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Summary:ASTERISK-13381: [patch] SIP Channel name is not unique
Reporter:nick_lewis (nick_lewis)Labels:
Date Opened:2009-01-16 04:31:04.000-0600Date Closed:2009-04-17 09:50:45
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) chan_sip.c-chname.patch
Description:The name of the asterisk channel that is created on an incoming sip call is not unique

There can be two trunks with different ITSP but same username e.g.
[trunk1]
username=nicklewis
host=sip.myitsp1.com

[trunk2]
username=nicklewis
host=sip.myitsp2.net

The sip channel name of the asterisk channel that is created when a call comes into these trunks is "SIP/nicklewis" in both cases. The sip channel should be named after the peername instead e.g. "SIP/trunk1" and "SIP/trunk2"
Comments:By: nick_lewis (nick_lewis) 2009-01-16 05:01:12.000-0600

This patch also helps the flash operator panel to work. The line state of a fop button is determined by the channel name (as shown in 'core show channels' etc) but the click-to-dial action of a fop button is determined by the peername (as in dialplan command Dial(SIP/peername)). Without this patch it is not possible to select a button name that gets both fop features to work
By: Joshua C. Colp (jcolp) 2009-01-22 14:38:56.000-0600

I can't actually get the behavior you are seeing in the latest version of anything. Can you retry and if it still happens provide the complete console output with debug and configuration so I can take a gander?
By: nick_lewis (nick_lewis) 2009-01-23 06:08:40.000-0600

Ok I will try to get you some more information with a live setup

What are you seeing in the latest version in response to the CLI command "core show channels" when you have an incoming sip trunk call?
By: Olle Johansson (oej) 2009-01-23 08:28:20.000-0600

"username" should not be used in setting up the channel name at all. If it is, that is a bug by itself.
By: nick_lewis (nick_lewis) 2009-01-27 03:50:11.000-0600

Agreed the channel name should be SIP/peername as sip specific information such as username should not be exposed to the rest of the pbx. The patch corrects this.

Here is an extract of conf and cli showing the problem:

[root@asterisk1 ~]# tail -15 /etc/asterisk/sip_additional.conf

[test2]
type=peer
context=from-trunk
host=sip.atltelecom.com
port=5060
outboundproxy=
outboundproxyport=
insecure=invite
username=2345
secret=2345
fromuser=2345
fromdomain=sip.atltelecom.com
fromname=PBX

[root@asterisk1 ~]# asterisk -r
asterisk1*CLI> core show channels
Channel              Location             State   Application(Data)
SIP/2345-09938190    s@app-announcement-1 Up      Wait(1)


The channel name is given as SIP/2345 instead of SIP/test2
By: Olle Johansson (oej) 2009-01-27 03:55:55.000-0600

Sorry for not believing you at first, because this is so stupid. I can't belive that this is the case...

Well, gotta fix it.
By: Olle Johansson (oej) 2009-01-27 03:56:36.000-0600

Do you have an 1.4 system to test with as well, just to check how far back we need to go?
By: nick_lewis (nick_lewis) 2009-01-27 04:16:35.000-0600

Sorry no active 1.4 system but tried on a 1.2 system and had same problem. Code examination suggests that 1.4 is also affected
By: Olle Johansson (oej) 2009-01-27 04:20:02.000-0600

OK then. Since it's not really a security issue, I don't think we'll change 1.2. But 1.4 and later needs to be fixed. Will discuss with file on how to push this forward, since this is assigned to him by him.
By: Leif Madsen (lmadsen) 2009-02-02 13:28:52.000-0600

Assigned back to file since we're waiting for feedback from him.
By: Joshua C. Colp (jcolp) 2009-02-10 08:35:40.000-0600

oej: I think we can just change this to use the peer name if non-NULL and user name otherwise. This should give the desired behavior for all scenarios. I'm just hesitant to change this in previous versions since people may have depended on the old wrong behavior... probably not though. Thoughts?
By: nick_lewis (nick_lewis) 2009-02-11 03:50:12.000-0600

It appears that pvt->peername and pvt->username are both populated when there is a match in check_user_ok() or check_peer_ok(). In cases where peername is null then username is also null so a policy of 'use username otherwise' would never actually be applied. Note that if there is no match then sip_new() internally requisitions fromdomain for the channel name.

If there is real worry about losing username from the channel name then perhaps there could be a global setting legacychannelname=yes
By: Joshua C. Colp (jcolp) 2009-02-13 13:13:30.000-0600

After looking at this further I do agree that using peername should be fine.

oej: Have you also given it a look and can agree?
By: Leif Madsen (lmadsen) 2009-03-04 12:37:19.000-0600

Ping?
By: nick_lewis (nick_lewis) 2009-03-26 06:45:07

file/Imadsen

Can this please proceed. I think OEJ made it quite clear that the application of the username in the channel was a bug in all versions. Quotes:

>"username" should not be used in setting up the channel name at all. If it is, that is a bug by itself.
>Since it's not really a security issue, I don't think we'll change 1.2. But 1.4 and later needs to be fixed.
By: Digium Subversion (svnbot) 2009-04-17 09:41:27

Repository: asterisk
Revision: 188946

U   branches/1.4/channels/chan_sip.c

------------------------------------------------------------------------
r188946 | file | 2009-04-17 09:41:26 -0500 (Fri, 17 Apr 2009) | 15 lines

Fix a bug where a value used to create the channel name was bogus.

