[asterisk-users] How can I use the GET VARIABLE variablename in AGI
Hi,All,
I wang to use AGI in asterisk1.4.
AGI file / myperl.agi
#!/usr/bin/perl
use strict;
..
print STDERR 7. Testing GET VARIABLE...;
print GET VARIABLE EXTEN \\\n;
my $result = STDIN;
checkresult($result);
..
when the agi execute; asterisk conosle show that :
AGI Rx GET VARIABLE EXTEN
AGI Tx 520-Invalid command syntax. Proper usage follows:
AGI Tx Usage: GET VARIABLE variablename
Returns 0 if variablename is not set. Returns 1 if variablename
is set and returns the variable in parentheses.
example return code: 200 result=1 (testvariable)
AGI Tx 520 End of proper usage.
--
I couldn't get the global variable ${EXTEN}, who can told me where is the wrong?
Thanks a lot,
Amy
李君
[EMAIL PROTECTED]
2007-03-02
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels ( when I use asyncgoto)
Hi All, I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ; when I dial ,there have this warning: -- Executing AsyncGoto(SIP/111-086497c8, SIP/113-08674628|dynamic-nway|111|1) in new stack Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting async goto (SIP/113-08674628) to dynamic-nway,111,1 Feb 2 16:53:10 DEBUG[4218]: channel.c:2834 ast_channel_masquerade: Planning to masquerade channel SIP/113-08674628 into the structure of AsyncGoto/SIP/113-08674628 Feb 2 16:53:10 DEBUG[4218]: channel.c:2847 ast_channel_masquerade: Done planning to masquerade channel SIP/113-08674628 into the structure of AsyncGoto/SIP/113-08674628 Feb 2 16:53:10 DEBUG[4218]: channel.c:2972 ast_do_masquerade: Got clone lock for masquerade on 'SIP/113-08674628' at 0x8677314 Feb 2 16:53:10 DEBUG[4218]: channel.c:3154 ast_do_masquerade: Putting channel SIP/113-08674628 in 2/2 formats Feb 2 16:53:10 DEBUG[4218]: channel.c:3189 ast_do_masquerade: Released clone lock on 'AsyncGoto/SIP/113-08674628ZOMBIE' Feb 2 16:53:10 DEBUG[4218]: channel.c:3198 ast_do_masquerade: Done Masquerading SIP/113-08674628 (6) Feb 2 16:53:10 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels SIP/111-086497c8 and AsyncGoto/SIP/113-08674628ZOMBIE Feb 2 16:53:10 DEBUG[4218]: app_dial.c:1636 dial_exec_full: Exiting with DIALSTATUS=ANSWER. -- Executing Set(SIP/111-086497c8, DYNAMIC_FEATURES=) in new stack -- Executing Goto(SIP/111-086497c8, dynamic-nway|111|1) in new stack -- Goto (dynamic-nway,111,1) == Channel 'SIP/111-086497c8' jumping out of macro 'nway-start' Feb 2 16:53:10 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels SIP/112-08641920 and SIP/111-086497c8 I want to know why there are this warning? How can I fix it? With Regards, Amy 李君 [EMAIL PROTECTED] 2007-02-02 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk cann't redirect the calling party to anothere Exten.
Hi All, I use the Asterisk Manager Interface to redirect the channels. There have two channels : SIP/voip_out_22-809c (None) Up Bridged Call(SIP/612-5456) SIP/612-5456 [EMAIL PROTECTED]:10 Up Dial(SIP/[EMAIL PROTECTED] Then I send a redirect request like below : Action: Redirect Channel: SIP/612-5456 ExtraChannel: SIP/voip_out_22-809c Exten: 111 Context: meetme-test Priority: 1 Then , the channel named SIP/voip_out_22-809c has been transfered to the conference 111. But, the channel named SIP/612-5456 has been hangup automatic. The context meetme-test is : [meetme-test] exten = 111,1,Answer exten = 111,n,MeetMe(111,pdMX) exten = 111,n,Hangup I want to redirect both channels to the conference 111. What's wrong it? With Regards, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels ( when I use asyncgoto)
Hi All, I download the app_asyncgoto.c, compile the app_asyncgoto.so. Then according to this page http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ; when I dial ,there have this warning: -- Executing AsyncGoto(SIP/111-086497c8, SIP/113-08674628|dynamic-nway|111|1) in new stack Feb 2 16:53:10 DEBUG[4218]: app_asyncgoto.c:95 asyncgoto_exec: Attempting async goto (SIP/113-08674628) to dynamic-nway,111,1 Feb 2 16:53:10 DEBUG[4218]: channel.c:2834 ast_channel_masquerade: Planning to masquerade channel SIP/113-08674628 into the structure of AsyncGoto/SIP/113-08674628 Feb 2 16:53:10 DEBUG[4218]: channel.c:2847 ast_channel_masquerade: Done planning to masquerade channel SIP/113-08674628 into the structure of AsyncGoto/SIP/113-08674628 Feb 2 16:53:10 DEBUG[4218]: channel.c:2972 ast_do_masquerade: Got clone lock for masquerade on 'SIP/113-08674628' at 0x8677314 Feb 2 16:53:10 DEBUG[4218]: channel.c:3154 ast_do_masquerade: Putting channel SIP/113-08674628 in 2/2 formats Feb 2 16:53:10 DEBUG[4218]: channel.c:3189 ast_do_masquerade: Released clone lock on 'AsyncGoto/SIP/113-08674628ZOMBIE' Feb 2 16:53:10 DEBUG[4218]: channel.c:3198 ast_do_masquerade: Done Masquerading SIP/113-08674628 (6) Feb 2 16:53:10 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels SIP/111-086497c8 and AsyncGoto/SIP/113-08674628ZOMBIE Feb 2 16:53:10 DEBUG[4218]: app_dial.c:1636 dial_exec_full: Exiting with DIALSTATUS=ANSWER. -- Executing Set(SIP/111-086497c8, DYNAMIC_FEATURES=) in new stack -- Executing Goto(SIP/111-086497c8, dynamic-nway|111|1) in new stack -- Goto (dynamic-nway,111,1) == Channel 'SIP/111-086497c8' jumping out of macro 'nway-start' Feb 2 16:53:10 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels SIP/112-08641920 and SIP/111-086497c8 I want to know why there are this warning? How can I fix it? With Regards, Amy 李君 [EMAIL PROTECTED] 2007-02-02 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] why there havn't app_meetme.so file about asterisk1.4.0?
