The call comes in with restricted cli from ss7 With asterisk 1.2.4, the cli/ani becomes simply ****** I changed this behaviour already I wanna forward all my incoming ss7 calls to sip server and I see now onw issue there - the ?don?t present the cli? information will be mapped wrong on sip invite ss7 debug shows the ani (because I changed the original source code) and this invite is sended out INVITE sip:0041?????@212.23.???.??? SIP/2.0 Via: SIP/2.0/UDP 212.23.???.???:5060;branch=z9hG4bK1c9b1aaf;rport From: "Anonymous" <sip:Anonymous@212.23.???.???>;tag=as3200ce93 To: <sip:0041?????????@212.23.???.???> Contact: <sip:Anonymous@212.23.???.???> Call-ID: 27c3fb525cc3bbba0eda4d0a06166265@212.23.???.??? CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Sun, 11 Mar 2007 22:31:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Type: application/sdp Content-Length: 367 v=0 o=root 16096 16096 IN IP4 212.23.???.??? s=session c=IN IP4 212.23.???.??? t=0 0 m=audio 12616 RTP/AVP 8 0 111 3 4 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:111 G726-32/8000 a=rtpmap:3 GSM/8000 a=rtpmap:4 G723/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - but this invite is wrong, it should be like this from and contact fields needs to be corrected INVITE sip:0041?????@212.23.???.??? SIP/2.0 Via: SIP/2.0/UDP 212.23.???.???:5060;branch=z9hG4bK1c9b1aaf;rport From: "anonymous" <sip:101@212.23.???.???>;tag=as3200ce93 To: <sip:0041?????????@212.23.???.???> Contact: <sip:101@212.23.???.???> If the incoming call is with restricted cli, then the sip signaling should be like above, the code can not just simple add everywhere UNKNOWN?s, no ! This is wrong, the code has to forward the original cli in from and contact fields in this case, but need changed the name in the from field with anonymous and not unknown or something like that More important is the cli forwarding to sip, because I need to bill my pstn customer. Currently because this anonymous cli issue, my sip server can not identify the customer, the display name can stay as anonymous, but the sip username should be identical with ani/cli. DOES ANYBODY KNOW, WHERE IN CODE THIS CAN BE FIXED Thanks Br ercan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20070311/1cfaefe4/attachment.htm
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