This commit fixes the scenario where an incoming call is authenticated
using a peer entry. Previously the channel name was created using either
the username setting from the sip.conf entry or the IP address that the
call came from. Now the channel name will be created using the peer name
itself. This commit will not change the way the channel name is generated
for users or friends.

(closes issue ASTERISK-13381)
Reported by: Nick_Lewis
Patches:
     chan_sip.c-chname.patch uploaded by Nick (license 657)
Tested by: Nick_Lewis, file

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

http://svn.digium.com/view/asterisk?view=rev&revision=188946
By: Digium Subversion (svnbot) 2009-04-17 09:44:57

Repository: asterisk
Revision: 188947

_U  trunk/
U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r188947 | file | 2009-04-17 09:44:56 -0500 (Fri, 17 Apr 2009) | 22 lines

Merged revisions 188946 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
 r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines
 
 Fix a bug where a value used to create the channel name was bogus.
 
 This commit fixes the scenario where an incoming call is authenticated
 using a peer entry. Previously the channel name was created using either
 the username setting from the sip.conf entry or the IP address that the
 call came from. Now the channel name will be created using the peer name
 itself. This commit will not change the way the channel name is generated
 for users or friends.
 
 (closes issue ASTERISK-13381)
 Reported by: Nick_Lewis
 Patches:
       chan_sip.c-chname.patch uploaded by Nick (license 657)
 Tested by: Nick_Lewis, file
........

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

http://svn.digium.com/view/asterisk?view=rev&revision=188947
By: Digium Subversion (svnbot) 2009-04-17 09:46:50

Repository: asterisk
Revision: 188948

_U  branches/1.6.0/
U   branches/1.6.0/channels/chan_sip.c

------------------------------------------------------------------------
r188948 | file | 2009-04-17 09:46:50 -0500 (Fri, 17 Apr 2009) | 29 lines

Merged revisions 188947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
 r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines
 
 Merged revisions 188946 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
   r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines
   
   Fix a bug where a value used to create the channel name was bogus.
   
   This commit fixes the scenario where an incoming call is authenticated
   using a peer entry. Previously the channel name was created using either
   the username setting from the sip.conf entry or the IP address that the
   call came from. Now the channel name will be created using the peer name
   itself. This commit will not change the way the channel name is generated
   for users or friends.
   
   (closes issue ASTERISK-13381)
   Reported by: Nick_Lewis
   Patches:
         chan_sip.c-chname.patch uploaded by Nick (license 657)
   Tested by: Nick_Lewis, file
 ........
................

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

http://svn.digium.com/view/asterisk?view=rev&revision=188948
By: Digium Subversion (svnbot) 2009-04-17 09:48:52

Repository: asterisk
Revision: 188949

_U  branches/1.6.1/
U   branches/1.6.1/channels/chan_sip.c

------------------------------------------------------------------------
r188949 | file | 2009-04-17 09:48:52 -0500 (Fri, 17 Apr 2009) | 29 lines

Merged revisions 188947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
 r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines
 
 Merged revisions 188946 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
   r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines
   
   Fix a bug where a value used to create the channel name was bogus.
   
   This commit fixes the scenario where an incoming call is authenticated
   using a peer entry. Previously the channel name was created using either
   the username setting from the sip.conf entry or the IP address that the
   call came from. Now the channel name will be created using the peer name
   itself. This commit will not change the way the channel name is generated
   for users or friends.
   
   (closes issue ASTERISK-13381)
   Reported by: Nick_Lewis
   Patches:
         chan_sip.c-chname.patch uploaded by Nick (license 657)
   Tested by: Nick_Lewis, file
 ........
................

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

http://svn.digium.com/view/asterisk?view=rev&revision=188949
By: Digium Subversion (svnbot) 2009-04-17 09:50:44

Repository: asterisk
Revision: 188950

_U  branches/1.6.2/
U   branches/1.6.2/channels/chan_sip.c

------------------------------------------------------------------------
r188950 | file | 2009-04-17 09:50:43 -0500 (Fri, 17 Apr 2009) | 29 lines

Merged revisions 188947 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk

................
 r188947 | file | 2009-04-17 11:44:56 -0300 (Fri, 17 Apr 2009) | 22 lines
 
 Merged revisions 188946 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
   r188946 | file | 2009-04-17 11:41:25 -0300 (Fri, 17 Apr 2009) | 15 lines
   
   Fix a bug where a value used to create the channel name was bogus.
   
   This commit fixes the scenario where an incoming call is authenticated
   using a peer entry. Previously the channel name was created using either
   the username setting from the sip.conf entry or the IP address that the
   call came from. Now the channel name will be created using the peer name
   itself. This commit will not change the way the channel name is generated
   for users or friends.
   
   (closes issue ASTERISK-13381)
   Reported by: Nick_Lewis
   Patches:
         chan_sip.c-chname.patch uploaded by Nick (license 657)
   Tested by: Nick_Lewis, file
 ........
................

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

http://svn.digium.com/view/asterisk?view=rev&revision=188950