wrote:
This is a production server. Will it affect theserver ?I already have
dont_optimize checked in the debug options
On 02/03/2012 04:32 PM, Paul Belanger wrote:
On 12-02-03 09:53 AM, Jonas Kellens wrote:
On 02/03/2012 03:48 PM, Paul Belanger wrote:
On 12-02-03 09:05 AM, Jonas Kellens wrote:
On 02/03/2012 02:52 PM, Paul Belanger wrote:
On 12-02-03 07:55 AM, Jonas Kellens wrote:
This is a production server
Hello,
ChanSpy can not be used on a Channel that is being recorded with
MixMonitor.
How can I verify if a channel which I want to spy on, is currently not
being recorded ?!
Kind regards,
Jonas.
--
_
-- Bandwidth and
.
On Wed, 2012-01-25 at 10:15 +0100, Jonas Kellens wrote:
This could work, yes.
But the context is not always the same.
Also ${CHANNELS(miq8) will return nothing...
Jonas.
On 01/24/2012 08:47 PM, Danny Nicholas wrote:
Did a little research on this using my Asterisk 10.0. This should
work
On 01/13/2012 06:58 PM, Administrator TOOTAI wrote:
Le 13/01/2012 14:32, Jonas Kellens a écrit :
On 01/13/2012 02:23 PM, Doug Lytle wrote:
Jonas Kellens wrote:
I have the following in dialplan :
[TrunkAccounts]
dialplan show TrunkAccounts
Make sure the sort order is what you're
On 01/25/2012 11:10 AM, Ishfaq Malik wrote:
I use ChanSpy successfully all the time. You do not have to specify the
full channel, just the prefix which is the peer name. As you can see it
also states 'This includes the audio coming in and out of the channel
being spied on.'
I confirm that
(inuse=${CHANNELS(miq8)})
exten = 1246,n,extenspy(${inuse}@default)
exten = 1246,n,hangup()
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Tuesday, January 24, 2012 9:52 AM
*To:* Asterisk Users Mailing List
Hello list,
to use ChanSpy, one needs to know the name of the channel.
But on an incoming call from the provider, or an outgoing call to the
provider there are always numbers added. How can one then know the
channel name ??
/core show channels verbose/ shows me for example :
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Tuesday, January 24, 2012 7:47 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] ChanSpy : how to know channel name ?
Hello list,
to use ChanSpy
to a bridged call which (it seems to me) should pick up both sides.
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Tuesday, January 24, 2012 8:46 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
is in hex). You can control (to a degree) the peer portion in
sip.conf/users.conf.
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Tuesday, January 24, 2012 9:07 AM
*To:* Asterisk Users Mailing List - Non
Hello,
I want to place an Asterisk-server A in front of 2 other
Asterisk-servers (B1 B2).
This first Asterisk-server A needs to send incoming calls to one of the
2 available Asterisk-servers (B1 or B2) behind it.
So I want the first Asterisk-server A to accept the call, and based upon
On 01/20/2012 03:42 PM, Kevin P. Fleming wrote:
On 01/20/2012 08:07 AM, Jonas Kellens wrote:
Hello,
I want to place an Asterisk-server A in front of 2 other
Asterisk-servers (B1 B2).
This first Asterisk-server A needs to send incoming calls to one of the
2 available Asterisk-servers (B1
they are trying to phase out Macros. We are
slowly removing them from our dialplans as time allows for testing.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
*From*: Jonas Kellens jonas.kell...@telenet.be
GoSUBs and everything worked perfectly. Since then I'm
using GoSUBs happily.
On Wed, Jan 18, 2012 at 4:54 PM, Jonas Kellens
jonas.kell...@telenet.be mailto:jonas.kell...@telenet.be wrote:
Can someone confirm that the nesting of macro's or the continuous
and simultaneous use of different
On 01/18/2012 01:51 PM, Matthew Jordan wrote:
Anyone else ? Maybe one of the developers can confirm this risk of
working with macros ?
I don't think you need an Asterisk developer to tell you the risks of using
macros in deeply nested situations. Quoting the documentation of Macro:
Because
Hello list,
where can I post the output of the trace taken from a file :
/tmp/core.sip.pbx.tld-2012-01-17T11:09:13+0100
I want someone to tell me what went wrong.
Kind regards,
Jonas.
--
_
-- Bandwidth and Colocation
Hello list,
can I conclude that it is better to use sub's in stead of macro's ?
I read the following in an Asterisk-book :
GoSub() works in a different manner from Macro(), though, in that it
doesn't have the stack space requirements, so
it nests effectively. Essentially, GoSub() acts like
Hello Paul,
this is a second backtrace taken from a core dump file later on. See
attachment.
Kind regards,
Jonas.
On 01/17/2012 03:59 PM, Paul Belanger wrote:
On 12-01-17 05:16 AM, Jonas Kellens wrote:
Hello list,
where can I post the output of the trace taken from a file :
/tmp
Hello,
I have the following in dialplan :
[TrunkAccounts]
exten = 32380837,1,GoTo(01,32380837,1)
exten = 32380838,1,GoTo(01,32380838,1)
exten = 32380839,1,GoTo(01,32380839,1)
[CheckOnNet]
include = TrunkAccounts
But when a call for 32380837 enters CheckOnNet, it is not found.
On 01/13/2012 02:23 PM, Doug Lytle wrote:
Jonas Kellens wrote:
I have the following in dialplan :
[TrunkAccounts]
dialplan show TrunkAccounts
Make sure the sort order is what you're expecting.
