[asterisk-users] Can't make Asterisk send authentication to remote peer on INVITE [SOLVED]

[Ben WIlliams]

Problem solved - It was missing username setting, I had assumed
fromuser would be used for authentication.

On Sat, Apr 14, 2012 at 9:30 PM, Ben WIlliams
<bwilliams+asterisk at jadeworld.com> wrote:
> This is a really simple problem that I just can't get to work. There
> are two Asterisk servers with the following sip user and peer. When a
> call is attempted, Asterisk is not sending authentication details in
> response to the 401. Note, if the secret is blank on 172.16.0.2 test,
> the INVITE succeeds.
>
> on 172.16.0.2:
>
> [test]
> type=friend
> secret=abcde
> host=dynamic
> context=demo
>
> on 172.16.0.1 :
>
> [natty]
> type=peer
> host=172.16.0.2
> fromuser=test
> secret=abcde
>
> originate SIP/natty/1234568 extension 200
>  == Using SIP RTP CoS mark 5
> Audio is at 172.16.0.1 port 19486
> Adding codec 0x2 (gsm) to SDP
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 172.16.0.2:5060:
> INVITE sip:1234568 at 172.16.0.2 SIP/2.0
> Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport
> Max-Forwards: 70
> From: "asterisk" <sip:test at 172.16.0.1:5066>;tag=as1689b2b6
> To: <sip:1234568 at 172.16.0.2>
> Contact: <sip:test at 172.16.0.1:5066>
> Call-ID: 2353cf0e59596e285c684b44220f8915 at 172.16.0.1
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
> Date: Sat, 14 Apr 2012 09:10:38 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 290
>
> v=0
> o=root 1594270426 1594270426 IN IP4 172.16.0.1
> s=Asterisk PBX 1.6.2.9-2ubuntu2.1
> c=IN IP4 172.16.0.1
> t=0 0
> m=audio 19486 RTP/AVP 3 0 8 101
> a=rtpmap:3 GSM/8000
> 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
>
> ---
>
> <--- SIP read from UDP:172.16.0.2:5060 --->
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 172.16.0.1:5066;branch=z9hG4bK59f50e30;received=172.16.0.1;rport=5066
> From: "asterisk" <sip:test at 172.16.0.1:5066>;tag=as1689b2b6
> To: <sip:1234568 at 172.16.0.2>;tag=as1a6c2364
> Call-ID: 2353cf0e59596e285c684b44220f8915 at 172.16.0.1
> CSeq: 102 INVITE
> Server: Asterisk PBX 1.6.2.9-2ubuntu2.1
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7a03a1d3"
> Content-Length: 0
>
>
> <------------->
> --- (11 headers 0 lines) ---
> Transmitting (no NAT) to 172.16.0.2:5060:
> ACK sip:1234568 at 172.16.0.2 SIP/2.0
> Via: SIP/2.0/UDP 172.16.0.1:5066;branch=z9hG4bK59f50e30;rport
> Max-Forwards: 70
> From: "asterisk" <sip:test at 172.16.0.1:5066>;tag=as1689b2b6
> To: <sip:1234568 at 172.16.0.2>;tag=as1a6c2364
> Contact: <sip:test at 172.16.0.1:5066>
> Call-ID: 2353cf0e59596e285c684b44220f8915 at 172.16.0.1
> CSeq: 102 ACK
> User-Agent: Asterisk PBX 1.6.2.9-2ubuntu2.1
> Content-Length: 0
>
>
> ---
> [Apr 14 21:10:38] NOTICE[31158]: chan_sip.c:17975
> handle_response_invite: Failed to authenticate on INVITE to
> '"asterisk" <sip:test at 172.16.0.1:5066>;tag=as1689b2b6'