search for: motiejus

...for it (cannot enable it). I need jack... Otherwise I will inotify Monitor WAVs, what is bad :`( After installing jack from sources: Add system-wide PKG_CONFIG_PATH=/usr/local/lib64/pkgconfig so pkg-config --libs jack recognizes it. Modified /etc/ld.so.conf so libjack.so is cached in ld.so.cache: motiejus at pbx3:/etc$ strings ld.so.cache | grep jack libjackserver.so.0 /usr/local/lib64/libjackserver.so.0 libjackserver.so /usr/local/lib64/libjackserver.so libjack.so.0 /usr/local/lib64/libjack.so.0 libjack.so /usr/local/lib64/libjack.so So... When I run in asterisk source dir: ./configure --disable-...

Hello, I need to capture calee's audio in real-time in order to capture operator messages (I've written sound recognition software that works with Jack: http://github.com/Motiejus/SoundPatty/). Jack does the following: Incoming call audio -> audio in to jack, audio out from jack -> current Asterisk application Outgoing call audio <- current Asterisk application However, I need vica-versa: Incoming call audio -> current Asterisk application Outgoing call audio &...

...angs + theoretical network failover (hmm since half a year my home has 100% connectivity, tested with nagios. But for the record - I have GSM failover...) :-) What are the options? Any phones I should avoid? Maybe anyone of you have done something by yourself? :) Best wishes, have a nice weekend! Motiejus

...ut overheads, system loads and other possible gotchas in this setup. Is there anything I should (re-)consider before implementing this? Anyone had difficulties running VoIP or VPN traffic over (virtualized if it makes any difference) VPN? We use mainly g729 and speex, and very little g711. Regards Motiejus

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hello there, I have successfully installed and configured asterisk for use as an office PBX using SIP trucks and Voip handsets (using g.729 codec) which works great. Now I wish to try and configure asterisk to do a HTTP request and submit callerID to an external website when a call is missed. eg Someone calls PBX and rings extension 100 -> Call is not answered -> HTTP request is initiated

Ok, here's the challenge: I would like to be able to find, match - and then react - upon prompts that are presented by the outbound/remote side of a call. Think mobile phone and "This user is temporarily unavailable". Collecting a limited number of known prompt snippets should not be a problem, but how would you then detect their presence in a longer recording (or live audio

Hello asterisk users! I, like many people, have a cell phone. I also have some SIP phone devices (software and hardware). I'd like to have one number that rings all my phones and routes the call to wherever I pick up. However, my cell phone has its own call forwarding voicemail. I can't just turn that off, because then direct-to-cell calls wouldn't ever get to voicemail - that

...- Auto fallthrough, channel 'SIP/PBX2-00000005' status is 'UNKNOWN' == End MixMonitor Recording SIP/1001-00000004 The question: how to execute G() right after answering (purpose behind this: I will need to set some leg B variables), and then continue the conversation? Thank you, Motiejus

...ck -- Executing [h at NPDB2:4] GotoIf("SIP/1002-00000201", "1?Bl") in new stack This case there is no cdr. If I do dial() before ResetCDR(), it's fine. With asterisk 1.2 this worked fine. Any clue what's wrong? Asterisk 1.6.2.7 self-compiled, Debian 5.0.5 Regards Motiejus

Hi, Is there a way to check on AMI if a user is currently engage on the phone? i would like to display on my portal whether a user is calling or not. thank you regards Ron

Hello list, I'm using asterisk 1.4.30 and realtime sip. I notice that the field "musiconhold" is not working as when putting someone on hold, the default musiconhold class is always used. musiconhold.conf : [default] mode=files directory=/var/lib/asterisk/moh random=yes ; [106002] mode=files directory=/var/lib/asterisk/moh/106002 random=yes my realtime sip peers have the

On Aug 7, 2007 'Mojo' wrote: Nicholas Blasgen wrote: > I've got 4 SIP phone lines with a call-limit of 2 for each. I've > written a handy macro to allow my users to dial a phone number and the > macro will figure out the next available line to use by first checking > if the GROUP() is over 2 and then checking to see if ChanIsAvail() as a > backup, and if it

Hi, Thanks for your info, What is the impact of the following line in dialplan, Dial(SIP/19201/19202,300) On Thu, Sep 3, 2015 at 7:20 PM, Vinicius Fontes <vinicius at aittelecom.com.br> wrote: > You might want to use the Originate() application instead. Check its usage > by issuing the command 'core show application originate' on Asterisk CLI. > > 2015-09-03

Hi, I have an xlite registered with asterisk server. When i dial a number AGI is invoked. and in this we are running to threads one to record files and one to play files. So i dialed the extension and i started recording and playing at the same time. On the xlite i m getting an indication when recording my voice and at the same time i could see playing the other file too. But in the directory

I use Jack for getting callee sound. Dial with option M(): JACK_HOOK(manipulate,i(rec_737219:input),o(rec_737219:output),c(rec_737219))=on This works fine, but I need to connect the sound channel to Jack *before* the actual answer. As you can see in the AMI log, between "Ringing" to JACK_HOOK there is a 6 second break. I don't want that. I need a way to launch Dialplan function