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Summary:ASTERISK-16950: Asterisk not send fax to fax extension
Reporter:mbrevda (lazytt)Labels:
Date Opened:2010-11-12 05:16:04.000-0600Date Closed:2010-11-22 13:42:13.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Resources/res_fax
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) 0018299.log2.log
( 1) dialplan-fax.txt
Description:Asterisk isnt sending fax to fax extension:

== Redirecting 'SIP/69.167.68.134-00000036' to fax extension due to CNG detection
== Spawn extension (from-trunk, fax, 1) exited non-zero on 'SIP/69.167.68.134-00000036'


****** ADDITIONAL INFORMATION ******

Dialplan as follows:
-= 7 extensions (39 priorities) in 1 context. =-
moshe*CLI> dialplan show from-trunk
[ Context 'from-trunk' created by 'pbx_config' ]
 Include =>        'from-pstn'                                   [pbx_config]

-= 0 extensions (0 priorities) in 1 context. =-
moshe*CLI> dialplan show from-pstn
[ Context 'from-pstn' created by 'pbx_config' ]
 Include =>        'from-pstn-custom'                            [pbx_config]
 Include =>        'ext-did'                                     [pbx_config]
 Include =>        'ext-did-post-custom'                         [pbx_config]
 Include =>        'from-did-direct'                             [pbx_config]
 Include =>        'ext-did-catchall'                            [pbx_config]

-= 0 extensions (0 priorities) in 1 context. =-
moshe*CLI> dialplan show ext-did
[ Context 'ext-did' created by 'pbx_config' ]
 'foo' =>          1. Noop(bar)                                  [pbx_config]
 Include =>        'ext-did-custom'                              [pbx_config]
 Include =>        'ext-did-0001'                                [pbx_config]
 Include =>        'ext-did-0002'                                [pbx_config]

-= 1 extension (1 priority) in 1 context. =-
moshe*CLI> dialplan show ext-did-0002
[ Context 'ext-did-0002' created by 'pbx_config' ]
 '[REDACTED]' =>  1. Set(__FROM_DID=${EXTEN})                   [pbx_config]
                   2. ExecIf($[ "${CALLERID(name)}" = "" ] ?Set(CALLERID(name)=${CALLERID(num)})) [pbx_config]
                   3. Set(__CALLINGPRES_SV=${CALLERPRES()})      [pbx_config]
                   4. Set(CALLERPRES()=allowed_not_screened)     [pbx_config]
                   5. Set(_RGPREFIX=M: )                         [pbx_config]
                   6. Set(CALLERID(name)=${RGPREFIX}${CALLERID(name)}) [pbx_config]
                   7. Set(FAX_DEST=ext-faxt^9999^1)        [pbx_config]
                   8. Answer()                                   [pbx_config]
                   9. Wait(4)                                    [pbx_config]
    [dest-ext]     10. Goto(ext-group,600,1)                     [pbx_config]
 'fax' =>          1. Goto(${CUT(FAX_DEST,^,1)},${CUT(FAX_DEST,^,2)},${CUT(FAX_DEST,^,3)}) [pbx_config]
 Include =>        'ext-did-0002-custom'                         [pbx_config]

-= 7 extensions (39 priorities) in 1 context. =-

BTW, adding a fax extension right on top (in from-trunk) doesn't help
Comments:By: Paul Belanger (pabelanger) 2010-11-12 07:12:23.000-0600

Attach a debug log (see below) plus your extensions.conf.

---
We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue:

https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information
By: mbrevda (lazytt) 2010-11-14 00:34:06.000-0600

debug log attached
By: Jeremy Kister (jkister) 2010-11-14 00:52:24.000-0600

I also run into this issue; I posted on asterisk-users and asterisk-dev on 2010.11.13

I upgraded from a perfectly working 1.6.2 asterisk installation
(including fax via app_fax_digium) to 1.8.0 -- and fax functionality has
broken.

