Re: [asterisk-users] Confrence call is not make
Hi This is due to module app_meetme.so is not loaded. Execute below command in asterisk cli and check the cli logger. > module load app_meetme.so If you are installed asterisk in a linux system without any analog interface this meetme application will not work. You have use application "Conference" instead of MeetMe. Thanks Nikhil On 12/08/2011 11:12 AM, Durgesh Mishra wrote: Hi, I am making confrence application. In sip.conf [phone1] type=friend host=dynamic Takes an alphanumeric string. context= employees [phone2] type=friend host=dynamic context= employees [phone3] type=friend host=dynamic context= employees In extension.conf [employees] exten => 101,1,Dial(SIP/phone1,20,tT) exten => 102,1,Dial(SIP/phone2,20,tT) exten => 103,1,Dial(SIP/phone3,20,tT) exten => 777,1,MeetMe(777) In meetme.conf [rooms] conf => 777 when i call 777 from phone1 ,its shows 603 declined. I check in CLI [Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No application 'MeetMe' for extension (employees, 777, 1) == Spawn extension (employees, 777, 1) exited non-zero on 'SIP/phone1-' Plz tell me , where i am wrong in configuration. Thanks -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confrence call is not make
I check in CLI [Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No application 'MeetMe' for extension (employees, 777, 1) == Spawn extension (employees, 777, 1) exited non-zero on 'SIP/phone1-' Plz tell me , where i am wrong in configuration. Chances are you didn't install DAHDI before building Asterisk, so you never built the MeetMe application. Go download & install DAHDI, then rerun ./configure & menuselect for Asterisk and make sure the MeetMe app is available. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Confrence call is not make
On Thu, 8 Dec 2011, Durgesh Mishra wrote: I am making confrence application. [Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No application 'MeetMe' for extension (employees, 777, 1) You don't have app_meetme.so loaded. What happens if you enter 'module load app_meetme.so' at Asterisk's CLI? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Confrence call is not make
Hi, I am making confrence application. In sip.conf [phone1] type=friend host=dynamic Takes an alphanumeric string. context= employees [phone2] type=friend host=dynamic context= employees [phone3] type=friend host=dynamic context= employees In extension.conf [employees] exten => 101,1,Dial(SIP/phone1,20,tT) exten => 102,1,Dial(SIP/phone2,20,tT) exten => 103,1,Dial(SIP/phone3,20,tT) exten => 777,1,MeetMe(777) In meetme.conf [rooms] conf => 777 when i call 777 from phone1 ,its shows 603 declined. I check in CLI [Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No application 'MeetMe' for extension (employees, 777, 1) == Spawn extension (employees, 777, 1) exited non-zero on 'SIP/phone1-' Plz tell me , where i am wrong in configuration. Thanks -- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
