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Asterisk time based routing.

This is a very common requirement that route the calls to Voice-mail after office hours. Or you can transfer the calls to your cell phone after certain time say 6:00pm. In Asterisk you can also control the call location based on time and date.

exten => start,1,Verbose(2,Entering our auto-attedant)
same => n,Answer()
same => n,Playback(silence/1)
; We’re closed on New Years Eve, New Years Day, Christmas Eve, and Christmas Day
same => n,GotoIfTime(*,*,31,dec?holiday,1)
same => n,GotoIfTime(*,*,1,jan?holiday,1)
same => n,GotoIfTime(*,*,24,dec?holiday,1)
same => n,GotoIfTime(*,*,25,dec?holiday,1);
We can set auto-attendant dialplan to check for national holidays, if that is true,jump to “exten => holiday,1,Verbose(2,We’re closed for a holiday.)”
; Our operational hours are Monday-Friday, 9:00am to 5:00pm.
same => n,GotoIfTime(0900-1700,mon-fri,*,*?open,1:closed,1) ; It true jump to exten open, else jump to exten closed.
exten => open,1,Verbose(2,We’re open!)
same => n,Background(custom/open-greeting)
exten => closed,1,Verbose(2,We’re closed.)
same => n,Playback(custom/closed-greeting)
same => n,Voicemail(general-mailbox@default,u)
same => n,Hangup()
exten => holiday,1,Verbose(2,We’re closed for a holiday.)
same => n,Playback(custom/closed-holiday)
same => n,Voicemail(general-mailbox@default,u)
same => n,Hangup()
GotoIfTime() application is not only good for an auto-attendant but it can be used anywhere in the dialplan. For example sometimes we just want to forward calls to different people based on time, such as when support staff is not in the office on weekends, but are on call:
exten => 100,1,Verbose(2,Calling Support Staff.)
same => n,GotoIfTime(*,sat&sun,*,*?on_call,1)
same => n,Dial(SIP/support,30)
same => n,Voicemail(support@default,u)
same => n,Hangup()
exten => on_call,1,Verbose(2,Calling On-Call Support Staff.)
same => n,Dial(SIP/didforsale/8888888888&SIP/didforsale/8555555555,30)
same => n,Voicemail(support@default,u)
same => n,Hangup()

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Asterisk one way audio issue

Are you having an audio issues in your Asterisk? 
Well it’s a common issue with PBX to have audio issues like one way audio or no audio. Sometimes only caller can hear remote party or remote party only can hear the caller. You must be wondering what causes this issue? This problem in audio is mainly because of the NAT issues. We recommend to use NAT with enabling 10000-20000 UDP ports on firewalls and also to enable natting on trunks

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How to limit the number of calls in asterisk

Learn how you can limit number of simultaneous calls on Asterisk based SIP Trunk. If you want to limit the number calls for your SIP peer or friend in Asterisk use call-limit in your trunk configuration.

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