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How to create extensions in Asterisk-PBX?
A SIP extension is configured in the SIP channel driver configuration file, called sip.conf. (This file resides in the Asterisk configuration directory, which is typically /etc/asterisk.)
Open sip.conf with your favorite text editor, scroll to the bottom of the file, and add a section for your extension. You’ll need to choose your own unique password for each account, and change the permit line to match the settings for your local network.
For example, to configure an extension 1500, following lines needs to be added to sip.conf:
[1500]
deny=0.0.0.0/0.0.0.0
type=friend
secret=****
qualify=yes
port=5060
pickupgroup=
permit=0.0.0.0/0.0.0.0
nat=yes
mailbox=1500@device
host=dynamic
dtmfmode=rfc2833
dial=SIP/1500
context=from-internal
canreinvite=no
callgroup=
callerid=device
accountcode=
call-limit=50
Make sure to restart Asterisk in order for these changes to take effect.
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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
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.
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.
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