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Hi! I am proving Asterisk 1.2.24 in realtime with MySQL 5.0.27 using Idefisk softphones. I followed the steps of "how to" of voip-org but always have this error: Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. Sep 25 20:29:07 WARNING[12000]: res_config_mysql.c:360 update_mysql: MySQL RealTime:

...binations are vaild. Vague idea of problem area: Authentication (unencrypted/encrypted mismatch?) Details: Network: WinXP connected to DSL giving IP masquerading to the LAN. Win2000 computer I log on most of the time (named Tyler-Durden). Redhat 8.0 computer (named CheeZe) that's running samba (I don't know how to find the version number). What does work: Daemons are running (not xinetd.conf) smb.conf passes testparm Linux box visable to Win2k machine. What doesn't work: Box comes up asking for user name / passwor...

Hi, I`ve just made a leap from * 1.2.7 to 1.4.19. It took a while to fix all the deprecated stuff, but everything seems to be working fine now, except for a little tiny thing. I lost all color in my CLI, which makes it harder to debug. Is there something that needs doing? I didn't explicitely disable colorization from the command line, and I did try using nocolor=no in the config files.

We're having a problem with Asterisk when we try to pass a call off to a Lucent PSTN using SIP. This behavior does not exist with SER: With Asterisk An ISDN call is started, at the T1 level we receive ?call proceeding? and immediately we receive a ?Call in Progress? just like the far end party has answered. With SER An ISDN call is started, at the T1 level we receive ?call proceeding?

I'm running sippeers and sipusers in my extconfig, and everything runs perfectly when a client is registered (ex. registers to port 1000), but when it re-registers the client is set to port 5060. This behavior does not take place if I use the static files. Both in my sip_buddies table for db, and sip.conf for static I have host=dynamic and nat=yes. :M

I would like to do the following: Dial an extension in Asterisk The extension runs an application which dials a number (like a hybrid of DIAL and READ). The dialed number is a box which does nothing but play DTMF tones then hangs up The first box captures the DTMF tones to a variable Dial exits Dialplan continues with the variable value available. Is this possible? Mik