When a fax caller connects, asterisk detects the CNG, logs a message
about switching to the fax extension, but then hangs up the call.

i have:
 exten => fax,1,NoOp( got to fax extension )
 exten => fax,n,Goto(fax,rx,1)

I never receive the noop message.

i've captured with:
  core set verbose 10
  core set debug 10
  fax set debug on
  sip set debug peer vgw1

(vgw1 is my cisco 1760 ata)

http://jeremy.kister.net/tmp/fax/console.txt
http://jeremy.kister.net/tmp/fax/messages.txt
http://jeremy.kister.net/tmp/fax/sip.txt
By: Jeremy Kister (jkister) 2010-11-14 00:54:02.000-0600

sending the caller directly to ReceiveFax does work as expected; it's just the redirecting process that is broken.
By: Paul Belanger (pabelanger) 2010-11-14 07:50:12.000-0600

what is the results of:

*CLI> dialplan show fax@from-trunk

Also, upload a sample extensions.conf which reproduces the issue.
By: Jeremy Kister (jkister) 2010-11-14 10:51:59.000-0600

the most concise extensions.conf that reproduces it is:

[incoming]
exten => s,1,Answer
exten => s,n,Background(local/main)
exten => s,n,WaitExten(10)

exten => _XX,1,Dial(SIP/1${EXTEN})

exten => fax,1,NoOp( fax extension called )
By: Paul Belanger (pabelanger) 2010-11-14 21:17:49.000-0600

What version of 1.6.2 are you using?
By: Jeremy Kister (jkister) 2010-11-14 21:35:29.000-0600

nothing about 1.6.2 is important except that fax was working when i was on 1.6.2.14.

the problem is with 1.8.0.
By: Paul Belanger (pabelanger) 2010-11-14 22:59:09.000-0600

Odd, I must be doing something wrong on my side.  I cannot even get it working under 1.6.2.
By: Paul Belanger (pabelanger) 2010-11-14 23:01:50.000-0600

I've also created an automated test located at:
http://svn.asterisk.org/svn/testsuite/asterisk/team/pabelanger/faxdetect/
By: mbrevda (lazytt) 2010-11-14 23:39:32.000-0600

As per the original ticket details, the issue is with 1.8.0 - the same dialplan works fine on 1.4/1.6.x. What are your test results?
By: Jeremy Kister (jkister) 2010-11-15 09:41:46.000-0600

getting fax to work on 1.6.2 is rather easy.

make sure you /don't/ have app_fax loaded and /do/ have res_fax and res_fax_(spandsp|digium) loaded

in the sip.conf general, make sure you have faxdetect=yes & t38pt_udptl = yes


im near sure that's all you need.
By: konstk (konstk) 2010-11-18 08:19:56.000-0600

i had the same issue, it was same problem as in 0018185 solved with the same patch.
By: Jeremy Kister (jkister) 2010-11-18 08:32:31.000-0600

the patch in 0018185 has solved the fax transfer issue in my case for asterisk 1.8.0
By: Digium Subversion (svnbot) 2010-11-22 12:46:42.000-0600

Repository: asterisk
Revision: 295790

U   branches/1.4/apps/app_macro.c
U   branches/1.4/include/asterisk/channel.h
U   branches/1.4/include/asterisk/frame.h
U   branches/1.4/main/channel.c
U   branches/1.4/main/pbx.c

------------------------------------------------------------------------
r295790 | rmudgett | 2010-11-22 12:46:27 -0600 (Mon, 22 Nov 2010) | 46 lines

The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.

To recreate the problem:
1) Party A calls Party B
2) Invoke CLI "channel redirect" command to redirect channel call leg
associated with A.
3) All associated channels are hung up.

Note that if the CLI command were done on the channel call leg associated
with B it works.

This regression was a result of the fix for issue ASTERISK-15731
(https://reviewboard.asterisk.org/r/740/).

The regression affects all features that use an async goto to execute the
dialplan because of an external event: Channel redirect, AMI redirect, SIP
REFER, and FAX detection.

The struct ast_channel._softhangup code is a mess.  The variable is used
for several purposes that do not necessarily result in the call being hung
up.  I have added doxygen comments to describe how the various _softhangup
bits are used.  I have corrected all the places where the variable was
tested in a non-bit oriented manner.

