I’m using asterisk 11.04 with freepbx 2.11, I have some problems to make anonymous calls through the pstn trunk. With SIP trunks I haven’t any problem, I just put “Anonymous” in the CID section of my outbound route, but the pstn line ignore CID and the only way (I think) is to prepend the #31# (I’m in in france) before the number to call. I tried this way but in the log files I saw that the “#” is replaced by %23. Is this the problem? Is there any solution?
Thank you and sorry for my bad english
<— SIP read from UDP:192.168.100.150:5061 —>
SIP/2.0 200 OK
CSeq: 103 BYE
Via: SIP/2.0/UDP 192.168.100.240:5060;branch=z9hG4bK596e56fa