Summary:ASTERISK-03417: SIP digest auth replaces FROM: with what's in TO:
Reporter:sirs69 (sirs69)Labels:
Date Opened:2005-02-02 07:20:40.000-0600Date Closed:2008-01-15 15:24:35.000-0600
Versions:Frequency of
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;branch=z9hG4bK243afb2c
From: "Matt S" <sip:+13142664000@>;tag=as5f22d23c
To: <sip:3142498555@sipfarm.netlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e
Call-ID: 6b3bf846529168f37cd8465357608dee@
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 "Noisy feedback tells:  pid=27326 req_src_ip= 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
ACK sip:3142498555@sipfarm.netlogic.net SIP/2.0
Via: SIP/2.0/UDP;branch=z9hG4bK243afb2c
From: "Matt S" <sip:+13142664000@>;tag=as5f22d23c
To: <sip:3142498555@sipfarm.netlogic.net>;tag=b27e1a1d33761e85846fc98f5f3a7e
Contact: <sip:+13142664000@>
Call-ID: 6b3bf846529168f37cd8465357608dee@
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

(no NAT) to
We're at 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;branch=z9hG4bK670cb15b
From: "Matt S" <sip:3142498555@>;tag=as5f22d23c
To: <sip:3142498555@sipfarm.netlogic.net>
Contact: <sip:3142498555@>



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.



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)



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)