/0031650747314-3):
CHANNEL(uniqueid)=dijkivr04-nwg.mobillion.biz-1353083131.2
ChanVariable(DAHDI/i1/0031650747314-3): AJ_TRACE_ID=AJ_ORIGINATE_2
-- above: AJ_TRACE_ID set
Regards,
Arjan Kroon
Mobillion BV
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Hello,
I have a problem.
One every couple of months my asterisk system crashes with a segmentation fault.
kernel: asterisk[20527]: segfault at 0808 rip 2aaac952d8f2 rsp
40edb910 error 4
(This is in /var/log/messages)
If I look at the same timestamp in the warning log
I'm using Bria,but X-Lite from counter path
I have good result with these programs under Lion
On 26 Apr 2012, at 12:05 PM, Alex Balashov wrote:
Have you looked into Blink?
On 04/26/2012 05:41 AM, Paolo Supino wrote:
Hi
I'm looking for a SIP client for Mac OS X (I'm running Lion) that
.
local_lostpackets = 7706
local_jitter = 2
local_maxjitter = 11
local_minjitter = 0
..
..
remote_lostpackets = 0
remote_jitter = 0
remote_maxjitter = 7
remote_minjitter = 14000
..
..
The only thing I see is this:
http://www.voip-info.org/wiki/view/Asterisk+func+channel
Regards,
Arjan Kroon
Hi,
Does anybody know if RAMI (Ruby Ami) is still functional?
And is this still compatible with asterisk 1.8
Best Regards,
Arjan Kroon
Mobillion BV
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voice apps.
On Wed, Jan 4, 2012 at 2:49 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nlmailto:arjan.kr...@mobillion.nl wrote:
Hi,
Does anybody know if RAMI (Ruby Ami) is still functional?
And is this still compatible with asterisk 1.8
Best Regards,
Arjan Kroon
Mobillion BV
...@lists.digium.com] On Behalf Of Arjan Kroon |
Mobillion
Sent: Tuesday, October 04, 2011 9:43 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Beep file with Record
Yes,
In the code I use set the language
exten = s,n,Set(CHANNEL(language)=nl/fvdb)
So
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
How are you calling the beep.alaw from the dialplan?
paste the relevant dialplan here and corresponding CLI logs.
On Wed, Oct 5, 2011 at 11:58 AM, Arjan Kroon | Mobillion
arjan.kr
and if it goes well revert back language after the
recording.
On Wed, Oct 5, 2011 at 12:20 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nlmailto:arjan.kr...@mobillion.nl wrote:
CLI::
-- Executing [s@ serviceline:93] Record(CAPI/ISDN1#02/318647615-37,
/var/lib/asterisk/sounds/recordings
/sounds/recordings/serviceline/${UNIQUEID)
exten = s,n,Record(${A_serviceline_file}.wav,0,60)
exten = s,n,Set(CHANNEL(language)=nl))
On Wed, Oct 5, 2011 at 12:29 PM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nlmailto:arjan.kr...@mobillion.nl wrote:
Yes I already try this (only with language nl
These are the directories which I gave in asterisk.conf
astetcdir = /etc/asterisk
astmoddir = /usr/lib64/asterisk/modules
astvarlibdir = /usr/share/asterisk
astdbdir = /var/spool/asterisk
astkeydir = /var/lib/asterisk
astdatadir = /usr/share/asterisk
astagidir = /usr/share/asterisk/agi-bin
Yes, That was the solution.
Thanks.
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Jeroen Eeuwes
Verzonden: 05-10-2011 10:15
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re:
Hi,
I'm using the functionality Record in asterisk 1.8.5.
But when I want to record something I get the following error message:
file.c:644 ast_openstream_full: File beep does not exist in any format
Could anybody tell me where I have to place the beep.gsm file?
