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...lgo bm -j DROP ?Its something like this Registration from '"30" <sp:30 at my_public_ip:5060> failed for '192.168.xxx.xxx:6373' - Wrong Password? ?and there are approx 10 request per minute of this type. Please suggest some way to stop this.? -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140627/d51df7...

...just first digit and terminates even before considering second digit. Error message : WARNING[5726][C-0000000a]: pbx.c:6696 __ast_pbx_run: Invalid extension '8', but no rule 'i' or 'e' in context 'testmacro' Please suggest what might be wrong. Anurag Rana http://newbie42.blogspot.in/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140908/e8ea5bbb/attachment.html>

Hi, I am using Twinkle to call mobile phone but there is too much noise on the mobile user's end. Mobile user's voice is echoed back to user. While on twinkle end everything is fine. Using Asterisk 11. Please suggest some way to mitigate the problem. Thanks. -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140625/29b489...

Hi All, In asterisk, default directory to store the call-recording files is /var/spool/asterisk/monitor. Can we change this directory? How? -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140626/720599...

Hi All, There is an option of starting the recording of call after the call is bridged. [ b option]. Is there any way of running an AGI script only if call is bridged otherwise not. Thanks -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140626/b87af2...

...ting a call using call files. In 'h' extension I am trying to collect the value of ANSWEREDTIME variable but it is returning null. While It works fine when call is not generated using call files instead is generated from softphone. any idea what might be wrong? thanks Anurag Rana http://newbie42.blogspot.in/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140917/69749bfe/attachment.html>

...ile created in above step. Now I know that 'a' option is used to append the recording to a file but I couldn't find any example on how to use it? Also if I use 'a' option and file doesn't exist then is it created or it is error? Any suggestions please? Anurag Rana http://newbie42.blogspot.in/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140928/d4e3652e/attachment.html>

...n hangup. for example -> [some-context] /// something here which call AGI script no matter what extension receive call. exten => 111,1,Dial(SIP/111) exten => 112,1,Dial(SIP/112) exten => h,1,AGI(pt.py) ;; executes no matter what extension hang up ?Thanks? -- Anurag Rana http://newbie42.blogspot.in/ On the trampoline of life's experiences, Striving towards a saintly life in the midst of these materialistic turbulences. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140628/01b47a...

...enough) but my voice's sound level is very weak. I barely can hear it. During the call receiver is able to hear me. But in recording my part of conversation is barely audible. I am recording using MixMonitor(). Is there anything that can be done to mitigate the problem? Anurag Rana http://newbie42.blogspot.in/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140713/75717b52/attachment.html>

...but no rule 'i' or 'e' in context 'demo3' -- Executing [h at demo3:1] NoOp("SIP/101-0000000d", "terminating call") in new stack [Sep 9 02:11:14] NOTICE[9984]: pbx_spool.c:402 attempt_thread: Call completed to SIP/101/009871888729 Anurag Rana http://newbie42.blogspot.in/ -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20140909/57c7591e/attachment.html>