Asterisk automatically calling to ring-group

Asterisk Asterisk General
1

I have downloaded a script which does automatically call to a extension.

500 = extension
600 = Ring-group

echo “Channel: SIP/500
MaxRetries: 2
RetryTime: 60
WaitTime: 30
Context: alarm
Extension: s
Priority: 1
Set: alarmid=1” > /var/spool/asterisk/outgoing/alarm1.call

If a replace Channel: SIP/500 with SIP/600 it doesn’t ring any of the extensions in the ring group.

How can i call automatically to a ring-group

Thanks!

2

Ring groups are a FreePBX concept, not an Asterisk one.

When you want to start a channel to an extension, rather than a device, you use a “local” channel, in Asterisk. wiki.asterisk.org/wiki/display/ … l+Channels

Having SIP resource names the same as extension numbers is bad security practice as well as encouraging this sort of confusion.

In future, uses Asterisk Support if you want answers in Asterisk terms and freepbx.org/forums/ if you want them in GUI terms.

3

[quote=“david55”]

Having SIP resource names the same as extension numbers is bad security practice as well as encouraging this sort of confusion.[/quote]

if you have sip/username and that later maps to an extension in a dialplan - dont’ you have to write a dial plan for each extension ?

4

You can use a database lookup. In any case, you would only need to add a hint extension, even without a database lookup.

For other technologies, like dahdi, you can’t, sensibly, use the equipment numbers as extension numbers.

5

You could also use a hash of a combination of a secret and the extension number, but you then loose the advantage of being able to do many to many mappings between extensions and devices.

More worryingly, if you weren’t aware of that advice, it means that you haven’t read one of the most important security documents, so you may have other security issues.

6

can you point me to what document you are referring to please

7

README-SERIOUSLY.bestpractices.txt, as included in all recent source distributions, in the top level directory. E.g. svn.digium.com/svn/asterisk/trun … ctices.txt

8

thank you for that

doesn’t seem like that file is included with asterisk now 3.0

9

A question to confirm the best practices if i may

in sip.conf have a non extension as a sip peer

[UserName]
password = somesecure password password

then in extensions.conf

have your standard

exten _[3-5]XX => Dial(Local/${EXTEN}@Context)
instead of exten _[3-5]XX => Dial(SiP/${EXTEN}@Context)

but what you now need is to map the sip device/user/peer to that extension so you will also need:

exten 311 => Dial(SIP/UserName)

i guess this adds the additional mapping of extension to user as you are dialing Local/Extension but it is more secure