I received it :-)
Am 28.03.2018 um 22:44 schrieb Matt Fredrickson:
Just a test.
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Hi Group,
we're just wondering, in German we call the different types of phone-numbers
(Geographic,mobile,national,VoIP...)
Rufnummerngassen (phone number alleys ;-) )
Is there an english word for this?
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Markus Weiler
markus_wei...@mailworks.org
Hi All,
this seems to be a really neat module, that could really help us.
https://github.com/drivefast/asterisk-res_json
Any opinions about if we should use it in a production system?
Maybe from "official" asterisk side?
thanks!!
Markus
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Hi Derek,
I think Homer (http://sipcapture.org/) is the right answer :-)
HEP Agent will send the SIP trace to a remote Server (res_hep).
Markus
Am 18.02.2017 um 00:18 schrieb Tim Pozar:
You can tell it to just capture SIP traffic and not the RTP traffic.
Nice write up of using TCPdump and
Nobody any idea?
It would be really helpful,
Markus
Am 06.01.2017 um 12:07 schrieb Markus Weiler:
Hi List,
we're calling a sip phone from our Asterisk Server, and try to add logic
depending on the dialstatus
Stripped down example;
exten = 494X,n,Dial(SIP/4120089,15,w)
exten
Hi List,
we're calling a sip phone from our Asterisk Server, and try to add logic
depending on the dialstatus
Stripped down example;
exten = 494X,n,Dial(SIP/4120089,15,w)
exten = 494X,n,Goto(98-${DIALSTATUS},1)
exten = 494X,n,Hangup()
.
exten = 98-BUSY,1,NoOp(Busy)
Am 20.08.2015 um 03:16 schrieb Pete Mundy:
Ah cr@p, sorry Steve, didn't mean to top-post there.
On 20/08/2015, at 5:23 AM, Markus Weiler markus_wei...@mailworks.org
mailto:markus_wei...@mailworks.org wrote:
We started the 500 calls and used milliwatt app on the first and
record
Am 19.08.2015 um 19:07 schrieb Steve Edwards:
Please don't top post.
On Wed, 19 Aug 2015, James Cass wrote:
Steve, would you be willing to share that quick bash script?
There's no magic in the script, but here it is, embarrassing myself:
cp sample-call-file /tmp/
chmod +x
Hi Stefan,
we ran into a similar problem using Debian.
There we are able to check the current limits using:
pidof asterisk - 23351
cat /proc/23351/limits
Output:
Limit Soft Limit Hard Limit Units
Max open files1024 1024 files
I
Hi Eric,
This is not really easy. Especially the Mail2Tiff Conversion is tricky
(lots of different MIME/File formats).
When you have the correct tiff file the rest ist easy. Try to narrow it
down to an empty Mail Body using (one) PDF attachment.
We used a self written Java app to prepare the
great,
would be the ideal time to comment the www.voip-info.org to contribute
to the community :-)
Markus
Am 14.06.2015 um 09:42 schrieb Luca Bertoncello:
Markus Weiler markus_wei...@mailworks.org schrieb:
Hi
from voipinfo...
If an Asterisk command specifies a sound file
Hi,
from voipinfo...
If an Asterisk command specifies a sound file in a*subdirectory*,
Asterisk looks in that subdirectory for the language subdirectory. For
example, theSayDigits
http://www.voip-info.org/wiki/view/Asterisk+cmd+SayDigitscommand may
play the sound file digits/6. Asterisk
definitely DNS...
check your Register lines...
Markus
Am 27.05.2015 um 20:14 schrieb Duncan Turnbull:
DNS failure could do this
Asterisk used to get stuck in a symmetric DNS request wait state which meant
everything ground to a halt as it waited for a reply while DNS timed out.
The
Hi Patrick,
try voipmon, there it's free and you can even track MOS.
Markus
Am 25.03.2015 um 14:21 schrieb Patrick Beaumont:
Hi everyone.
We regularly get customers complaining about call quality issues. Most of
the time it turns out to be their own broadband. Very occasionally server
load.
Hi all,
currently we're looking to program a new asterisk application. Years ago
we used AMI and Asterisk Java.
When we did this we pretty soon encountered performance issues when
using a lot of channels.
We want to place calls, bridge channels, disconnect channels, monitor
them, hangup.
