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Summary:ASTERISK-17026: attended transfer weird behaviour
Reporter:Giorgio Incantalupo (gincantalupo)Labels:
Date Opened:2010-11-25 11:35:53.000-0600Date Closed:2011-01-19 15:35:30.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Applications/app_dial
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) log.tar.gz
Description:Just installed 1.8.1-rc1 and tried the attended transfer function with 3 snoms (firmware 8.x), A,B and C. When A calls B and B transfers to C but C is busy or does not answer, 'pbx-invalid.gsm' sound is played...but the called number is right!

Another test: when I try to transfer the call to a wrong number I get this message:
WARNING[31448]: features.c:1861 builtin_atxfer: Did not read data
and after that the call is bounced back to the transferrer (shouldn't Asterisk say invalid extension???)

My test extensions:
exten => 12,1,Dial(SIP/81,5,tT)
exten => 12,2,NoOp(${DIALSTATUS})
exten => 12,3,Hangup

exten => 14,1,Dial(SIP/8,5,tT)
exten => 14,2,NoOp(${DIALSTATUS})
exten => 14,3,Hangup

exten => 17,1,Dial(SIP/70,5,tT)
exten => 17,2,NoOp(${DIALSTATUS})
exten => 17,3,Hangup


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

I do not know if is something related to ticket ASTERISK-1800254, seems similar, not the same. Sorry for eventual wrong report.
Comments:By: Leif Madsen (lmadsen) 2010-12-07 11:13:46.000-0600

Please provide a SIP trace, console trace (with debug level logging) and a trace with SIP history enabled in sip.conf.

Without that information it is going to be difficult (if not impossible) to reproduce this without your hardware.
By: Giorgio Incantalupo (gincantalupo) 2010-12-13 03:47:33.000-0600

Installed Asterisk 1.8.1 instead of 1.8.1-rc1 hoping the behaviour has changed but nothing changed so the log is referring to version 1.8.1.
Test is: exten 12 calling exten 14 who transfers to exten 17.
By: Kun Liu (knkcn) 2010-12-13 22:07:13.000-0600

I think you second test lacks some exten. As my understand, if the transfer input cann't match any exten under the dialplan, the transfer function will think nothing was inpu. That is why there is "WARNING[31448]: features.c:1861 builtin_atxfer: Did not read data" even you pressed some key.

I believe if you add some exten like "exten => _X.,1,NOOP" you will get "pbx-invalid.gsm" played
By: Giorgio Incantalupo (gincantalupo) 2010-12-14 04:43:24.000-0600

Yes, knkcn, you are right, it works as you say but if you call a non-existent exten, Asterisk prints out a NOTICE while if you make a transfer, you have a different behaviour: you need to have _X. extension in your dialplan to get a pbx-invalid message. Isn't it weird? Isn't transferring a call a sort of "second call"? Shouldn't it pass thru the same dialplan (the context is the same)?
By: Michael Shepelev (docent) 2010-12-29 02:23:08.000-0600

