Anonymous calls through pstn trunk

Hi,
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
To: sip:%2331%230033number@192.168.100.150:5061;tag=cc7a56be7fc1db88i1
From: sip:number@192.168.100.240;tag=as2a87c69d
Call-ID: 4de202c264fef6f75505a769529d61a4@192.168.100.240:5060
CSeq: 103 BYE
Via: SIP/2.0/UDP 192.168.100.240:5060;branch=z9hG4bK596e56fa
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0

You need ISDN to properly control this. The URI encoding is required by the SIP standards. I would hope that it it undone by Asterisk, but you could always use another prefix and translate in the dialplan.

Please use AsterIsk Support, in future.