Hi Nick,
maybe this will help?
exten = _XXX,n,Dial(SIP/${EXTEN})
exten = _XXX,n,NoOp(SIP return code :
${HASH(SIP_CAUSE,${CDR(dstchannel)})})
(http://www.voip-info.org/wiki/view/Asterisk+variable+hangupcause)
Markus
Am 27.02.2015 um 18:56 schrieb Nick Olsen:
Hello Everyone.
In my outbound
very simple,
yet effective
http://www.palner.com/blog/171/asterisk-no-matching-peer-found-block/
Am 27.06.2014 16:58, schrieb Steven Howes:
On 27 Jun 2014, at 15:37, Anurag Rana anuragrana31...@gmail.com
mailto:anuragrana31...@gmail.com wrote:
There are lot of requests coming in and I am not
Am 03.01.2013 21:21, schrieb Nick Khamis:
Oh that's so smart!!! So, if I did not misunderstand you, for this one
call, have:
rtpstart=10004
rtpend=1008
do you mean 1_000_8 ?
Markus
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Hi,
one more hint... (trying to translate the commands to english)
in Thunderbird open - Extras - Filter.. -
Filter-Name: enter Top Posting
Subject - Contains: enter Top Posting
Action: Delete
Markus
Am 02.01.2013 21:31, schrieb Steve Totaro:
On Wed, Jan 2, 2013 at 12:25 PM,
Hi Felix,
ngrep -W byline port 5060|grep -B1 INVITE sip
Markus
Am 16.11.2012 17:50, schrieb Ruben Rögels:
Hi Felix,
you have several things to check:
netstat -a -n --udp --tcp
will show you connections and connection attempts on network layer level.
You have to look for incoming
hi,
try to catch in in a cron job per minute.
asterisk -rx 'module unload res_musiconhold.so'
Markus
Am 13.11.2012 19:15, schrieb Markus:
Am 13.11.2012 19:01, schrieb Eric Wieling:
module unload res_musiconhold.so
and
module load res_musiconhold.so
Great, that works, but only if no
Hi,
I suppose WebRTC is the best solution nowadays, it's extremely interesting.
I developed a C2C app in 2008, starting with call files and AMI, ended
with asterisk-java and asterisk.NET to solve it.
Hint: Try to solve (al)most (all) of your problems using
Dialplans/Variables. Basically it's
Hi,
we are facing the problem , that we cannot distinguish between a trunk
an an extension.
On our trunk side, if the remote user puts us on hold the same
Musiconhold is played as if we would call another extension on the sam
Asterisk PBX.
Asterisk should play the music from the remote End not
Hi,
I still don't know if it's a bug or if it's already fixed esp. what
exactly is the source...how could i find this out?
or where could i open the bug report?
thanks
Markus
Am 02.01.2010 19:16, schrieb Steve Totaro:
Did you open a bug report?
On Sat, Jan 2, 2010 at 12:37 PM, Markus Weiler
Could anybody give me a hint how to investigate that problem?
cheers Markus
Am 31.12.2009 18:17, schrieb Markus Weiler:
Sorry wrong topic...
Hi,
I'm issuing a Bridgeaction through the manager interface.
One Person is called, when answered second one is called first gets MoH.
After
Hi,
I'm issuing a Bridgeaction through the manager interface.
One Person is called, when answered second one is called first gets MoH. After
the second person
answers both channels are bridged together.
Randomly (approx. 1/5.000 calls (sometimes twice a day, sometimes once a week))
asterisk
Sorry wrong topic...
Hi,
I'm issuing a Bridgeaction through the manager interface.
One Person is called, when answered second one is called first gets MoH. After
the second person
answers both channels are bridged together.
Randomly (approx. 1/5.000 calls (sometimes twice a day, sometimes once
Hi,
we're using the limit option like this:
Dial L(6:3)
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] --
Limit Data for this call:
[Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41]
timelimit = 6
[Jun 10 16:14:41] VERBOSE[12196]
Hi,
In VI:
In 'vi', moving the cursor over any bracket,
brace, etc, and then pressing '%' moves the cursor to the 'matching'
bracket/brace character.
That can be very useful when programming, to find missing/extra
brackets and braces. It even seems to find matching pairs of #ifdef /
#endif
Hi,
I installed Digiums Free Fax for Asterisk and found out, that it
automatically retries failed faxes, is there a way to stop that?
Thanks
Markus
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asterisk-users mailing
Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Weiler
Sent: Wednesday, 13 May 2009 5:44 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Free Fax for asterisk
Hi,
I installed Digiums Free Fax
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