Summary: | ASTERISK-03417: SIP digest auth replaces FROM: with what's in TO: | ||
Reporter: | sirs69 (sirs69) | Labels: | |
Date Opened: | 2005-02-02 07:20:40.000-0600 | Date Closed: | 2008-01-15 15:24:35.000-0600 |
Priority: | Minor | Regression? | No |
Status: | Closed/Complete | Components: | Core/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ( 0) sipchall.txt | |
Description: | Problem: When proxy requests digest challenge (SIP) Asterisk responds normally with the exception that for some reason it changes the FROM: (Also changes Contact: )to what's in the original TO: line. Why on earth is it doing this?! It must be a bug, I've gone over my extensions.conf several times to no avail. It seems older vers of ast don't do this. near end is error: Sip read: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK243afb2c From: "Matt S" <sip:+13142664000@206.80.66.138>;tag=as5f22d23c To: <sip:3142498555@sipfarm.netlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e 58.2b61 Call-ID: 6b3bf846529168f37cd8465357608dee@206.80.66.138 CSeq: 102 INVITE Proxy-Authenticate: Digest realm="sipfarm.netlogic.net", nonce="blah blah", qop="auth" Server: Sip EXpress router (0.8.14 (i386/linux)) Content-Length: 0 Warning: 392 206.80.70.46:5060 "Noisy feedback tells: pid=27326 req_src_ip=206.80.66.138 req_src_port=5060 in_uri=sip:3142498555@sipfarm.netlogic.net out_uri=sip:3142498555@sipfarm.netlogic.net via_cnt==1" 10 headers, 0 lines Transmitting: ACK sip:3142498555@sipfarm.netlogic.net SIP/2.0 Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK243afb2c From: "Matt S" <sip:+13142664000@206.80.66.138>;tag=as5f22d23c To: <sip:3142498555@sipfarm.netlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e 58.2b61 Contact: <sip:+13142664000@206.80.66.138> Call-ID: 6b3bf846529168f37cd8465357608dee@206.80.66.138 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 (no NAT) to 206.80.70.46:5060 We're at 206.80.66.138 port 60922 Answering with capability 0x100 (g729) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting: INVITE sip:3142498555@sipfarm.netlogic.net SIP/2.0 Via: SIP/2.0/UDP 206.80.66.138:5060;branch=z9hG4bK670cb15b From: "Matt S" <sip:3142498555@206.80.66.138>;tag=as5f22d23c To: <sip:3142498555@sipfarm.netlogic.net> Contact: <sip:3142498555@206.80.66.138> --snip-- ****** ADDITIONAL INFORMATION ****** The only reason this is considered major is because our SIP handoff provider uses callerid as one of it's identifiers for billing, not only that if you were to dial a sprint phone for example and the users phone didn't have a password set you'd get right into their voicemail ;) | ||
Comments: | By: Brian West (bkw918) 2005-02-02 09:38:11.000-0600 First find in the RFC what it says about this. It maybe wrong or a config issue. Also NOT MAJOR. bkw By: Mark Spencer (markster) 2005-02-02 10:53:18.000-0600 Please attach the full sip debug of the transaction as an attachment ending in .txt, not just a snippit, pointing out which portion you believe to be in error. Thanks! By: sirs69 (sirs69) 2005-02-02 12:10:58.000-0600 I marked the first from line in the txt file, I upgraded to latest CVS and still found the same issue. Ty :-) By: Mark Spencer (markster) 2005-02-02 12:36:23.000-0600 I think I see the issue. Please try latest CVS and confirm that fixes your problem. Thanks! By: Mark Spencer (markster) 2005-02-02 23:41:35.000-0600 Assuming fixed, please feel free to reopen if you continue to have trouble. Thanks. By: sirs69 (sirs69) 2005-02-03 07:02:57.000-0600 Sorry for delay, I tried it again and this time I'm getting forbidden (challenge fail). I doublechecked the passphrases, not only that, nothing has changed on any one of the configs :-) By: sirs69 (sirs69) 2005-02-03 07:06:01.000-0600 Ok, weird I tried it again and it worked.. Must have been a fluke.. ?? By: nick (nick) 2005-02-03 13:59:17.000-0600 OK, looks like this works. Reclosing. By: Russell Bryant (russell) 2005-02-06 22:13:40.000-0600 fixed in 1.0 By: Digium Subversion (svnbot) 2008-01-15 15:24:07.000-0600 Repository: asterisk Revision: 4952 U trunk/channels/chan_sip.c ------------------------------------------------------------------------ r4952 | markster | 2008-01-15 15:24:06 -0600 (Tue, 15 Jan 2008) | 2 lines Make sure we always transmit the same from line (bug ASTERISK-3417) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=4952 By: Digium Subversion (svnbot) 2008-01-15 15:24:35.000-0600 Repository: asterisk Revision: 4977 U branches/v1-0/channels/chan_sip.c ------------------------------------------------------------------------ r4977 | russell | 2008-01-15 15:24:35 -0600 (Tue, 15 Jan 2008) | 2 lines Make sure we always transmit the same from line (bug ASTERISK-3417) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=4977 |