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Summary:ASTERISK-25851: Bug in chan_sip - Forbidden 403
Reporter:Alexey A. Astashov (Alexey_Astashov)Labels:
Date Opened:2016-03-16 18:21:41Date Closed:2016-03-17 08:07:00
Priority:MinorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:13.7.2 Frequency of
Occurrence
Related
Issues:
Environment:Attachments:
Description:I found a problem with call processing using SIP protocol.
For example:
I'm trying to make a call from a station at an Asterisk (PBX-1, assigned numbers: 11ХХ) directly to a station at another Asterisk (PBX-2, assigned numbers: 10ХХ-11ХХ). The original CID is passed during the call. It may happen so that PBX-2 determines the incoming CID as existing in its own configuration. in this case PBX-2 returns "Forbidden 403" as if an authentication error occured.  
For example, I'm trying to make a call from number 1107 on PBX-1 (this number exists in the configuration of both PBXs), to number 1000 (PBX-2)
{code}
   Using INVITE request as basis request - 195851bb4abc4fdc17ba8da524330781@172.16.15.196:5060
   Found peer '1107' for '1107' from 172.16.15.196:5060
   [Mar 17 01:02:55] WARNING[901][C-00000049]: chan_sip.c:16702 check_auth: username mismatch, have <1107>, digest has <s>
   [Mar 17 01:02:55] NOTICE[901][C-00000049]: chan_sip.c:25603 handle_request_invite: Failed to authenticate device "Astashov A." <sip:1107@172.16.15.196>;tag=as5ce36944
{code}
Both PBXs are configured according to the documentation.

However, if I remove the station number (1107) from the PBX-2 configuration, the error does not occur, the call proceeds normally.

The error is occurring with Asterisk version 13, it does not occur on Asterisk version 11 when using the same configuration.
Also, the error does not occur when using the IAX2 protocol even on version 13.
So, the problem is only with Asterisk 13 when using SIP.

This issue is preventing me from using FMC (Fixed Mobile Convergence) provided by our mobile carrier.


My configuration is as follows:
h2.PBX-1
h3.sip.conf
{code}#cat sip.conf (PBX-1)
   [general]
   context=public
   udpbindaddr=0.0.0.0
   allowoverlap=no
   bindaddr=0.0.0.0
   bindport=5060
   register => Astashov-EDU-15-196:MyEDU-8.117!@172.16.15.196/Astashov-EDU-15-196

   [1100]
   type=friend
   context=phones
   host=dynamic
   secret=MyEDU!!
   dial=SIP/1100

   [1107]
   type=friend
   context=phones
   host=dynamic
   secret=MyEDU!!
   dial=SIP/1107

   [Astashov-EDU-8-117]
   type=friend
   secret=MyEDU-8.117!
   contex=8-117-incoming
   host=dynamic
   disallow=all
   allow=alaw
   insecure=invite
{code}
h3.extension.conf
{code}
#cat extension.conf (PBX-1)
   [globals]

   [general]
   autofallthrough=yes

   [outgoing_calls]
   exten => _10XX,1,NoOp()
   exten => _10XX,n,Dial(SIP/172.16.8.117/${EXTEN})

   [internal]
   exten => 1107,1,Verbose(1|Extension 1107)
   exten => 1107,n,Dial(SIP/1107,30)
   exten => 1107,n, Hangup()

   [public]
   exten => _11XX,1,NoOp()
   exten => _11XX,n,Dial(SIP/${EXTEN},30)
   exten => _11XX,n,Hangup()

   [phones]
   include => internal
   include => outgoing_calls

   [8-117-incoming]
   include => internal
{code}
h2.PBX-2
h3.sip.conf
{code}
#cat sip.conf  (PBX-2)
   [general]
   context=public
   udpbindaddr=0.0.0.0
   allowoverlap=no
   bindaddr=0.0.0.0
   bindport=5060
   register => Astashov-EDU-8-117:MyEDU-8.117!@172.16.15.196/Astashov-EDU-8-117

   [1000]
   type=friend
   context=phones
   host=dynamic
   secret=MyEDU!!
   dial=SIP/1000

   [1107]
   type=friend
   context=phones
   host=dynamic
   secret=MyEDU!!
   dial=SIP/1107

   [Astashov-EDU-15-196]
   type=friend
   secret=MyEDU-8.117!
   contex=15-196-incoming
   host=dynamic
   disallow=all
   allow=alaw
   insecure=invite
{code}
h3.extension.conf
{code}
#cat extension.conf (PBX-2)
   [globals]

   [general]
   autofallthrough=yes

   [outgoing_calls]
   exten => _11XX,1,NoOp()
   exten => _11XX,n,Dial(SIP/172.16.15.196/${EXTEN})

   [internal]
   exten => 1000,1,Verbose(1|Extension 1000)
   exten => 1000,n,Dial(SIP/1000,30)
   exten => 1000,n, Hangup()

   [public]
   exten => _10XX,1,NoOp()
   exten => _10XX,n,Dial(SIP/${EXTEN},30)
   exten => _10XX,n,Hangup()

   [phones]
   include => internal
   include => outgoing_calls

   [15-196-incoming]
   include => internal
{code}
Comments:By: Asterisk Team (asteriskteam) 2016-03-16 18:21:42.256-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
By: Joshua C. Colp (jcolp) 2016-03-17 06:18:52.799-0500

As mentioned on the community site I do not believe this is an actual bug. If discussion there yields otherwise then we can reopen this.