Eric van der Vlist wrote: > After a migration to asterisk 1.6, I don't receive sip incoming calls > anymore. > > As fas as I understand the SIP debug traces, my server receives the > request and reject it: > > ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > <--- SIP read from UDP:212.27.52.5:5060 ---> > INVITE sip:[email protected]:5060;transport=udp SIP/2.0 > Call-ID: [email protected] > Contact: <sip:172.17.20.241:5062> > Content-Type: application/sdp > CSeq: 239836027 INVITE > From: "096160XXXX" > <sip:[email protected];user=phone>;tag=25151-GA-0eaf098c-32a97dc05 > Max-Forwards: 28 > Record-Route: <sip:C=on-88.165.134.117.5060;[email protected]:5060;lr> > To: <sip:[email protected];user=phone> > Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-RDKI-00720204-55801b4f > Allow: UPDATE,REFER,INFO > User-Agent: Cirpack/v4.41c (gw_sip) > Content-Length: 173 > > v=0 > o=cp10 125830752022 125830752022 IN IP4 212.27.52.129 > s=SIP Call > c=IN IP4 212.27.52.129 > t=0 0 > m=audio 36480 RTP/AVP 8 > b=AS:64 > a=rtpmap:8 PCMA/8000/1 > a=ptime:30 > > <-------------> > --- (13 headers 9 lines) --- > == Using SIP RTP CoS mark 5 > Sending to 212.27.52.5 : 5060 (no NAT) > Using INVITE request as basis request - > [email protected] > Found peer 'freephonie_appelsortant' for '096160XXXX' from 212.27.52.5:5060 > asterisk*CLI> > <--- Reliably Transmitting (no NAT) to 212.27.52.5:5060 ---> > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP > 212.27.52.5:5060;branch=z9hG4bK-RDKI-00720204-55801b4f;received=212.27.52.5 > From: "096160XXXX" > <sip:[email protected];user=phone>;tag=25151-GA-0eaf098c-32a97dc05 > To: <sip:[email protected];user=phone>;tag=as03dcbe68 > Call-ID: [email protected] > CSeq: 239836027 INVITE > Server: Asterisk PBX 1.6.2.0~rc2-0ubuntu1 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="78360854" > Content-Length: 0 > ++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ >
I'm not sure you've provided enough of the trace here. It finds the peer, but rejects it with a 401 Unauthorized, which is not uncommon. And I don't see any authentication information in the first INVITE. This is why the 401 is sent back, as the WWW-Authenticate line contains the realm and nonce which should be used by the other end to generate the authentication, and then send another INVITE back with authentication. Since you've only shown the two packets in the trace, it is impossible to tell what is going on beyond the 401 response from Asterisk. Leif. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
