[Home]

Summary:ASTERISK-11701: [patch] Add Server: instead of User-Agent: header in Asterisk generated SIP responses
Reporter:Raj Jain (rjain)Labels:
Date Opened:2008-03-22 19:36:09Date Closed:2008-03-25 13:07:22
Priority:TrivialRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) chan_sip.c.diff
Description:Asterisk currently inserts User-Agent: header in the SIP responses it generates. A SIP UAS should insert Server: header instead. The Server: and User-Agent: are meant for human consumption and not automaton, thus this isn't really a software bug. But, it is inconsistent with other SIP implementations and a bit of annoyance when you're looking at SIP traces that include Asterisk SIP messaging.  

Below are sections of RFC 3261 that explain the roles of User-Agent: and Server: headers.

20.35 Server

  The Server header field contains information about the software used
  by the UAS to handle the request.

  Revealing the specific software version of the server might allow the
  server to become more vulnerable to attacks against software that is
  known to contain security holes.  Implementers SHOULD make the Server
  header field a configurable option.

  Example:

     Server: HomeServer v2

20.41 User-Agent

  The User-Agent header field contains information about the UAC
  originating the request.  The semantics of this header field are
  defined in [H14.43].

  Revealing the specific software version of the user agent might allow
  the user agent to become more vulnerable to attacks against software
  that is known to contain security holes.  Implementers SHOULD make
  the User-Agent header field a configurable option.

  Example:

     User-Agent: Softphone Beta1.5



****** ADDITIONAL INFORMATION ******

Correct usage of Server: header in a SIP response message after appliying this patch:

SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.15.100:7419;received=192.168.15.100
From: <sip:9001@192.168.15.101>;tag=7acfe13be2914fe29495ff9151a1ec06;epid=66ccc20c01
To: <sip:9001@192.168.15.101>;tag=as7adb408b
Call-ID: 09c0e782ad51448d93d060b25c147bbe
CSeq: 45 REGISTER
Server: Asterisk PBX SVN-trunk-r110578M
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Expires: 120
Contact: <sip:192.168.15.100:7419>;expires=120
Date: Sat, 22 Mar 2008 13:35:11 GMT
Content-Length: 0
Comments:By: Olle Johansson (oej) 2008-03-25 05:46:19

I've noticed this too, but had other more important issues to work with... :-)

I can't see this as an important bug fix for 1.4, so let's merge it into trunk.
By: Digium Subversion (svnbot) 2008-03-25 05:49:44

Repository: asterisk
Revision: 110625

U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r110625 | oej | 2008-03-25 05:49:42 -0500 (Tue, 25 Mar 2008) | 6 lines

Use the "Server" header when responding to SIP requests.
(closes issue ASTERISK-11701)
Reported by: rjain
Patches:
     chan_sip.c.diff uploaded by rjain (license 226)

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=110625
By: Digium Subversion (svnbot) 2008-03-25 10:35:14

Repository: asterisk
Revision: 110634

_U  branches/1.6.0/

------------------------------------------------------------------------
r110634 | file | 2008-03-25 10:35:13 -0500 (Tue, 25 Mar 2008) | 13 lines

Blocked revisions 110625 via svnmerge

........
r110625 | oej | 2008-03-25 07:54:07 -0300 (Tue, 25 Mar 2008) | 6 lines

Use the "Server" header when responding to SIP requests.
(closes issue ASTERISK-11701)
Reported by: rjain
Patches:
     chan_sip.c.diff uploaded by rjain (license 226)

........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=110634
By: Digium Subversion (svnbot) 2008-03-25 12:59:04

Repository: asterisk
Revision: 110693

_U  team/group/cdr_backend_ast_str/
U   team/group/cdr_backend_ast_str/CHANGES
U   team/group/cdr_backend_ast_str/Makefile
U   team/group/cdr_backend_ast_str/channels/chan_iax2.c
U   team/group/cdr_backend_ast_str/channels/chan_sip.c
U   team/group/cdr_backend_ast_str/configs/extensions.conf.sample
U   team/group/cdr_backend_ast_str/configs/sip.conf.sample
U   team/group/cdr_backend_ast_str/configs/voicemail.conf.sample
U   team/group/cdr_backend_ast_str/include/asterisk/options.h
U   team/group/cdr_backend_ast_str/main/app.c
U   team/group/cdr_backend_ast_str/main/asterisk.c
U   team/group/cdr_backend_ast_str/main/channel.c

