search for: dairiten

...y 'i' oops! ;-( exten => i,3,playback(' is incorrect! please again ') # This exten lines are figure for instruction. # I know to use with gsm filename. but ${EXTEN} meaning 'i' that isn't dialed number. Does anyone have good idea? please help --- Masakazu Nakano. Dairiten.com - an open source VoIP and Ubiquitus Portal site in Japan. http://www.dairiten.com:81/modules/news/ powered by xoops at http://www.xoops.org _______________________________________________ Asterisk-Users mailing list mailto:Asterisk-Users at lists.digium.com http://lists.digium.com/mailman/list...

...y 'i' oops! ;-( exten => i,3,playback(' is incorrect! please again ') # This exten lines are figure for instruction. # I know to use with gsm filename. but ${EXTEN} meaning 'i' that isn't dialed number. Does anyone have good idea? please help --- Masakazu Nakano. Dairiten.com - an open source VoIP and Ubiquitus Portal site in Japan. http://www.dairiten.com:81/modules/news/ powered by xoops at http://www.xoops.org

I am running Asterisk CVS-04/16/03-18:57:13, and mpg123-0.59r It all sounds great and it plays at the correct pitch and speed. However at the end of the file it simply does nothing. It does not go on the the next step in the extension.conf nor does it hang up. It just sits there. During play I have two processes running for the mp3 stream: root 6300 6299 8 22:32 ?

hi guys, have any of you guys managed to use msn messenger to authenticate with asterisk using its DNS name? based on my experience with other sip proxies, msn will not authenticate if it sees a different realm than the realm of the client. one workaround i did was to edit the chan_sip.c to send a pre-defined realm, and also edit the Contact: field. after this, asterisk would send a 401 to the

We needed the client browser to be open all the time for dynamic data to load without the page refreshing. After looking at all of our options we decided on programming it ourselves using flash rather than java. We have a flash frontend thats tied to our backend mysql DB. We use it for loading web site traffic data, email opens, click-throughs, bouncebacks, stats, etc. It could also be used with

...p:110 at 192.168.0.1;user=phone>;tag=4dad5671 To: "mack2" <sip:mack2 at 192.168.0.1>;tag=4wvwq4r7lt Call-ID: 3e6096da795e-fc52831snxub at 210.194.204.16 CSeq: 103 ACK User-Agent: Asterisk PBX Content-Length: 0 to 192.168.0.14:5060 --- Masakazu Nakano as mack at irc http://www.dairiten.com:81/

hi. I got these compile errors while install asterisk. readline and openssl are compiled using gnu source, and kernel version is 2.4.17. Compile errors message is follows. Someone cleared this problem? Please, help! Regards. ---------------------------------------------------------------------------------- gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config. o channel.o

ok guys I have a PHP GUI that will be great for both of you. direct editor to the whole file intact OR click to go to an extension. I will post a link to it tomorrow morning... as soon as I can get it off my production server HEHE.... it can do CRC checks on the *.cnf files and it will allow you to edit and parse out for you all your config entries with complex cnf files and default sample

Hi all, A first basic version of DIAX as an ActiveX can be downloaded from: http://www.laser.com/dante/diax/activediax.zip There is only one small file (diax.ocx) and a readme.txt with the usage instructions. For the moment you can only place authenticated (or not) calls and there is no feedback (ring, messages, etc) Put this simple thing on your web page and you will be able to dial from any