3 Oct
2008
3 Oct
'08
11:13 p.m.
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