I already tried the following
because you have to do record(foo.gsm) but you have
to playback using playback(foo).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Arjan Kroon |
Mobillion
Sent: Tuesday, October 04, 2011 9:21 AM
To: asterisk
...@lists.digium.com] Namens Andrew Latham
Verzonden: 04-10-2011 16:41
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On Tue, Oct 4, 2011 at 11:37 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
This is my complete CLI logging
Discussion
Onderwerp: Re: [asterisk-users] Beep file with Record
On 10/04/2011 10:37 AM, Arjan Kroon | Mobillion wrote:
exten =
s,n,Set(A_serviceline_file=/var/lib/asterisk/sounds/recordings/serviceline/${UNIQUEID)
Hello Arjam,
Did you notice that there's a missing '}' around the end of the line
Hi,
I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk
Could anybody give me an advise which card I can use?
Regards,
Arjan Kroon
Mobillion.
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something really professional, for Serverside, I advise you
sangoma.
Tamer
Am 06.09.2011 09:08, schrieb Arjan Kroon | Mobillion:
Hi,
I'm looking for a PCIe card with 1 ISDN2 connection which works with Asterisk
Could anybody give me an advise which card I can use?
Regards,
Arjan Kroon
Hi,
Is there a easy way to configure the sip settings so it is not possible to
register more than one sip user with the same username/password.
Now it is possible to register more than one sip user with the same
username/password.
So if I call that sip user, both sip clients will ring.
...@lists.digium.com] Namens Doug Lytle
Verzonden: 10-06-2011 14:01
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Connected Line ID
Arjan Kroon | Mobillion wrote:
But are there also pathes for version 1.6
The last patch available for the 1.6 series
Discussion
Onderwerp: Re: [asterisk-users] Connected Line ID
Arjan Kroon | Mobillion wrote:
And if I can see it, this patch is already included in version 1.6.2.12. Or
am I wrong?
That I can't answer. I'm still using 1.4.x and am experimenting with
1.8.x. I recall reading that it wasn't
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle
Verzonden: 20-06-2011 13:11
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Connected Line ID
Arjan Kroon | Mobillion wrote
Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Connected Line ID
On Mon, Jun 20, 2011 at 5:39 AM, Arjan Kroon | Mobillion
arjan.kr...@mobillion.nl wrote:
Oke,
But is there a patch from version 1.6.2.12?
Greeting,
Arjan
-Oorspronkelijk bericht-
Van
, but this isn't
the case in version 1.6
Regards,
Arjan Kroon
Mobillion BV
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New to Asterisk? Join us for a live introductory webinar every Thurs
[mailto:asterisk-users-boun...@lists.digium.com] Namens Doug Lytle
Verzonden: 10-06-2011 12:58
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] Connected Line ID
Arjan Kroon | Mobillion wrote:
Does anybody have problems with a wrong Connected Line ID
Hi,
I'm using asterisk version 1.8.3.3.
In earlier versions I used queues, but with the new version the queuing
mechanism doesn't work
If I look in the CLI at I see that the queue-member is invalid:
Members:
DADHI/g3/0655871460 (Invalid) has taken no calls yet
The
Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension
after Dial command completes.
On Monday 04 April 2011 06:58:23 Arjan Kroon | Mobillion wrote:
Hi,
Does anybody have a solution to this problem?
Because in this issue
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Arjan Kroon | Mobillion
Verzonden: 05-04-2011 08:21
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension
after Dial command completes.
Hi
-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Arjan Kroon | Mobillion
Verzonden: 05-04-2011 09:32
Aan: Asterisk Users Mailing List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension
after
Hi,
Does anybody have a solution to this problem?
Because in this issue the solution is not mentioned.
https://issues.asterisk.org/view.php?id=18522
Arjan Kroon
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List - Non-Commercial Discussion
Onderwerp: Re: [asterisk-users] CDR fields not being written from h extension
after Dial command completes.
On Mon, 2011-04-04 at 13:58 +0200, Arjan Kroon | Mobillion wrote:
Hi,
Does anybody have a solution to this problem?
Because in this issue
Maybe this helps:
https://issues.asterisk.org/view.php?id=18603
Arjan
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Jerry Geis
Verzonden: 20-03-2011 21:24
Aan: Asterisk Users Mailing List - Non-Commercial
actionid: 129675971_656137#
variable: CALLERID(dnid)
channel: DAHDI/11-1
Arjan Kroon
Mobillion BV
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New to Asterisk? Join us for a live introductory webinar
Hi,
We had the same problems.
