[asterisk-bugs] [Asterisk 0013764]: 302 Redirect (forward no answer) to bad extension causes channel to be left up (Ringing)
Asterisk Bug Tracker
noreply at bugs.digium.com
Wed Nov 26 09:21:57 CST 2008
A NOTE has been added to this issue.
======================================================================
http://bugs.digium.com/view.php?id=13764
======================================================================
Reported By: davidw
Assigned To: putnopvut
======================================================================
Project: Asterisk
Issue ID: 13764
Category: Applications/app_dial
Reproducibility: have not tried
Severity: minor
Priority: normal
Status: ready for testing
Asterisk Version: 1.6.0
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 2008-10-22 10:35 CDT
Last Modified: 2008-11-26 09:21 CST
======================================================================
Summary: 302 Redirect (forward no answer) to bad extension
causes channel to be left up (Ringing)
Description:
When testing something, we accidentally used a phone that had been
configured to call forward no-answer to a bad extension. It took us a
while to realise, so we made several calls. Subsequent core/sip show
channels show that all of these attempts are still ringing on the attempted
correct number, even though the SIP ones show the ACK to the 302 as the
last thing sent.
======================================================================
----------------------------------------------------------------------
(0095540) davidw (reporter) - 2008-11-26 09:21
http://bugs.digium.com/view.php?id=13764#c95540
----------------------------------------------------------------------
Still the same problem.
centos*CLI> core show channels
Channel Location State Application(Data)
SIP/6106-085d3b28 6106 at internal:1 Ringing AppDial((Outgoing Line))
1 active channel
0 active calls
1 call processed
[2008-11-26 14:59:11.163] NOTICE[392]: chan_sip.c:18078
handle_request_subscribe
: Received SIP subscribe for peer without mailbox: djw-messenger
centos*CLI> sip show channels
Peer User/ANR Call ID Format Hold
Last Mes
sage
192.168.81.160 6106 47ed6fbd3a871a0 0x4 (ulaw) No
Tx: ACK
1 active SIP dialog
History from a second run:
centos*CLI>
* SIP Call
1. NewChan Channel SIP/6106-085dba90 - from
421f7355177bd2ec60ab08e10c16
8e
2. TxReqRel INVITE / 102 INVITE - -UNKNOWN-
3. Rx SIP/2.0 / 102 INVITE / 100 Trying
4. Rx SIP/2.0 / 102 INVITE / 180 Ringing
5. Rx SIP/2.0 / 102 INVITE / 302 Moved Temporarily
6. TxReq ACK / 102 ACK - -UNKNOWN-
We are getting:
app_dial.c: Unable to create local channel for call forward
not the
Failed to dial on local channel for call forward to '%s'\n
we would get if the patched code were executed.
Looking at this fragment, c's nullness doesn't seem to change between the
two ifs, and your patch is in the else at the end, so the message we get is
incompatible with going down the branch with the patch.
to 'Local/6199 at internal' (cause = 0)
499 if (c) {
500 if (single)
501
ast_channel_make_compatible(o->chan, in)
501 ;
502 ast_channel_inherit_variables(in,
o->chan);
503 ast_channel_datastore_inherit(in,
o->chan);
504 } else
505 ast_log(LOG_NOTICE, "Unable to create
local chan
505 nel for call forward to '%s/%s' (cause = %d)\n", tech, stuff,
cause);
506 }
507 if (!c) {
508 ast_clear_flag64(o, DIAL_STILLGOING);
509 handle_cause(cause, num);
510 } else {
Issue History
Date Modified Username Field Change
======================================================================
2008-11-26 09:21 davidw Note Added: 0095540
======================================================================
More information about the asterisk-bugs
mailing list