The primary fix is the new AST_CONTROL_END_OF_Q frame.  It acts as a weak
hangup request so the soft hangup requests that do not normally result in
a hangup do not hangup.

JIRA SWP-2470
JIRA SWP-2489

(closes issue ASTERISK-16838)
Reported by: SantaFox
(closes issue ASTERISK-16847)
Reported by: kwemheuer
(closes issue ASTERISK-16873)
Reported by: zahir_koradia
(closes issue ASTERISK-16891)
Reported by: vmarrone
(closes issue ASTERISK-16950)
Reported by: mbrevda
(closes issue ASTERISK-16972)
Reported by: nerbos

Review: https://reviewboard.asterisk.org/r/1013/

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=295790
By: Digium Subversion (svnbot) 2010-11-22 13:28:36.000-0600

Repository: asterisk
Revision: 295843

_U  branches/1.6.2/
U   branches/1.6.2/apps/app_macro.c
U   branches/1.6.2/include/asterisk/channel.h
U   branches/1.6.2/include/asterisk/frame.h
U   branches/1.6.2/main/channel.c
U   branches/1.6.2/main/pbx.c

------------------------------------------------------------------------
r295843 | rmudgett | 2010-11-22 13:28:27 -0600 (Mon, 22 Nov 2010) | 53 lines

Merged revisions 295790 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
 r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines
 
 The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
 
 To recreate the problem:
 1) Party A calls Party B
 2) Invoke CLI "channel redirect" command to redirect channel call leg
 associated with A.
 3) All associated channels are hung up.
 
 Note that if the CLI command were done on the channel call leg associated
 with B it works.
 
 This regression was a result of the fix for issue ASTERISK-15731
 (https://reviewboard.asterisk.org/r/740/).
 
 The regression affects all features that use an async goto to execute the
 dialplan because of an external event: Channel redirect, AMI redirect, SIP
 REFER, and FAX detection.
 
 The struct ast_channel._softhangup code is a mess.  The variable is used
 for several purposes that do not necessarily result in the call being hung
 up.  I have added doxygen comments to describe how the various _softhangup
 bits are used.  I have corrected all the places where the variable was
 tested in a non-bit oriented manner.
 
 The primary fix is the new AST_CONTROL_END_OF_Q frame.  It acts as a weak
 hangup request so the soft hangup requests that do not normally result in
 a hangup do not hangup.
 
 JIRA SWP-2470
 JIRA SWP-2489
 
 (closes issue ASTERISK-16838)
 Reported by: SantaFox
 (closes issue ASTERISK-16847)
 Reported by: kwemheuer
 (closes issue ASTERISK-16873)
 Reported by: zahir_koradia
 (closes issue ASTERISK-16891)
 Reported by: vmarrone
 (closes issue ASTERISK-16950)
 Reported by: mbrevda
 (closes issue ASTERISK-16972)
 Reported by: nerbos
 
 Review: https://reviewboard.asterisk.org/r/1013/
........

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=295843
By: Digium Subversion (svnbot) 2010-11-22 13:36:26.000-0600

Repository: asterisk
Revision: 295866

_U  branches/1.8/
U   branches/1.8/apps/app_macro.c
U   branches/1.8/include/asterisk/channel.h
U   branches/1.8/include/asterisk/frame.h
U   branches/1.8/main/channel.c
U   branches/1.8/main/pbx.c

------------------------------------------------------------------------
r295866 | rmudgett | 2010-11-22 13:36:11 -0600 (Mon, 22 Nov 2010) | 60 lines

Merged revisions 295843 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2

................
 r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines
 
 Merged revisions 295790 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
   r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines
   
   The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
   
   To recreate the problem:
   1) Party A calls Party B
   2) Invoke CLI "channel redirect" command to redirect channel call leg
   associated with A.
   3) All associated channels are hung up.
   
   Note that if the CLI command were done on the channel call leg associated
   with B it works.
   
   This regression was a result of the fix for issue ASTERISK-15731
   (https://reviewboard.asterisk.org/r/740/).
   