These problems accours when we try to send (from different servers) a lot of
IAX calls to one server. (see couple of 100 calls at the same time)
When we upgraded asterisk to version 1.8 we didn't get these problems.
Regards,
Arjan Kroon
Van:
span=7,1,0,ccs,hdb3,yellow
bchan=187-201,203-217
dchan=202
span=8,1,0,ccs,hdb3,yellow
bchan=218-232,234-248
dchan=233
We use two seperate cards. (TE4/1/3 T4XXP (PCI))
Arjan Kroon
Mobillion BV
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-- Bandwidth and Colocation
(sbs)
The sbs stream is a mp3 stream with a bitrate of 64/128 kpbs
The Hitz stream I don't know what kind of stream this is? Maybe someone knows
this?
Does anybody have an idea how the sbs stream must be streamend?
Regards,
Arjan Kroon
Mobillion BV
-Oorspronkelijk bericht-
Van: asterisk
__ast_read: Dropping
incompatible voice frame on SIP/arjankroon- of format gsm since our
native format has changed to 0x4 (ulaw)
I'm using asterisk 1.8
Can anybody help me?
Kind regards,
Arjan Kroon
Mobillion BV
Hi Dhaval,
I 'm in the almost same situation.
I've already post a issue with asterisk.
https://issues.asterisk.org/view.php?id=17826
Is you only use an originate and not an originate en then redial maybe this
link helps you further.
https://issues.asterisk.org/view.php?id=17592nbn=16#bugnotes
capability: 0x00 - SPEECH
-- Called g1/0031655871460
-- DAHDI/2-1 is proceeding passing it to DAHDI/1-1
-- DAHDI/2-1 is ringing
-- DAHDI/2-1 answered DAHDI/1-1
Does anybody have this same problem, or does anybody knows a solution?
Asterisk Version: 1.6.2.9
Arjan Kroon
Mobillion BV
Mayby Freepbx.
http://www.freepbx.org/
Regards,
Arjan Kroon
-Oorspronkelijk bericht-
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Christian
Verzonden: 09-07-2010 14:41
Aan: asterisk-users@lists.digium.com
Onderwerp:
?
Example:
Group 1
channel = 1-15,17-31
channel = 32-46,48-62
group=2
channel = 63-77,79-93
channel = 94-108,110-124
Regards,
Arjan Kroon
Mobillion BV
--
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New
channel = 1-15,17-31
channel = 32-46,48-62
group=2
channel = 63-77,79-93
channel = 94-108,110-124
We are using the group number for the dial en originate command.
For example: Zap/g3/0612345678
Regards,
Arjan Kroon
Mobillion BV
Hi,
I have a question about the dial command.
Is the following scenario possible.
1)
- Our asterisk server had a successful outbound call.
- Our asterisk server has to call another caller and when
answered asterisk has to connect this call to the another outbound
Hi,
Does anybody have any experience with asterisk where are four PCIe cards
are used in one server (TE420).
So you can have max 4 * 4 * 30 channels = 480 channels used.
Regards,
Arjan Kroon
Mobillion BV
but maybe save you some worries.
Christian
2010/2/22 Arjan Kroon | Mobillion arjan.kr...@mobillion.nl:
Hi,
Does anybody have any experience with asterisk where are four PCIe cards are
used in one server (TE420).
So you can have max 4 * 4 * 30 channels = 480 channels used.
Regards,
Arjan
;
done
Regards,
Arjan Kroon
Mobillion BV
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com
I don't know if you server is running under Unix.
If so, here is a wiki link about mounting
http://en.wikipedia.org/wiki/Mount_%28Unix%29
Van: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens ABBAS SHAKEEL
Verzonden: 21-10-2009 08:59
Aan:
Maybe a central server is an idée.
You'll have to mount an directory on server A, B and C to a directory on the
central server.
A disadvantage is, that you'll have to have a stable internet connection
between al servers.
