search for: velusamy

...encrypt passwords = yes winbind separator = + idmap backend = lwopen idmap uid = 10000-20000 idmap gid = 10000-20000 winbind enum users = yes winbind enum groups = yes template homedir = /dev/null template shell = /bin/true [adshare] path = /home/velusamy/Pictures/ writable = yes valid users = ASSURANCE+velu browseable = yes Now, executed the smb-clinet. smbclient //192.168.5.136/adshare -U velu It asked password, given, it connected to the share. But, I was unable to access the share form different machine which is connected...

...My configurations as follows, [2000] type=peer host=dynamic insecure=port,invite ; (both) context=Testing But when I call '2000', I noticed the following message in Asterisk console, "NOTICE[5702]: chan_sip.c:10543 handle_request_invite: Failed to authenticate user "Velusamy" <sip:727 at 192.168.1.222<sip%3A727 at 192.168.1.222> >;tag=yj66acQcycvrN" What would be the problem?? Please help me to solve this problem..... Best Regards, Velusamy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/p...

Dear all, How do I set the UUI informations for outgoing calls and read the UUI information for incoming call in asterisk? Thanks in advance.. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100531/ffcceeee/attachment.htm

...have originated the call to one extension, once answered I have called DeadAGI to control the call. I have problem that after hangup the call AGI "GET FULL VARIABLE" returns -1 for ANSWEREDTIME channel variable. What is the problem? Where I made wrong. Please suggest me.. Regards, Velusamy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100708/4855358f/attachment.htm

Dear all, In Sip.conf file how to setup incoming calls not to use authentication? Please provide some steps to do it.. Thanks... Regards, Velusamy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090801/d6c39afc/attachment.htm

...ing message in console. " WARNING[8446]: chan_oh323.c:3555 cleanup_h323_connection: Call 'ip$192.168.8.96:30005/27890-f5194af7' not found (clear). " Please any one give suggestions to disable the gatekeeper access in Asterisk... Thanks in Advance... Regards, Velusamy.K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090714/80273471/attachment.htm

...stalled Asterisk 1.6.1.9 to use Bridge Application in AMI. After inatallation I have tried to connect the AMI via telnet. But it didn't connected. I used netstat to know the listening socket. But it was not available. How to start the AMI server socket. Please any one help me....... Thanks, Velusamy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091109/fe45e2fa/attachment.htm

...nup_h323_connection: Call 'ip$192.168.8.96:30005/32113-b632393e' not found (clear)." I am using the Asterisk 1.2.13 version. What would be the problem? Please any one give suggestions to execute the gatekeeper in routing mode... Thanks in Advance, Regards, Velusamy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090721/33009f71/attachment.htm

...was listed as a #include but it does not exist. [Nov 7 13:13:26] NOTICE[14031]: manager.c:4081 __init_manager: Unable to open AMI configuration manager.conf. Asterisk management interface (AMI) disabled. What is the problem? How can I over come this problem? Please any one help me..... Thanks, Velusamy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091107/e36010f6/attachment.htm

Dear All, I am using asterisk 1.4.21.2. I have used Originate manager application to to call the two persons. I have called AGI application to call another person. There I have used GET FULL VARIABLE AGI command to get the value. I am able to call another person form AGI. But when one end cut the call another one didn't disconnected. The following errors are displayed in Asterisk console,