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...I execute this sencente from a Server X to a Server Y /usr/local/bin/rsync -av --include=/usr/local/mysql/bin/listado_images.txt 192.168.1.23::mysql_prueba I'd like to rsync only files contained in listado_images.txt but I don't get it. Text of file listado_images.txt is /datos/web/neweb/images/noticia41309_fotoLP.jpg /datos/web/neweb/images/noticia41309_foto1N.jpg I have attempt with + /datos/web/neweb/images/noticia41309_fotoLP.jpg + /datos/web/neweb/images/noticia41309_foto1N.jpg But I don't get it. Where is the problem ??

Hi all Now I'm making IVR sequance that is customised [mainmanu]. I wish to notify invaid command like a following exten => i,1,playback('your command is ...') exten => i,2,playback(${EXTEN}) ; <---- Say 'i' oops! ;-( exten => i,3,playback(' is incorrect! please again ') # This exten lines are figure for instruction. # I know to use with gsm filename.

Hi all. I found a connection error from sipphone.com. It seems 'realm based authentication' by sipphone.com. any ideas? Regards. mack

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 ?

I have a test SIP account set up with WorldCom and I have been trying to have Asterisk register to the WorldCom server with no luck. It appears that the SIP headers are different coming from Asterisk. I have included a packet capture from a successful login with a Windows Messenger client for reference. I have also copied in the SIP packet I captured with sip debug turned on. In my sip.conf file,

Hi Isamar maybe I think disclose your code to CVS is best and fast :-) mack > > Hi folks, > > I'm trying now to play with fsk_modem.c and callerid.c > to get the Japanese callerid working and I already got to make some > steps.. > I don't know if anybody accomplished that already... but > Since two or more minds think better than one, send private messages >

Hi all I was tryed to connect to mysip.ch scs_client by siemens that isn't works well. Does anyones knows to work H/W or S/W applictations in asterisk SIP videosupport? Regards mack_jpn

sorry, this problem is fixed by myself. we must need set 'canreinvite=no' each user. --- I'm try to discconect a call with SIP. when caller make a call, 'show channels' result is following. mack*CLI> show channels Channel (Context Extension Pri ) State Appl. Data SIP/mack-1bfc (default 1 ) Ringing AppDial (Outgoing