[Asterisk-Users] Caller-ID is being sent wrong. How to fix it?

[John Fraizer]

OK.  I upgraded to 0.7.2 but and also set a "callerid=" entry in sip.conf. 
The behavior is the same.

Caller-ID is sent as "Name of Calling Party" <number of CALLED party> 
instead of "Name of Calling Party" <number of CALLING party> like it should be.


If you look at the sip debug of a call between extenstion 2288888 and 
extension 100, you can see what is causing the problem:

     -- Executing Dial("SIP/2288888-76d1", "SIP/100|20") in new stack
We're at 66.35.64.38 port 13694
Answering with preferred capability 4
Answering with preferred capability 8
Answering with preferred capability 256
Answering with non-codec capability 1
12 headers, 11 lines
Reliably Transmitting:
INVITE sip:100 at 24.33.239.118 SIP/2.0
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK025b4e85
From: "John Fraizer" <sip:100 at 66.35.64.38>;tag=as2e305230
To: <sip:100 at 24.33.239.118>
Contact: <sip:100 at 66.35.64.38>
Call-ID: 5f40370b06526ca4094b543d1808816a at 66.35.64.38
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Sat, 07 Feb 2004 19:07:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 10168 10168 IN IP4 66.35.64.38
s=session
c=IN IP4 66.35.64.38
t=0 0
m=audio 13694 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
  (NAT) to 24.33.239.118:5060
     -- Called 100




It is this part that is causing it to get the wrong caller ID number:

INVITE sip:100 at 24.33.239.118 SIP/2.0
Via: SIP/2.0/UDP 66.35.64.38:5060;branch=z9hG4bK025b4e85
From: "John Fraizer" <sip:100 at 66.35.64.38>;tag=as2e305230
To: <sip:100 at 24.33.239.118>


Notice that we're inviting 100@ while claiming that the call is COMING from 
100@?  It puts the right caller ID *name* in the invite but, the "sip:100" 
in the from field is flat out wrong.  It should be "sip:2288888".

Surely I am not the only one to notice that this is broken.


John