asterisk-users@lists.digium.com hi, I install asterisk1.4.0 , when I use the meetme application. The console show that WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for extension . I found that there havn't app_meetme.so in the directory of moudles. Then I complied the asterisk1.4.0 again , there is no app_meetme.so also. How to overcome this problem? Thanks, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: why there havn't app_meetme.so fileaboutasterisk1.4.0?
Steven,hello! Thank you so much, but I have installed Zaptel before Asterisk. You have to compile and install Zaptel first, for asterisk to build meetme. -- -- Steven http://www.glimasoutheast.org 李君 [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] asterisk-users@lists.digium.com hi, I install asterisk1.4.0 , when I use the meetme application. The console show that WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for extension . I found that there havn't app_meetme.so in the directory of moudles. Then I complied the asterisk1.4.0 again , there is no app_meetme.so also. How to overcome this problem? Thanks, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = = = = = = = = = = = = = = = = = = = = 致 礼！ 李君 [EMAIL PROTECTED] 2007-02-01 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: RE: [asterisk-users] Re: why there havn'tapp_meetme.sofileaboutasterisk1.4.0?
Bill Gibbs,hello Thank you so much. According to this method , I get the app_meetme.so . === 2007-02-01 22:49:43 您在来信中写道：=== Removed the DEPENDS_app_meetme=ZAPTEL from the file menuselect.makedeps And in menuselect.makeopts I removed the DEPSFAILED line that had meetme in it. It then compiled. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of ?? Sent: Thursday, February 01, 2007 9:01 AM To: Asterisk Users Mailing List - No Subject: Re: [asterisk-users] Re: why there havn't app_meetme.sofileaboutasterisk1.4.0? Steven,hello! Thank you so much, but I have installed Zaptel before Asterisk. You have to compile and install Zaptel first, for asterisk to build meetme. -- -- Steven http://www.glimasoutheast.org 李君 [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] asterisk-users@lists.digium.com hi, I install asterisk1.4.0 , when I use the meetme application. The console show that WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for extension . I found that there havn't app_meetme.so in the directory of moudles. Then I complied the asterisk1.4.0 again , there is no app_meetme.so also. How to overcome this problem? Thanks, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = = = = = = = = = = = = = = = = = = = = 致 礼！ 李君 [EMAIL PROTECTED] 2007-02-01 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = = = = = = = = = = = = = = = = = = = = 致 礼！ 李君 [EMAIL PROTECTED] 2007-02-02 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk cann't redirect the calling party to anothere Exten.
Hi All, I use the Asterisk Manager Interface to redirect the channels. There have two channels : SIP/voip_out_22-809c (None) Up Bridged Call(SIP/612-5456) SIP/612-5456 [EMAIL PROTECTED]:10 Up Dial(SIP/[EMAIL PROTECTED] Then I send a redirect request like below : Action: Redirect Channel: SIP/612-5456 ExtraChannel: SIP/voip_out_22-809c Exten: 111 Context: meetme-test Priority: 1 Then , the channel named SIP/voip_out_22-809c has been transfered to the conference 111. But, the channel named SIP/612-5456 has been hangup automatic. The context meetme-test is : [meetme-test] exten = 111,1,Answer exten = 111,n,MeetMe(111,pdMX) exten = 111,n,Hangup I want to redirect both channels to the conference 111. What's wrong it? With Regards, Amy ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re: [asterisk-users] How to exit from console?
Derek Whitten,hello！ *CLI stop now You can input help to see all the commands. Like this: *CLI help === 2007-01-23 23:10:12 === Rudolf Ladyzhenskii wrote: Hi, all Stupid question, but how do you exit asterisk console without stopping the asterisk? Tried quit and exit: *CLI exit No such command 'exit' (type 'help' for help) *CLI quit No such command 'quit' (type 'help' for help) *CLI Any other ideas? I started asterisk with -cg option. Same problem if use asterisk -r to connect. Can not exit. Any ideas? Thanks, Rudolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ctrl-c :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users = = = = = = = = = = = = = = = = = = = = 致 礼！ 李君 [EMAIL PROTECTED] 2007-01-24 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
