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What are the working settings for Asterisk


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Due to a larger need for configuration of Asterisk PBX's, we've worked out a set of default settings that work for several users. Below you will find snippets of the configuration files that have the various settings you will need.

This is a known working config, all you have to do is insert the right data where we have all caps words placed.

SIP.CONF

[general]
register => YOURNUMBER:YOURPASS@INSERT PROXY HERE/YOURNUMBER; registration of your pbx

The number after the slash (/) is required, otherwise you won't receive calls from us because you tell our server that you want to answer calls for "s".
[main]
context=global ;context to use
type=peer ; user type
fromuser=YOUR NUMBER ; callerid number
username=YOUR NUMBER
authuser=YOUR NUMBER
secret=YOUR PASSWORD
host= INSERT PROXY HERE ;host to contact
fromdomain= INSERT PROXY HERE ; domain you're on
nat=no ; network address translation (set to yes if you have problems)
canreinvite=yes
insecure=very ; security level
qualify=yes ; require password (we do require this)
dtmfmode=inband (needed on BOTH ends to allow keytone passing)
dtmf=inband ; same as previous item

EXTENSIONS.CONF

[outgoing]
exten => _1NXXNXXXXXX,1,dial(SIP/${EXTEN:0}@main,60,Tt)
exten => _1NXXNXXXXXX,2,congestion
exten => _NXXNXXXXXX,1,dial,(SIP/1${EXTEN:0}@main,60,Tt)
exten => _NXXNXXXXXX,2,Hangup

[incoming]
The incoming section MUST include as it's start YOUR NUMBER as the pattern match. Example for number 15181231234
exten =>_15181231234,1,START PROCESSING CALL FROM TRUNK
OR
exten =>_1NXXNXXXXXX,1,START PROCESSING ANY VALID NUMBER

If you do not have this pattern in your incoming, you will not receive calls, because you have not instructed your system to do so.

Additionally, be aware that your service codes may conflict with our service codes. One primary example is *123, which is the service code to grab voicemail from the server. If you wish to use voicemail internally as well as through our service, you must use a code different than *123, or setup your outgoing plan to intercept the number you wish to use, but dial outbound as *123.

Example to use *86 (*VM) as your external voicemail config. Put the following in your outgoing section.

exten => *86,1,Dial(SIP/*123$@main,120,t)

*4378 (*HELP) is our quick support dial, please make sure you're not using this code if you wish the feature to work.

Also, if you want per extension callback last caller, you'll want to define it in your extensions.conf. Otherwise, the last number to call your office will be used.

The extensions.conf file edit does not need to be in there, but is posted to allow understanding of a working extension setup for a "trunk" (outgoing/incoming line)

Thank you,
The Engineering Dept





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