[asterisk-users] 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, [Feb 8 11:12:16] ERROR[4115]: chan_sip.c:15553 sipsock_read: We could NOT get the channel lock for SIP/700-081da948! [Feb 8 11:12:16] ERROR[4115]: chan_sip.c:15554 sipsock_read: SIP transaction failed: 458b7e8f26d86ee10ec99cba52605...@192.168.1.44 Could you please help me to solve this problem? Thanks. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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] ERROR[14031]: config.c:1083 process_text_line: The file 'manager.d/*.conf' 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. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Help in Perl AGI
Dear All, In Perl AGI, I have two number like 700, 800. I have to call first 700. Next I have to call 800. After that I have to connect this two numbers in the call. How can I do it in Perl AGI? Please anyone provide some idea... Thanks, Velusamy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Fwd: User Authentication in sip.conf
Please any one help for this problem. -- Forwarded message -- From: velusamy velu velu.techni...@gmail.com Date: Mon, Aug 3, 2009 at 10:22 AM Subject: User Authentication in sip.conf To: asterisk-users@lists.digium.com 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 Velusamy sip:7...@192.168.1.222sip%3a...@192.168.1.222 ;tag=yj66acQcycvrN What would be the problem?? Please help me to solve this problem. Best Regards, Velusamy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 Velusamy sip:7...@192.168.1.222sip%3a...@192.168.1.222 ;tag=yj66acQcycvrN What would be the problem?? Please help me to solve this problem. Best Regards, Velusamy ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 WARNING[20199]: chan_oh323.c:3555 cleanup_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. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] 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 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 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