Doug
Hello,
The order is correct for as far as I'm sure.
[TrunkAccounts]
exten
On 01/13/2012 02:37 PM, Andreas Sikkema wrote:
On 1/13/12 2:32 PM, Jonas Kellens wrote:
So the context TrunkAccounts is not included.
Do you know why ?
Does reloading the dialplan (dialplan reload) give any useful output
relating to these two contexts?
I include this context in 2 other
On 01/13/2012 02:59 PM, Doug Lytle wrote:
Jonas Kellens wrote:
Everything works fine when including context 'TrunkAccounts' in
context 'PROVIDERin
dialplan showPROVIDERin
Doug
Meaning ?
--
_
-- Bandwidth and Colocation
On 01/13/2012 03:07 PM, Doug Lytle wrote:
Jonas Kellens wrote:
Meaning ?
Meaning I want to see the dialplan order of that context. I'm
guessing that's your inbound context. With includes that also include
sub-contexts.
Usually, there is something ordered differently then expected
On 01/13/2012 04:22 PM, Doug Lytle wrote:
Jonas Kellens wrote:
Does this mean the Return() comes before Asterisk looks into the
context [TrunkAccounts] ??
No, I believe the includes are read first, but the order in important.
Since you may be matching against another context that may cause
Hello list,
any idea why this call goes to the extension 3292000101 :
/INVITE sip:3292000...@ip.ip.ip.ip:5060 SIP/2.0
Call-ID: otrc74rls5c2pbyulb3hsjz...@ip.ip.ip.ip
CSeq: 102 INVITE
From: 32433885116 sip:32433885...@ip.ip.ip.ip;tag=74706
Via: SIP/2.0/UDP
On 12/27/2011 08:45 PM, Kevin P. Fleming wrote:
On 12/27/2011 01:43 PM, Jonas Kellens wrote:
Hello list,
any idea why this call goes to the extension 3292000101 :
/INVITE sip:3292000...@ip.ip.ip.ip:5060 SIP/2.0
Call-ID: otrc74rls5c2pbyulb3hsjz...@ip.ip.ip.ip
CSeq: 102 INVITE
From: 32433885116
Hello,
when using BLF with Asterisk 1.6, I notice that the Caller-ID
information is not displayed on the monitoring key of my Innovaphone IP200A.
If the IP-phone of my colleague rings, I should see on my partner key
the number of the caller. This is information that is being send in the
-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, December 15, 2011 9:42 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] Partner Keys on Innovaphone
Hello,
when using BLF
-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Thursday, December 15, 2011 9:47 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] Partner Keys on Innovaphone
But blind transfer has
Hello,
the wav sound files that are created by using MixMonitor()-command are
not playable with Windows Media Player.
I can play them with vlc-player and on my Fedora with Totem.
This is one of the files :
/var/ftp/104/2011-11-30_11:54:39_89000404.wav: RIFF (little-endian)
data, WAVE
seems kosher, have you tried just renaming the
file without the -, _ and : ?
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Wednesday, November 30, 2011 1:55 PM
*To:* Asterisk Users Mailing List - Non-Commercial
said : I can play the sound file with Totem on Linux or
VLC-player on Windows. So it's not that the wav-file has no sound...
Jonas.
*From:*asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of *Jonas
Kellens
*Sent:* Wednesday, November 30
On 11/28/2011 04:48 PM, Bryant Zimmerman wrote:
I think I may have found a partial answer to my own question but I
have not come up with any examples. Anyone have any examples for
PARKINGLOT
PARKINGDYNAMIC
PARKINGDYNCONTEXT
PARKINGDYNPOS
Hello Bryant,
I would also like to know how to make
On 11/22/2011 06:13 PM, Alex Vishnev wrote:
it is strange that Aastra acks 401, sends another invite but does not increase
CSeq. Is that the same behavior with others?
On Nov 22, 2011, at 11:51 AM, Jonas Kellens wrote:
This is a trace taken when an Alcatel-Lucent PBX sends an INVITE
Hello list,
this is the communication between an Aastra 5000 PBX and Asterisk, where
the Aastra makes a call to Asterisk. For some reason, Asterisk responds
with 401-Unauthorized and I don't know why.
Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT
with this Aastra.
On 11/22/2011 04:25 PM, Bruce Ferrell wrote:
Jonas,
May I suggest that you present us your sip.conf entry for this peer,
properly redacted, of course. That might help more. What I do for
gateways at known addresses is to put an entry like this into the
sip.conf entry:
[peer]
type=peer
On 11/22/2011 04:37 PM, Bruce Ferrell wrote:
On 11/22/2011 07:29 AM, Jonas Kellens wrote:
On 11/22/2011 04:25 PM, Bruce Ferrell wrote:
Jonas,
May I suggest that you present us your sip.conf entry for this peer,
properly redacted, of course. That might help more. What I do for
gateways
On 11/22/2011 05:31 PM, Alex Vishnev wrote:
Your registration should have also have the flow
PEER ASTERISK
REGISTER---
--401
REGISTER(nonce) -
200OK
Then the server controls the life of the registration and 200 Expires
Header
On 11/22/2011 05:42 PM, Alex Vishnev wrote:
I doubt it. Unknown headers should be ignored by UAS. is it possible
to post the trace?
On Nov 22, 2011, at 11:39 AM, Jonas Kellens wrote:
What trace do you need ? Have you read my original post ? Asterisk SIP
debug trace is posted in my original
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