When you got a new VoIP phone or PBX up and running, you may want to test that you can do IP dialing.

For this purpose we got two SIP URIs that you can use to test:

sip:announce@code.osso.nl
sip:echo@code.osso.nl

They're reachable over both TCP and UDP, they allow alaw and ulaw audio. The first one plays some default message included in the Asterisk sounds package, the second one echoes any audio it gets.

They use SRV records, which I'm sure you have enabled, and are running on the default port ...

In the Asterisk PBX dialplan, expressions can be formed using the $[...] syntax. Addition, subtraction, comparison and so on are defined. As is a regex operator: =~

Unfortunately, the documentation about the details of the implementation is hard to find.

Here, a breakdown of my findings:

• static struct val * op_eqtilde is defined in main/ast_expr2.y
• It uses the REG_EXTENDED flag when calling regcomp: so extended regular expression syntax is used.
• The REG_ICASE flag is NOT used: case sensitive.
• If you don't use parentheses (substring addressing) the returned value is the amount of characters that match: rm[0].rm_eo - rm[0 ...