Today I had a strange behaviour:
I am routing calls to an Asterisk PBX. It has a very sophisticated
dynamic least cost router built in - so I use it to terminate mobile and
international calls.

For a first test I created a dialplan which checks for ^231$ in the
dialled number and then routes the call to $1/asterisk.pbx.
It didn't work. Grepping the SIP traffic I could see that Freeswitch
makes an invite $1@asterisk.pbx. Yes it's $1 in the SIP message.

Second I checked for the mobile number using ^(01[6-7]\d{4,13})$ in the
dialled number and then route the call to $1/asterisk.pbx.
This works.

I am confused as in the second case $1 is translated to the dialled
number and in the frist case not.

Anybody has a clue?

Best regards
Peter