--- On Wed, 5/12/10, Philipp von Klitzing <[email protected]> wrote:
> > <--- SIP read from 192.168.250.111:5060 ---> > > SIP/2.0 407 Proxy Authentication Required > > You need to run the SIP debug on 192.168.250.111 to learn > more about WHY > the 407 is issued. Have a close look and you are likely to > understand it > right away. > > Also: Do not forget the "reload" after applying changes to > sip.conf. I always do a "sip reload" after changes to sip settings. Here are the SIP messages on 192.168.250.111 (Asterisk server 1 - receiving end): <-- SIP read from 192.168.250.112:5060: INVITE sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: "device" <sip:[email protected]>;tag=as18a568d6 To: <sip:[email protected]> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Date: Wed, 12 May 2010 09:20:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO upported: replaces Content-Type: application/sdp Content-Length: 270 v=0 o=root 20611 20611 IN IP4 192.168.250.112 s=session c=IN IP4 192.168.250.112 t=0 0 m=audio 14648 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=silenceSupp:off - - - - a=ptime:20 a=sendrecv --- (14 headers 13 lines) --- Using INVITE request as basis request - [email protected] Sending to 192.168.250.112 : 5060 (NAT) Reliably Transmitting (NAT) to 192.168.250.112:5060: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;received=192.168.250.112;rport=5060 From: "device" <sip:[email protected]>;tag=as18a568d6 To: <sip:[email protected]>;tag=as57a19dac Call-ID: [email protected] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1327c5b6" Content-Length: 0 --- Scheduling destruction of call '[email protected]' in 15000 ms Found user '4053' <-- SIP read from 192.168.250.112:5060: ACK sip:[email protected] SIP/2.0 Via: SIP/2.0/UDP 192.168.250.112:5060;branch=z9hG4bK504ddbc7;rport From: "device" <sip:[email protected]>;tag=as18a568d6 To: <sip:[email protected]>;tag=as57a19dac Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 ACK User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Can you deduce from this what I'm doing wrong? Thanks, Vieri -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
