Gordon Messmer
2012-Jan-17 21:06 UTC
[asterisk-users] SIP trunk call initiated as Anonymous@anonymous.invalid
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