Summary:ASTERISK-12205: Jerry Geis's dialplan 'works in 1.4, but not in 1.6' problem
Reporter:Steve Murphy (murf)Labels:
Date Opened:2008-07-23 16:20:10Date Closed:2008-07-23 16:56:23
Versions:Frequency of
Description:I am reporting this from Jerry Geiss's letter to asterisk-users on 21 July 2008
with subject 'what is the magic needed from upgrading from 1.4 to 1.6'.
(See thread starting at:

****** STEPS TO REPRODUCE ******

> On Mon, 2008-07-21 at 16:12 -0400, Jerry Geis wrote:
> >/ <------------>
> />/ ?[Jul 21 12:53:56] NOTICE[4881]: chan_sip.c:16416 handle_request_invite:
> /       Call from 'devcentos5x64_to_ebox4300' to extension
>       'mediaport_audio_visual' rejected because extension not found.
> Jerry--
> from the console, type "dialplan show smvoice-mediaport", and
> let's verify for certain that it's in there.
> I'll try to reproduce your problem in my test system here.
> murf

I get this:

dialplan show smvoice-mediaport
There is no existence of 'smvoice-mediaport' context
Command 'dialplan show smvoice-mediaport' failed.

my extensions.conf has a context:

; media
exten => public_address,1,Goto(smvoice-mediaport-public-address,s,1)

#include "/etc/asterisk/express.dnis.conf"

Then express.dnis.conf has:
; This file is generated from MessageNet EMACS
; Phone Caller ID & DNIS Manager screen

; MMAUDIO   : EBOX 4300                      -
exten => mediaport_audio_visual,1,Goto(smvoice-mediaport-audio-visual,s,1)

exten => s,1,Playback(beep)
exten => s,n,Dial(Console/dsp)
exten => s,n,Hangup

Not seeing what the problem is here. especially since 1.2 and 1.4 both work.
Comments:By: Digium Subversion (svnbot) 2008-07-23 16:56:15

Repository: asterisk
Revision: 133299

U   trunk/main/pbx.c

r133299 | murf | 2008-07-23 16:56:00 -0500 (Wed, 23 Jul 2008) | 27 lines

(closes issue ASTERISK-12205)
Reported by: murf
Tested by: murf
For: J. Geis

The 'data' field in the ast_exten struct was being
'moved' from the current dialplan to the replacement
dialplan. This was not good, as the current dialplan
could have problems in the time between the change
and when the new dialplan is swapped in.

So, I modified the merge_and_delete code to strdup
the 'data' field (the args to the app call), and
then it's freed as normal.

I improved a few messages; I added code to limit
the number of calls to the context_merge_incls_swits_igps_other_registrars()
to one per context. I don't think having it called
multiple times per context was doing anything bad,
but it was inefficient.

I hope this fixes the problems Mr. Geiss was noting in
asterisk-users, see