search for: velusami

Displaying 10 results from an estimated 10 matches for "velusami".

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2012 Jun 29
2
Samba with Active directory integration problem
Hi, I have followed the all the steps given, in https://help.ubuntu.com/community/ActiveDirectoryWinbindHowto. to integrate the samba with active directory. I have the following configuration file, [global] workgroup = ASSURANCE security = ads realm = ASSURANCE.LOCAL encrypt passwords = yes winbind separator = + idmap backend = lwopen idmap uid =
2009 Aug 03
1
User Authentication in sip.conf
Dear all, I want to setup the incoming calls, that don't use authentication in sip.conf file. 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
2010 May 31
3
Read and set the UUI in asterisk
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
2010 Jul 08
1
AGI get full variable
Dear All, I have "get full variable" AGI call to get the ANSWEREDTIME channel variable. I 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..
2009 Aug 01
1
how to setup incoming calls not to use authentication
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
2009 Jul 14
3
Help in oh323 Gatekeeper
Dear All, I have installed GNU gatekeeper in my machine. I tested the calls using gatekeeper successfully. Now I have tried to Disable the gatekeeper in oh323.conf file gatekeeper=DISABLE Now I have tried to call, but the connection is not established. I have got following warning message in console. " WARNING[8446]: chan_oh323.c:3555
2009 Nov 09
1
How to know AMI status
Dear All, I have installed 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
2009 Jul 21
0
Gatekeeper Routing Mode not Working
Dear All, I have enabled the gatekeeper in oh323.conf file. I started the gatekeeper and also restarted the Asterisk. When I called, it was worked fine. After then enabled the routing mode in gatekeeper.ini file then I restarted the gatekeeper. When I called the routing mode didn't work. I have received the following warning in Asterisk console. " Jul 21 14:19:11
2009 Nov 07
0
AMI is not loaded
Dear All, I have the following entry in the /etc/asterisk/manager.conf file, [general] enabled = yes webenabled = yes port = 5038 bindaddr = 0.0.0.0 [admin] secret = admin read = system,call,log,verbose,command,agent,user,config write = system,call,log,verbose,command,agent,user,config When I did 'reload manager' in CLI I have received following error, [Nov 7 13:13:26]
2010 Feb 08
0
Call doesn't disconnect in SIP
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,