Another solution is to make a script on the server A,B and C that
how to handle a AGI(..) returns -1 condition?
thx
Arjan Kroon
Mobillion BV
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...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] Namens Michiel van Baak
Verzonden: 26-06-2009 11:29
Aan: asterisk-users@lists.digium.com
Onderwerp: Re: [asterisk-users] Centrale FastAgi server down
On 10:42, Fri 26 Jun 09, Arjan Kroon | Mobillion wrote:
Hi,
How do you all handle
Hi,
Your correct, this the best way.
But we don't have any 'balancing' on the localhost.
In some cases we have to connect directly to a central database. (we
have only one central database)
If the machine where the central database is running on, is down, than
FastAgi will try to connect to
Hey,
I record the message in ULAW
exten = s,1,Record(${A_record}:ulaw,0,60)
After that I call sox with this command:
/usr/bin/sox -c 1 -1 -t ul -r 8000 $in_fl -t wav -2 -r 8000 -c 1
$wav_fl
Regards,
Arjan Kroon
Mobillion BV
-Oorspronkelijk bericht-
Van: asterisk-users-boun
Hi,
Is it possible top use a form of Karaoke Functionality?
When a caller calls a number, he hears a voicefile.
During this voicefile he sings along with this voicefile.
Is it possible to record what the caller is singing?
Grt,
functionality
Hi,
Why not use MixMonitor(), so you have a single file with the singer
and the music?
Thanks.
Andy
On 5/20/08, Sherwood McGowan [EMAIL PROTECTED] wrote:
Arjan Kroon | Mobillion wrote:
Hi,
Is it possible top use a form of Karaoke Functionality?
When a caller calls
Wolfe
Sent: woensdag 9 april 2008 17:19
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] queue logging
You could ASTassistant to see this. Its Freeware.
www.astassistant.com
- Original Message -
From: Arjan Kroon | Mobillion
Hi,
I' using with asterisk a queue with tree members and round robin.
When a caller enters this queue and it is connecting to one of the
members, is there a possibility to see which member the caller is
connected to?
And is there a way to see in de application to see if the connection
http://www.voip-info.org/tiki-index.php?page=Asterisk+variable+hangupcau
se
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tobias
Ahlander
Sent: maandag 17 maart 2008 15:35
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users]
Sorry,
I tried to use underscore(s) before the variable names, but without any
success.
H234m_gw is a functionality which we use for video calling on asterisk.
(http://sip.fontventa.com/)
--
Arjan Kroon
Mobillion BV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED
and during the intro.3gp I press the #-key the
call will be ended.
But I got three different CDR's.
Does anybody know how I can use one CDR instead of 3 different CDR's
Kind Regards,
Arjan Kroon
Mobillion BV
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Hi,
I'm using videocalling on asterisk 1.4.10.
When I setup the videocall with exten = n,1,h324m_gw([EMAIL PROTECTED]),
I loose the variable DNID (${CALLERID(dnid)})
Before the videocall is set up, this variable is filled and after this
videocall this variable is empty.
Also all local
Hi,
In my application I jump to different extensions
For example:
[begin]
exten = s,1,Goto(starts,s,1)
[start]
exten = s,1,Play(welkom)
.
exten = h,1,Goto(end,s,1)
[end]
exten = s,1,Macro(end_call)
exten = s,n, Hangup
When I look at my CDR record I see three
] SET with pipe symbol
On Tuesday 29 January 2008 08:32:44 Arjan Kroon | Mobillion wrote:
I want to place a pipe symbol in a variable by using the command Set
I tried the following code:
Set(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))
When I call to my applicatie I see
Hi,
I want to place a pipe symbol in a variable by using the command Set
I tried the following code:
Set(M_CHANNELVAR=${UNIQUEID}|${CALLERID(number))
When I call to my applicatie I see the following output in my CLI :
Ignoring entry '612345678' with no = (and not
] On Behalf Of Tilghman
Lesher
Sent: dinsdag 30 oktober 2007 15:41
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Size of Exten when using IAX
On Tuesday 30 October 2007 08:40:51 Arjan Kroon | Mobillion wrote:
We are use IAX protocol between two asterisk
Hi,
We are use IAX protocol between two asterisk servers.
Now we send information through this protocol by using EXTEN
We see that the variable EXTEN only holds 66 characters.
If we set a value larger then 66 characters, for example 70 characters.
The last 4 characters are cut off.
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