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How to create a multitenant users in Asterisk-pbx?

Multi-tenancy is an architecture in which a single instance of a software application serves multiple customers. Each customer is called a tenant.
For creating multitenant we need to create custom extensions in /etc/asterisk/extensions_custom.conf and give relevant context route calls:


exten => 1234512345,1,Set(__FROM_DID=${EXTEN})
exten => 1234512345,n,Gosub(app-blacklist-check,s,1)
exten => 1234512345,n,ExecIf($[ “${CALLERID(name)}” = “” ] ?Set(CALLERID(name)=${CALLERID(num)}))
exten => 1234512345,n,Set(__CALLINGPRES_SV=${CALLERPRES()})
exten => 9498851902,n,Set(CALLERPRES()=allowed_not_screened)
exten => s,1,Dial(SIP/1500)


exten => 1234567890,1,Set(__FROM_DID=${EXTEN})
exten => 1234567890,n,Gosub(app-blacklist-check,s,1)
exten => 1234567890,n,ExecIf($[ “${CALLERID(name)}” = “” ] ?Set(CALLERID(name)=${CALLERID(num)}))
exten => 1234567890,n,Set(__CALLINGPRES_SV=${CALLERPRES()})
exten => 1234567890,n,Set(CALLERPRES()=allowed_not_screened)
exten => s,1,Dial(SIP/1701)

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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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