search for: quy

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...

...erisk 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 Quyps

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

...-endian) data, WAVE audio, > ITU G.711 A-law, mono 8000 Hz > --------- > > How could file command recognize the format as there is no header in the > output file? Or Did I probably miss something making asterisk yield > incorrect alaw files? > > Please help, thanks > > Quyps > > On Thu, 2010-05-20 at 00:50 -0600, Steve Murphy wrote: > > Quyps-- > > > > I've noticed in general that the ulaw, alaw, gsm, slin files used and > > generated by > > asterisk do not have headers (the RIFF stuff), and asterisk is not > > expecting...

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 stre...

...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...

...isk, 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 dummy call in order to do above task. Any idea or comment is appreciated. Thanks Quyps

...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

...Segmentation fault (core dumped) nice -n $PRIORITY ${ASTBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY} Asterisk exited with exit status 139 Asterisk exited on signal 11 Automatically restarting Asterisk. ----------- What is the problem? Thanks in advance. Quyps

...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...

...descriptors from DPMI host err:int31:XMS_Handler int31: unknown/not implemented parameters: int31: AX 0600, BX 10c4, CX 3e14, DX 00bc, SI 5a66, DI 0b5e, DS 199b, ES 199b namhv at ubuntu:/media/disk/ktgd$ dir autobackup.bat DBF DI270309 FOXPROX.EXE PRG VNI B.DBF DEN DI301208 FOXUSER.DBF QUY WSTMP BUTRU DI DICH.FXP FOXUSER.FPT QUY1 BUTRU000.108 DI151208 DICH.PRG GD.BAT saoke CONFIG.FP DI240209 E.IDX LOI saoke_lai CONFIG.FXD DI260309 FOXPRO LUU TEMP please help me to fix this problem.

...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 to do above task. Any idea or comment is appreciated. Thanks Quyps

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

...d tend to leap into an AGI also, but did you try setting an absolute > timeout? > Externivr() may also be a good approach. Hi, I've been searching for some example of ExternIvr(), but didnt find any. Can you point where can i find example of how to use ExternIvr() Thanks in advances Quyps