[asterisk-bugs] [Asterisk 0012857]: Hangup extension doesn't seem to work in ast-1.4.19.1
noreply at bugs.digium.com
noreply at bugs.digium.com
Mon Jun 16 07:55:26 CDT 2008
A NOTE has been added to this issue.
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http://bugs.digium.com/view.php?id=12857
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Reported By: binuvb
Assigned To:
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Project: Asterisk
Issue ID: 12857
Category: Applications/General
Reproducibility: always
Severity: minor
Priority: normal
Status: feedback
Asterisk Version: 1.4.19
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: 06-14-2008 07:24 CDT
Last Modified: 06-16-2008 07:55 CDT
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Summary: Hangup extension doesn't seem to work in
ast-1.4.19.1
Description:
Hello all,
I have made a call back service using the call files. Here people
would call on some predetermined number, the system recognizes the caller
id (we have a caller id database) and disconnects the call and call back.
If the system could not recognize the caller id, it plays a message and
gives options to register the caller id, etc. All these works well in
asterisk 1.4.4. The problem started when I upgraded asterisk to version
1.4.19.1.
I am attaching the portion of the context.
When the system receives a call, it just hangs up and the rest is
done in "h" extension.
I guess this is a problem with h extension in the version 1.4.19.1 (
as I said earlier that it works very well with 1.4.4).
In the asterisk console I am able to see
"Playback(prompt_register_cid)" , but it is not actually playing.
Thanks all
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----------------------------------------------------------------------
binuvb - 06-16-08 07:55
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-- Executing [17813821904 at trikon:1] NoOp("SIP/216.143.130.12-081fd2e8",
"912224318815") in new stack
-- Executing [17813821904 at trikon:2]
Goto("SIP/216.143.130.12-081fd2e8", "callback|s|1") in new stack
-- Goto (callback,s,1)
-- Executing [s at callback:1] Hangup("SIP/216.143.130.12-081fd2e8", "")
in new stack
== Spawn extension (callback, s, 1) exited non-zero on
'SIP/216.143.130.12-081fd2e8'
-- Executing [h at callback:1] Set("SIP/216.143.130.12-081fd2e8",
"authcallerid = 0|caller=912224318815|callerfull=912224318815") in new
stack
[Jun 16 18:31:21] WARNING[2456]: pbx.c:5870 pbx_builtin_setvar: Setting
multiple variables at once within Set is deprecated. Please separate each
name/value pair into its own line.
-- Executing [h at callback:2] NoOp("SIP/216.143.130.12-081fd2e8",
"912224318815") in new stack
-- Executing [h at callback:3] NoOp("SIP/216.143.130.12-081fd2e8", "") in
new stack
-- Executing [h at callback:4] NoOp("SIP/216.143.130.12-081fd2e8",
"912224318815") in new stack
-- Executing [h at callback:5] Set("SIP/216.143.130.12-081fd2e8",
"caller=4318815") in new stack
-- Executing [h at callback:6] GotoIf("SIP/216.143.130.12-081fd2e8",
"0?trikon|s|11") in new stack
-- Executing [h at callback:7] DeadAGI("SIP/216.143.130.12-081fd2e8",
"/usr/local/ownmail/bin/auth_user|1|4318815") in new stack
-- Launched AGI Script /usr/local/ownmail/bin/auth_user
-- AGI Script Executing Application: (Set) Options: (authcallerid=0)
-- AGI Script /usr/local/ownmail/bin/auth_user completed, returning 0
-- Executing [h at callback:8] GotoIf("SIP/216.143.130.12-081fd2e8",
"0?+1:+4") in new stack
-- Goto (callback,h,12)
-- Executing [h at callback:12] Playback("SIP/216.143.130.12-081fd2e8",
"trikon/register_cid") in new stack
== Spawn extension (callback, h, 12) exited non-zero on
'SIP/216.143.130.12-081fd2e8'
[Jun 16 18:31:21] NOTICE[2456]: cdr.c:432 ast_cdr_free: CDR on channel
'SIP/216.143.130.12-081fd2e8' not posted
It clearly says executing Playback, but actually line is disconnected
whereas it works perfectly in ast-1.4.4
Issue History
Date Modified Username Field Change
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06-16-08 07:55 binuvb Note Added: 0088749
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