[asterisk-bugs] [Asterisk 0018213]: Missing SIP status code
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Nov 3 07:38:34 CDT 2010
A NOTE has been added to this issue.
======================================================================
https://issues.asterisk.org/view.php?id=18213
======================================================================
Reported By: colonel72
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 18213
Category: Channels/chan_sip/Interoperability
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.8.0
JIRA:
Regression: No
Reviewboard Link:
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Request Review:
======================================================================
Date Submitted: 2010-10-26 14:18 CDT
Last Modified: 2010-11-03 07:38 CDT
======================================================================
Summary: Missing SIP status code
Description:
When a phone not responding (HangupCauseCode:20) the caller from the sip
trunk receive SIP response 503. The correct SIP response "Temporarily
unavailable" 480! (Also occur Asterisk 1.6.2.12)
======================================================================
----------------------------------------------------------------------
(0128586) colonel72 (reporter) - 2010-11-03 07:38
https://issues.asterisk.org/view.php?id=18213#c128586
----------------------------------------------------------------------
To resolve: /usr/src/asterisk-1.8.0/channels/chan_sip.c change the lines:
6361-
case AST_CONTROL_CONGESTION:
if (ast->_state != AST_STATE_UP) {
switch (ast->hangupcause){
case AST_CAUSE_UNREGISTERED:
transmit_response_reliable(p, "480
Temporarily unavailable", &p->initreq);
case AST_CAUSE_NO_ANSWER:
transmit_response_reliable(p, "480
Temporarily unavailable", &p->initreq);
case AST_CAUSE_NORMAL_UNSPECIFIED:
transmit_response_reliable(p, "480
Temporarily unavailable", &p->initreq);
default:
transmit_response_reliable(p, "503 Service
Unavailable", &p->initreq);
}
p->invitestate = INV_COMPLETED;
sip_alreadygone(p);
ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
break;
}
res = -1;
break;
Issue History
Date Modified Username Field Change
======================================================================
2010-11-03 07:38 colonel72 Note Added: 0128586
======================================================================
More information about the asterisk-bugs
mailing list