Re: [asterisk-users] More testing - sorry guys

I received it :-) Am 28.03.2018 um 22:44 schrieb Matt Fredrickson: Just a test. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.as

[asterisk-users] Half Off Topic Questions

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_wei...@mailwork

[asterisk-users] res_json

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 --

Re: [asterisk-users] Turn on SIP debugging from DialPlan

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 w

Re: [asterisk-users] Issue with handling of 480 DND

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)

[asterisk-users] Issue with handling of 480 DND

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)

Re: [asterisk-users] asterisk server stress test

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 <mailto:markus_wei...@mailworks.org>> wrote: We started the 500 calls and used milliwatt app on the first and record on the second hos

Re: [asterisk-users] asterisk server stress test

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

Re: [asterisk-users] 786 000 files limit Centos 7 - Asterisk keep complaining

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 thin

Re: [asterisk-users] asterisk email to fax

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 f

Re: [asterisk-users] German sounds on Asterisk

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 schrieb: Hi from voipinfo... If an Asterisk command specifies a sound file in a*subdirectory*, Asterisk looks in

Re: [asterisk-users] German sounds on Asterisk

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 command may play the sound file "digits/6". Asterisk

Re: [asterisk-users] Strange and complete failure of Asterisk 1.8

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 recomm

Re: [asterisk-users] Call Quality Measuring

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.

[asterisk-users] Asterisk API

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

Re: [asterisk-users] 603 Declined > Dialstatus Busy

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 outboun

Re: [asterisk-users] Attack on Sip server.

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 > wrote: There are lot of requests coming in and I am not able to stop it because

Re: [asterisk-users] Moving User Agent To Remote Location

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 -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] Top Posting

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, j...@mil

Re: [asterisk-users] Intruder

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 connecti

Re: [asterisk-users] Restarting MOH

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 call

Re: [asterisk-users] Web based Click to Call Application

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 j

[asterisk-users] Musiconhold Problem

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

Re: [asterisk-users] Random crashes on Bridgeaction

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

Re: [asterisk-users] Random crashes on Bridgeaction

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 f

[asterisk-users] Random crashes on Bridgeaction

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 a

Re: [asterisk-users] identifying channel for softhangup

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 cra

[asterisk-users] Dial option limit call duration

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] logger

Re: [asterisk-users] Can YOU find a trailing parenthesis?

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

Re: [asterisk-users] Free Fax for asterisk

app_rxfax modules? > > Regards > David. > > -Original 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.co

[asterisk-users] Free Fax for asterisk

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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing lis