------------------------------------------------------------------------
r110693 | tilghman | 2008-03-25 12:58:50 -0500 (Tue, 25 Mar 2008) | 140 lines

Merged revisions 110610,110615,110619,110621,110625,110629,110631,110636,110639,110689,110691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r110610 | file | 2008-03-24 10:28:25 -0500 (Mon, 24 Mar 2008) | 6 lines

Only print out the set_address_from_contact host verbose message if debugging is enabled on the dialog.
(closes issue ASTERISK-11703)
Reported by: rjain
Patches:
     chan_sip.c.diff uploaded by rjain (license 226)

................
r110615 | russell | 2008-03-24 12:36:04 -0500 (Mon, 24 Mar 2008) | 10 lines

Merged revisions 110614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008) | 2 lines

Turn a NOTICE into a DEBUG message.

........

................
r110619 | mmichelson | 2008-03-24 14:19:37 -0500 (Mon, 24 Mar 2008) | 23 lines

Merged revisions 110618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar 2008) | 15 lines

This is a revert for revision 108288. The reason is that that revision
was not for an actual bug fix per se, and so it really should not have been in 1.4 in
the first place. Plus, people who compile with DO_CRASH are more likely
to encounter a crash due to this change. While I think the usage of DO_CRASH
in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4
and should be done instead in a developer branch based on trunk
so that all scheduler functions are fixed at once.

I also am reverting the change to trunk and 1.6 since they also suffer from
the DO_CRASH potential.

(closes issue ASTERISK-11695)
Reported by: qq12345


........

................
r110621 | mmichelson | 2008-03-24 15:14:07 -0500 (Mon, 24 Mar 2008) | 11 lines

Remove the "Event: registration" header from Asterisk-generated
SIP REGISTER requests. rjain points out that RFC 3265 specifies
that the Event: header is not a valid header for REGISTER requests
and that the "registration" value is not defined at IANA.

(closes issue ASTERISK-11702)
Reported by: rjain
Patches:
     chan_sip.c.diff uploaded by rjain (license 226)


................
r110625 | oej | 2008-03-25 05:54:07 -0500 (Tue, 25 Mar 2008) | 6 lines

Use the "Server" header when responding to SIP requests.
(closes issue ASTERISK-11701)
Reported by: rjain
Patches:
     chan_sip.c.diff uploaded by rjain (license 226)

................
r110629 | file | 2008-03-25 09:39:45 -0500 (Tue, 25 Mar 2008) | 12 lines

Merged revisions 110628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines

Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue ASTERISK-9755)
Reported by: tracinet

........

................
r110631 | file | 2008-03-25 10:18:41 -0500 (Tue, 25 Mar 2008) | 4 lines

Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue ASTERISK-8972)
Reported by: sunder

................
r110636 | mmichelson | 2008-03-25 10:41:33 -0500 (Tue, 25 Mar 2008) | 15 lines

Merged revisions 110635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines

When reverting a commit, I accidentally left in this bit which was an experiment
to see what would happen. It passed the compile test, and I didn't notice I had
left this change in too.

So this is a revert of a revert...sort of.


........

................
r110639 | mmichelson | 2008-03-25 10:44:01 -0500 (Tue, 25 Mar 2008) | 3 lines

Oops here too. I need to stop coding for a while...


................
r110689 | tilghman | 2008-03-25 12:40:28 -0500 (Tue, 25 Mar 2008) | 6 lines

Update the sample configuration, to use Macro less (since it's now deprecated).
(closes issue ASTERISK-11716)
Reported by: pprindeville
Patches:
      bugid-0012293.1.6.patch uploaded by pprindeville (license 347)

................
r110691 | tilghman | 2008-03-25 12:46:34 -0500 (Tue, 25 Mar 2008) | 6 lines

Update sample configurations to make virtual hosting more obvious.
(closes issue ASTERISK-11414)
Reported by: pprindeville
Patches:
      acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347)

................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=110693
By: Digium Subversion (svnbot) 2008-03-25 13:07:22

Repository: asterisk
Revision: 110694

_U  team/murf/bug11210/
U   team/murf/bug11210/CHANGES
U   team/murf/bug11210/Makefile
U   team/murf/bug11210/channels/chan_iax2.c
U   team/murf/bug11210/channels/chan_sip.c
U   team/murf/bug11210/configs/extensions.conf.sample
U   team/murf/bug11210/configs/sip.conf.sample
U   team/murf/bug11210/configs/voicemail.conf.sample
U   team/murf/bug11210/include/asterisk/options.h
U   team/murf/bug11210/main/app.c
U   team/murf/bug11210/main/asterisk.c
U   team/murf/bug11210/main/channel.c