   The regression affects all features that use an async goto to execute the
   dialplan because of an external event: Channel redirect, AMI redirect, SIP
   REFER, and FAX detection.
   
   The struct ast_channel._softhangup code is a mess.  The variable is used
   for several purposes that do not necessarily result in the call being hung
   up.  I have added doxygen comments to describe how the various _softhangup
   bits are used.  I have corrected all the places where the variable was
   tested in a non-bit oriented manner.
   
   The primary fix is the new AST_CONTROL_END_OF_Q frame.  It acts as a weak
   hangup request so the soft hangup requests that do not normally result in
   a hangup do not hangup.
   
   JIRA SWP-2470
   JIRA SWP-2489
   
   (closes issue ASTERISK-16838)
   Reported by: SantaFox
   (closes issue ASTERISK-16847)
   Reported by: kwemheuer
   (closes issue ASTERISK-16873)
   Reported by: zahir_koradia
   (closes issue ASTERISK-16891)
   Reported by: vmarrone
   (closes issue ASTERISK-16950)
   Reported by: mbrevda
   (closes issue ASTERISK-16972)
   Reported by: nerbos
   
   Review: https://reviewboard.asterisk.org/r/1013/
 ........
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=295866
By: Digium Subversion (svnbot) 2010-11-22 13:42:13.000-0600

Repository: asterisk
Revision: 295867

_U  trunk/
U   trunk/apps/app_macro.c
U   trunk/include/asterisk/channel.h
U   trunk/include/asterisk/frame.h
U   trunk/main/channel.c
U   trunk/main/pbx.c

------------------------------------------------------------------------
r295867 | rmudgett | 2010-11-22 13:42:03 -0600 (Mon, 22 Nov 2010) | 67 lines

Merged revisions 295866 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
 r295866 | rmudgett | 2010-11-22 13:36:10 -0600 (Mon, 22 Nov 2010) | 60 lines
 
 Merged revisions 295843 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.6.2
 
 ................
   r295843 | rmudgett | 2010-11-22 13:28:23 -0600 (Mon, 22 Nov 2010) | 53 lines
   
   Merged revisions 295790 via svnmerge from
   https://origsvn.digium.com/svn/asterisk/branches/1.4
   
   ........
     r295790 | rmudgett | 2010-11-22 12:46:26 -0600 (Mon, 22 Nov 2010) | 46 lines
     
     The channel redirect function (CLI or AMI) hangs up the call instead of redirecting the call.
     
     To recreate the problem:
     1) Party A calls Party B
     2) Invoke CLI "channel redirect" command to redirect channel call leg
     associated with A.
     3) All associated channels are hung up.
     
     Note that if the CLI command were done on the channel call leg associated
     with B it works.
     
     This regression was a result of the fix for issue ASTERISK-15731
     (https://reviewboard.asterisk.org/r/740/).
     
     The regression affects all features that use an async goto to execute the
     dialplan because of an external event: Channel redirect, AMI redirect, SIP
     REFER, and FAX detection.
     
     The struct ast_channel._softhangup code is a mess.  The variable is used
     for several purposes that do not necessarily result in the call being hung
     up.  I have added doxygen comments to describe how the various _softhangup
     bits are used.  I have corrected all the places where the variable was
     tested in a non-bit oriented manner.
     
     The primary fix is the new AST_CONTROL_END_OF_Q frame.  It acts as a weak
     hangup request so the soft hangup requests that do not normally result in
     a hangup do not hangup.
     
     JIRA SWP-2470
     JIRA SWP-2489
     
     (closes issue ASTERISK-16838)
     Reported by: SantaFox
     (closes issue ASTERISK-16847)
     Reported by: kwemheuer
     (closes issue ASTERISK-16873)
     Reported by: zahir_koradia
     (closes issue ASTERISK-16891)
     Reported by: vmarrone
     (closes issue ASTERISK-16950)
     Reported by: mbrevda
     (closes issue ASTERISK-16972)
     Reported by: nerbos
     
     Review: https://reviewboard.asterisk.org/r/1013/
   ........
 ................
................

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=295867