search for: markus_weil

Markus Weiler <markus_weiler at mailworks.org> schrieb: 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+...

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 /tmp/sample-call-file >

Just a test. -- Matthew Fredrickson Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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? -- ----------------------------- Markus Weiler markus_weiler at mailworks.org -----------------------------

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

Hi again I'd like to configured my Asterisk to use german sounds for the "Say"-commands... I installed the sounds-files and I tried them with "Playback(de/demo-echodone)" and it works. Now I tried to add an extension to say the current time: exten => 24,1,Verbose(2,Time asked by ${CALLERID(num)}) Exten => 24,n,Set(CHANNEL(language)=de) 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 = 494XXXXXXXXX,n,Dial(SIP/4120089,15,w) exten = 494XXXXXXXXX,n,Goto(98-${DIALSTATUS},1) exten = 494XXXXXXXXX,n,Hangup() ..... exten = 98-BUSY,1,NoOp(Busy) exten = 98-BUSY,n,ExecIf($["${Voicemail}" =

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. What's the status with Asterisk REST API? Any experiences on performance,stability,documentation,

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. Does anyone have any advice or links to advice on measuring call >

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+SayDigits>command may play the sound file "digits/6". Asterisk will, if the language code is "de", first look for

Hi, we're using the limit option like this: Dial ....L(60000:30000) [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 = 60000 [Jun 10 16:14:41] VERBOSE[12196] logger.c: [Jun 10 16:14:41] > play_warning = 30000 [Jun 10 16:14:41] VERBOSE[12196]

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 "its own" see also https://issues.asterisk.org/view.php?id=16901 I Guess the Problem