------------------------------------------------------------------------
r110694 | murf | 2008-03-25 13:07:18 -0500 (Tue, 25 Mar 2008) | 140 lines

Merged revisions 110610,110615,110619,110621,110625,110629,110631,110636,110639,110689,110691 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
r110610 | file | 2008-03-24 09:28:25 -0600 (Mon, 24 Mar 2008) | 6 lines

Only print out the set_address_from_contact host verbose message if debugging is enabled on the dialog.
(closes issue ASTERISK-11703)
Reported by: rjain
Patches:
     chan_sip.c.diff uploaded by rjain (license 226)

................
r110615 | russell | 2008-03-24 11:36:04 -0600 (Mon, 24 Mar 2008) | 10 lines

Merged revisions 110614 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008) | 2 lines

Turn a NOTICE into a DEBUG message.

........

................
r110619 | mmichelson | 2008-03-24 13:19:37 -0600 (Mon, 24 Mar 2008) | 23 lines

Merged revisions 110618 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar 2008) | 15 lines

This is a revert for revision 108288. The reason is that that revision
was not for an actual bug fix per se, and so it really should not have been in 1.4 in
the first place. Plus, people who compile with DO_CRASH are more likely
to encounter a crash due to this change. While I think the usage of DO_CRASH
in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4
and should be done instead in a developer branch based on trunk
so that all scheduler functions are fixed at once.

I also am reverting the change to trunk and 1.6 since they also suffer from
the DO_CRASH potential.

(closes issue ASTERISK-11695)
Reported by: qq12345


........

................
r110621 | mmichelson | 2008-03-24 14:14:07 -0600 (Mon, 24 Mar 2008) | 11 lines

Remove the "Event: registration" header from Asterisk-generated
SIP REGISTER requests. rjain points out that RFC 3265 specifies
that the Event: header is not a valid header for REGISTER requests
and that the "registration" value is not defined at IANA.

(closes issue ASTERISK-11702)
Reported by: rjain
Patches:
     chan_sip.c.diff uploaded by rjain (license 226)


................
r110625 | oej | 2008-03-25 04:54:07 -0600 (Tue, 25 Mar 2008) | 6 lines

Use the "Server" header when responding to SIP requests.
(closes issue ASTERISK-11701)
Reported by: rjain
Patches:
     chan_sip.c.diff uploaded by rjain (license 226)

................
r110629 | file | 2008-03-25 08:39:45 -0600 (Tue, 25 Mar 2008) | 12 lines

Merged revisions 110628 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines

Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases.
(closes issue ASTERISK-9755)
Reported by: tracinet

........

................
r110631 | file | 2008-03-25 09:18:41 -0600 (Tue, 25 Mar 2008) | 4 lines

Add a special dialplan variable to chan_sip which will cause an audio file to be played upon completion of an attended transfer.
(closes issue ASTERISK-8972)
Reported by: sunder

................
r110636 | mmichelson | 2008-03-25 09:41:33 -0600 (Tue, 25 Mar 2008) | 15 lines

Merged revisions 110635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines

When reverting a commit, I accidentally left in this bit which was an experiment
to see what would happen. It passed the compile test, and I didn't notice I had
left this change in too.

So this is a revert of a revert...sort of.


........

................
r110639 | mmichelson | 2008-03-25 09:44:01 -0600 (Tue, 25 Mar 2008) | 3 lines

Oops here too. I need to stop coding for a while...


................
r110689 | tilghman | 2008-03-25 11:40:28 -0600 (Tue, 25 Mar 2008) | 6 lines

Update the sample configuration, to use Macro less (since it's now deprecated).
(closes issue ASTERISK-11716)
Reported by: pprindeville
Patches:
      bugid-0012293.1.6.patch uploaded by pprindeville (license 347)

................
r110691 | tilghman | 2008-03-25 11:46:34 -0600 (Tue, 25 Mar 2008) | 6 lines

Update sample configurations to make virtual hosting more obvious.
(closes issue ASTERISK-11414)
Reported by: pprindeville
Patches:
      acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347)

................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=110694