search for: quys

Hi quys, I look for a solution for interconnection beetwen PSTN and VoIP. My application have to treat few protocols comming from PSTN lines and mixing data , dtmf and voice. Can I use Asterisk for : PSTN ----------> Asterisk (converting analog call to IP) ----------> MyApplication ( translation p...

Hi all, Here is the script i want to make - Caller call to a number to record a message - Asterisk answer and start recording message as following + User press * to start recording + Record is finished if: + User press # + OR message duration reach 60 second + Hangup How do you counting down 60s, and how to detect # (i make a test using Read() but it cant read #) Thanks in advance

Hi all, I have to play a alaw file with .wav ext. How can I do this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100721/de46328f/attachment.htm

Interesting announcement today; http://www.crn.com.au/News/174872,microsoft-response-point-voip-gets-a-d irt-nap.aspx Microsoft this week revealed its intention to discontinue its Response Point, its small business VoIP system for companies with up to 50 employees. While not unexpected, the move is significant because Response Point was once a promising product in which Microsoft

/var/lib/asterisk/sounds/soundfile.alaw /var/lib/asterisk/sounds/soundfile.wav to go from alaw to mp3, first convert to wav, then use lame <options> /var/lib/asterisk/sounds/soundfile.wav /var/lib/asterisk/sounds/soundfile.mp3 sox looks like it can ogg/vorbis, but mine doesn't list mp3. You might fetch the source for sox and see if it can do mp3; lame is probably just as easy to obtain

Hi all, I've been looking at opus lately as a replacement to vorbis and it sure seems to me that, at least on paper, it is the 'next generation' streaming codec. Has anyone been using opus as a streaming codec to Icecast? If so, how do the following general characteristics compare to vorbis? + Quality + Latency + Bandwidth + Performance (on stream client and Icecast server) +

...t is stable for all streams. > >The is one downside on OPUS at the moment, there are not that much >clients able to play opus, plus on track change the stream makes a >little frame noticeable dropout. I do not know wheather this is cased >by >liquidsoap or the player. > >The quys from liquidsoap say it is a problem how the player implemented >the opus codec (VLC). > >If you want, you can listen to my (dutch streams, day hours non stop >music, evenings programms hosted by a announcer) streams, send my a >private message and i will send you al my diffrent strea...

...this for quite a while now and it is stable for all streams. The is one downside on OPUS at the moment, there are not that much clients able to play opus, plus on track change the stream makes a little frame noticeable dropout. I do not know wheather this is cased by liquidsoap or the player. The quys from liquidsoap say it is a problem how the player implemented the opus codec (VLC). If you want, you can listen to my (dutch streams, day hours non stop music, evenings programms hosted by a announcer) streams, send my a private message and i will send you al my diffrent stream links, so you can...

Hi all, What i'm going to do is that enable caller sing while playing a background music (likes karaoke). My approach is using Monitor and Meetme apps.Caller make a call to asterisk, asterisk join caller in to a voice conference and create a dummy caller which will play music, then Monitor app record both music and singer's voice. But i dont know how to create a dummy caller or throw a

hi all, We going to implement a music service which enable user to playback a song by dialing to a service number. The problem is that the amount of data is huge so we have to plae it on an different server which is connected to the asterisk's via internet. Does asterisk support playing a audio file from an resource locate in a remote host? Please help, Thanks, Quyps

Hi all, I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1. It started ok with out cdr_addon_mysql.o. But when I put cdr_addon_mysql.o in to modules folder, it fail at start and the following out has been thrown: ---------- [root at localhost modules]# /usr/sbin/safe_asterisk: line 145: 13270 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTBINDIR}/asterisk -f ${CLIARGS}

...treams. > > The is one downside on OPUS at the moment, there are not that much > clients able to play opus, plus on track change the stream makes a > little frame noticeable dropout. I do not know wheather this is cased by > liquidsoap or the player. > > The quys from liquidsoap say it is a problem how the player implemented > the opus codec (VLC). > > If you want, you can listen to my (dutch streams, day hours non stop > music, evenings programms hosted by a announcer) streams, send my a > private message and i will send you...

...is stable for all streams. > > The is one downside on OPUS at the moment, there are not that much > clients able to play opus, plus on track change the stream makes a > little frame noticeable dropout. I do not know wheather this is cased by > liquidsoap or the player. > > The quys from liquidsoap say it is a problem how the player implemented > the opus codec (VLC). > > If you want, you can listen to my (dutch streams, day hours non stop > music, evenings programms hosted by a announcer) streams, send my a > private message and i will send you al my diffrent s...

I install Wine on Ubuntu 9.04, i have errors when run MS foxpro : namhv at ubuntu:/media/disk/ktgd$ wine foxprox.exe fixme:gdi:get_default_fonts unhandled charset 0x000000a3 - use ANSI_CHARSET for default stock objects err:int:DOSVM_Int15Handler int15: unknown/not implemented parameters: int15: AX 2403, BX e6f5, CX 0c6c, DX 00fc, SI 0cfe, DI 571e, DS 199b, ES f000 err:int:DOSVM_Int2fHandler

Hi all, What i'm going to do is that enable caller sing while playing a background music. My approach is using Monitor and Meetme app. Caller make a call to asterisk, asterisk join caller in to a voice conference and create a dummy caller which will play music. Monitor app record both music and singer's voice. But i dont know how to create a dummy caller or throw a dummy call in order

Hi all, The question has already asked here, http://www.mail-archive.com/asterisk-users at lists.digium.com/msg98176.html but it's been two years since then, so is there any better solution with latest release version? Quyps

>> I think you would be more successful and have more control if you wrote >> it as an AGI. Then you could set a timer that would interrupt the >> process and you could do what you like from there (hangup?). I think >> you are asking too much of the dialplan. > I would tend to leap into an AGI also, but did you try setting an absolute > timeout? >