[asterisk-bugs] [Asterisk 0013867]: [patch] Reject an incoming call to peer due to call limit with "603 Declined". It`s not correct.

Thu Nov 20 11:33:02 CST 2008

A NOTE has been added to this issue. 
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http://bugs.digium.com/view.php?id=13867 
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Reported By:                still_nsk
Assigned To:                putnopvut
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Project:                    Asterisk
Issue ID:                   13867
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     assigned
Asterisk Version:           Older 1.4 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
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Date Submitted:             2008-11-10 05:40 CST
Last Modified:              2008-11-20 11:33 CST
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Summary:                    [patch] Reject an incoming call to peer  due to call
limit with "603 Declined". It`s not correct.
Description: 
Correct - 486 "486 Busy here"
Because when convert(RFC3398) SIP Response to ISUP Q.391(SS7), calling
party cause value - 21 call rejected.

Example:

[Nov  7 15:29:54] ERROR[9791]: chan_sip.c:3335 update_call_counter: Call
to peer '11201' rejected due to usage limit of 1
    -- Couldn't call 11201
Scheduling destruction of SIP dialog
'2a58f6a26d6e095069c761a776990f36 at 192.168.222.19' in 6400 ms (Method:
INVITE)
  == Everyone is busy/congested at this time (0:0/0/0)
    -- Executing [25899 at from-pstn:2] Hangup("SIP/192.168.160.10-08175dc8",
"") in new stack
  == Spawn extension (from-pstn, 25899, 2) exited non-zero on
'SIP/192.168.160.10-08175dc8'
Scheduling destruction of SIP dialog
'1150b8685a251a9e70cf8562126f42cd at 192.168.160.10' in 32000 ms (Method:
INVITE)

<--- Reliably Transmitting (NAT) to 192.168.160.10:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP
192.168.1.10:5060;branch=z9hG4bK44373204;received=192.168.1.10;rport=5060
From:<sip:25411 at 192.168.1.10>;tag=as575a51da
To: <sip:25899 at 192.168.1.240>;tag=as22e9478b
Call-ID: 1150b8685a251a9e70cf8562126f42cd at 192.168.160.10
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:25899 at 192.168.1.240>
Content-Length: 0                                                         

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---------------------------------------------------------------------- 
 (0095194) svnbot (reporter) - 2008-11-20 11:33
 http://bugs.digium.com/view.php?id=13867#c95194 
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Repository: asterisk
Revision: 158053

U   branches/1.4/apps/app_dial.c
U   branches/1.4/channels/chan_sip.c

------------------------------------------------------------------------
r158053 | mmichelson | 2008-11-20 11:33:01 -0600 (Thu, 20 Nov 2008) | 12
lines

Make sure to set the hangup cause on the calling channel in the case
that ast_call() fails. For incoming SIP channels, this was causing
us to send a 603 instead of a 486 when the call-limit was reached on
the destination channel.

(closes issue http://bugs.digium.com/view.php?id=13867)
Reported by: still_nsk
Patches:
      13867.diff uploaded by putnopvut (license 60)
Tested by: blitzrage

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http://svn.digium.com/view/asterisk?view=rev&revision=158053 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-11-20 11:33 svnbot         Checkin                                      
2008-11-20 11:33 svnbot         Note Added: 0095194                          
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