[asterisk-users] SIP trunk call initiated as Anonymous at anonymous.invalid

[Gordon Messmer]

I have a Grandstream HT-502 device connected to my Asterisk PBX.  It is 
configured not to place anonymous calls, and from my mostly layman 
reading of the invitation that the device sends, it should not be 
anonymous.  However, the Asterisk PBX sends an anonymous invitation to 
our SIP trunk provider.  Can anyone explain why?

The two INVITE packets follow.

The devices sends the following INVITE:

INVITE sip:2223334444 at pbx.xxxxx.com SIP/2.0
Via: SIP/2.0/UDP 192.168.9.197:46538;branch=z9hG4bK526774101;rport
From: "222333555" <sip:222333555 at pbx.xxxxx.com>;tag=2072922124
To: <sip:2223334444 at pbx.xxxxx.com>
Call-ID: 1082640776-46538-3 at BJC.BGI.J.BJH
CSeq: 21 INVITE
Contact: "222333555" <sip:222333555 at 192.168.9.197:46538>
Authorization: Digest username="222333555", realm="asterisk", 
nonce="02774xxx", uri="sip:2223334444 at pbx.xxxxx.com", 
response="0d1b93729332670aae5b6916ecfxxxxx", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT-502  V1.2A 1.0.5.10
Privacy: none
P-Asserted-Identity: "222333555" <sip:222333555 at pbx.xxxxx.com>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, 
REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   400

v=0
o=222333555 8000 8000 IN IP4 192.168.9.197
s=SIP Call
c=IN IP4 192.168.9.197
t=0 0
m=audio 58270 RTP/AVP 0 8 4 18 112 97 102 100
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:102 G729E/8000
a=rtpmap:100 AAL2-G726-16/8000

Our PBX sends this INVITE to our SIP trunk provider:

INVITE sip:2223334444 at 10.250.0.5 SIP/2.0
Via: SIP/2.0/UDP 66.77.88.99:5060;branch=z9hG4bK1b55d480;rport
Max-Forwards: 70
From: "Anonymous" <sip:Anonymous at anonymous.invalid>;tag=as567ac377
To: <sip:2223334444 at 10.250.0.5>
Contact: <sip:Anonymous at 66.77.88.99:5060>
Call-ID: 08be883c133cae41515d1f914d62f6ce at 66.77.88.99:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.9.0(1.8.7.2)
Date: Thu, 12 Jan 2012 19:55:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, 
INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 525025075 525025075 IN IP4 66.77.88.99
s=Asterisk PBX 1.8.7.2
c=IN IP4 66.77.88.99
t=0 0
m=audio 15408 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv