search for: lijun820311

Displaying 9 results from an estimated 9 matches for "lijun820311".

2007 Feb 01
1
Re: why there havn't "app_meetme.so"fileaboutasterisk1.4.0?
...ot;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 > > > >"??" <lijun820311@163.com> wrote in message news:45C1B35E.0037E8.32263@m5-81.163.com... >> 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_help...
2007 Feb 01
0
Re: why there havn't"app_meetme.so"fileaboutasterisk1.4.0?
...ou 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 >> >> >> >>"??" <lijun820311@163.com> wrote in message news:45C1B35E.0037E8.32263@m5-81.163.com... >>> 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...
2007 Mar 01
2
How can I use the "GET VARIABLE variablename" in AGI
...de: 200 result=1 (testvariable) AGI Tx >> 520 End of proper usage. ------------------------------------------------------------------------------------------ I couldn't get the global variable ${EXTEN}, who can told me where is the wrong? Thanks a lot, Amy ??? ?????????? ????????lijun820311@163.com ??????????2007-03-02
2007 Feb 01
0
Re: why there havn't "app_meetme.so" fileaboutasterisk1.4.0?
Steven,hello! Thank you so much, but I have installed Zaptel before Asterisk. >You have to compile and install Zaptel first, for asterisk to build meetme. > >-- >-- >Steven > >http://www.glimasoutheast.org > > > >"??" <lijun820311@163.com> wrote in message news:45C1B35E.0037E8.32263@m5-81.163.com... >> asterisk-users@lists.digium.com >> >> hi, >> >> I install asterisk1.4.0 , when I use the meetme application. The console show that >> " WARNING[9872]: pbx.c:1755 pbx_extension_help...
2007 Feb 02
1
WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels ( when I use asyncgoto)
...111-086497c8' jumping out of macro 'nway-start' Feb 2 16:53:10 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge failed on channels SIP/112-08641920 and SIP/111-086497c8 I want to know why there are this warning? How can I fix it? With Regards, Amy ?????????? ????????lijun820311@163.com ??????????2007-02-02
2007 Feb 01
1
why there havn't "app_meetme.so" file about asterisk1.4.0?
asterisk-users@lists.digium.com hi, I install asterisk1.4.0 , when I use the meetme application. The console show that " WARNING[9872]: pbx.c:1755 pbx_extension_helper: No application 'Meetme' for extension " . I found that there havn't "app_meetme.so" in the directory of moudles. Then I complied the asterisk1.4.0 again , there is no
2007 Apr 03
0
I can't use the 'Group', 'CallGroup' , 'PickupGroup' in SIP channel (asterisk1.4.2)
...EN}@300) , this dialplan works fine. exten => _X.,1,Dial(SIP/g1/${EXTEN}) , this dialplan can't work. and there is always show "no such host g1". Who can tell me what is wong ,and give me some hints. Thank you very much. ????????? ?? ????????Amy Li ????????lijun820311@163.com ??????????2007-04-04
2007 Jan 23
12
How to exit from console?
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 -cvvvvg option. Same problem if use asterisk -r to connect. Can not exit. Any
2007 Feb 01
1
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 s@macro-monitor:10 Up Dial(SIP/0882@voip_out Then I send a redirect request like below : Action: Redirect Channel: SIP/612-5456 ExtraChannel: