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Summary:ASTERISK-13041: [patch] Reject an incoming call to peer due to call limit with "603 Declined". It`s not correct.
Reporter:still_nsk (still_nsk)Labels:
Date Opened:2008-11-10 05:40:05.000-0600Date Closed:2008-11-20 11:33:03.000-0600
Priority:MinorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 13867.diff
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@192.168.222.19' in 6400 ms (Method: INVITE)
 == Everyone is busy/congested at this time (0:0/0/0)
   -- Executing [25899@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@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@192.168.1.10>;tag=as575a51da
To: <sip:25899@192.168.1.240>;tag=as22e9478b
Call-ID: 1150b8685a251a9e70cf8562126f42cd@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@192.168.1.240>
Content-Length: 0                                                                                                                                      
Comments:By: Mark Michelson (mmichelson) 2008-11-10 14:45:10.000-0600

Give the attached patch a try. It should fix your issue.
By: still_nsk (still_nsk) 2008-11-13 03:09:52.000-0600

Yes, this patch fix problem.

<------------>
   -- Executing [25899@from-pstn:1] Dial("SIP/192.168.1.10-08210398", "SIP/11201|30|t") in new stack
[Nov 13 15:05:02] ERROR[24898]: chan_sip.c:3337 update_call_counter: Call to peer '11201' rejected due to usage limit of 1
   -- Couldn't call 11201
Scheduling destruction of SIP dialog '4ccb013f05c96c8923300f5e4e93f546@192.168.222.19' in 32000 ms (Method: INVITE)
 == Everyone is busy/congested at this time (0:0/0/0)
   -- Executing [25899@from-pstn:2] Hangup("SIP/192.168.1.10-08210398", "") in new stack
 == Spawn extension (from-pstn, 25899, 2) exited non-zero on 'SIP/192.168.1.10-08210398'
Scheduling destruction of SIP dialog '653efaf7016ca342311d12677f80ee9e@192.168.1.10' in 32000 ms (Method: INVITE)

<--- Reliably Transmitting (NAT) to 192.168.1.10:5060 --->
SIP/2.0 486 Busy here
Via: SIP/2.0/UDP
By: Leif Madsen (lmadsen) 2008-11-20 09:08:55.000-0600

Issue assigned to putnopvut because he made the patch, and it works! SHIP IT!
By: Digium Subversion (svnbot) 2008-11-20 11:33:01.000-0600

Repository: asterisk
Revision: 158053

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

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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 ASTERISK-13041)
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