[asterisk-ss7] wrong mapping from ss7 to sip

[Ercan Yücebas]

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 at 212.23.???.???>;tag=as3200ce93
To: <sip:0041?????????@212.23.???.???>
Contact: <sip:Anonymous at 212.23.???.???>
Call-ID: 27c3fb525cc3bbba0eda4d0a06166265 at 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 at 212.23.???.???>;tag=as3200ce93
To: <sip:0041?????????@212.23.???.???>
Contact: <sip:101 at 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
 
 
 
 
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