I have the same problem sometimes, but I get this message while dialing exten:
[2010-12-29 10:08:27] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '*' received on SIP/111-000062c0
[2010-12-29 10:08:27] DTMF[21357]: channel.c:3931 __ast_read: DTMF begin passthrough '*' on SIP/111-000062c0
[2010-12-29 10:08:28] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '*' received on SIP/111-000062c0, duration 220 ms
[2010-12-29 10:08:28] DTMF[21357]: channel.c:3876 __ast_read: DTMF end accepted with begin '*' on SIP/111-000062c0
[2010-12-29 10:08:28] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '*' on SIP/111-000062c0
[2010-12-29 10:08:28] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '*' received on SIP/111-000062c0
[2010-12-29 10:08:28] DTMF[21357]: channel.c:3931 __ast_read: DTMF begin passthrough '*' on SIP/111-000062c0
[2010-12-29 10:08:28] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '*' received on SIP/111-000062c0, duration 140 ms
[2010-12-29 10:08:28] DTMF[21357]: channel.c:3876 __ast_read: DTMF end accepted with begin '*' on SIP/111-000062c0
[2010-12-29 10:08:28] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '*' on SIP/111-000062c0
[2010-12-29 10:08:28]     -- Started music on hold, class 'default', on SIP/chelngn54-000062b2
[2010-12-29 10:08:28]     -- <SIP/111-000062c0> Playing 'pbx-transfer.gsm' (language 'ru')
[2010-12-29 10:08:28] WARNING[31437]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 49443304
[2010-12-29 10:08:29] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '1' received on SIP/111-000062c0
[2010-12-29 10:08:29] DTMF[21357]: channel.c:3925 __ast_read: DTMF begin ignored '1' on SIP/111-000062c0
[2010-12-29 10:08:29] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '1' received on SIP/111-000062c0, duration 100 ms
[2010-12-29 10:08:29] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '1' on SIP/111-000062c0
[2010-12-29 10:08:29] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '3' received on SIP/111-000062c0
[2010-12-29 10:08:29] DTMF[21357]: channel.c:3925 __ast_read: DTMF begin ignored '3' on SIP/111-000062c0
[2010-12-29 10:08:30] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '3' received on SIP/111-000062c0, duration 120 ms
[2010-12-29 10:08:30] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '3' on SIP/111-000062c0
[2010-12-29 10:08:30] WARNING[21357]: features.c:1861 builtin_atxfer: Did not read data.
[2010-12-29 10:08:30]     -- Stopped music on hold on SIP/chelngn54-000062b2
[2010-12-29 10:08:30]     -- <SIP/111-000062c0> Playing 'beeperr.gsm' (language 'ru')
[2010-12-29 10:08:30] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '5' received on SIP/111-000062c0
[2010-12-29 10:08:30] DTMF[21357]: channel.c:3925 __ast_read: DTMF begin ignored '5' on SIP/111-000062c0
[2010-12-29 10:08:30] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '5' received on SIP/111-000062c0, duration 120 ms
[2010-12-29 10:08:30] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '5' on SIP/111-000062c0
[2010-12-29 10:08:32] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '*' received on SIP/111-000062c0
[2010-12-29 10:08:32] DTMF[21357]: channel.c:3931 __ast_read: DTMF begin passthrough '*' on SIP/111-000062c0
[2010-12-29 10:08:32] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '*' received on SIP/111-000062c0, duration 140 ms
[2010-12-29 10:08:32] DTMF[21357]: channel.c:3876 __ast_read: DTMF end accepted with begin '*' on SIP/111-000062c0
[2010-12-29 10:08:32] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '*' on SIP/111-000062c0
[2010-12-29 10:08:32] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '*' received on SIP/111-000062c0
[2010-12-29 10:08:32] DTMF[21357]: channel.c:3931 __ast_read: DTMF begin passthrough '*' on SIP/111-000062c0
[2010-12-29 10:08:33] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '*' received on SIP/111-000062c0, duration 140 ms
[2010-12-29 10:08:33] DTMF[21357]: channel.c:3876 __ast_read: DTMF end accepted with begin '*' on SIP/111-000062c0
[2010-12-29 10:08:33] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '*' on SIP/111-000062c0
[2010-12-29 10:08:33]     -- Started music on hold, class 'default', on SIP/chelngn54-000062b2
[2010-12-29 10:08:33]     -- <SIP/111-000062c0> Playing 'pbx-transfer.gsm' (language 'ru')
[2010-12-29 10:08:33] WARNING[31437]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 49443304
[2010-12-29 10:08:34] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '1' received on SIP/111-000062c0
[2010-12-29 10:08:34] DTMF[21357]: channel.c:3925 __ast_read: DTMF begin ignored '1' on SIP/111-000062c0
[2010-12-29 10:08:34] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '1' received on SIP/111-000062c0, duration 120 ms
[2010-12-29 10:08:34] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '1' on SIP/111-000062c0
[2010-12-29 10:08:34] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '3' received on SIP/111-000062c0
[2010-12-29 10:08:34] DTMF[21357]: channel.c:3925 __ast_read: DTMF begin ignored '3' on SIP/111-000062c0
[2010-12-29 10:08:34] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '3' received on SIP/111-000062c0, duration 120 ms
[2010-12-29 10:08:34] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '3' on SIP/111-000062c0
[2010-12-29 10:08:34] WARNING[21357]: features.c:1861 builtin_atxfer: Did not read data.
[2010-12-29 10:08:34]     -- Stopped music on hold on SIP/chelngn54-000062b2
[2010-12-29 10:08:34]     -- <SIP/111-000062c0> Playing 'beeperr.gsm' (language 'ru')
[2010-12-29 10:08:35] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '5' received on SIP/111-000062c0
[2010-12-29 10:08:35] DTMF[21357]: channel.c:3925 __ast_read: DTMF begin ignored '5' on SIP/111-000062c0
[2010-12-29 10:08:35] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '5' received on SIP/111-000062c0, duration 100 ms
[2010-12-29 10:08:35] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '5' on SIP/111-000062c0
[2010-12-29 10:08:39] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '*' received on SIP/111-000062c0
[2010-12-29 10:08:39] DTMF[21357]: channel.c:3931 __ast_read: DTMF begin passthrough '*' on SIP/111-000062c0
[2010-12-29 10:08:39] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '*' received on SIP/111-000062c0, duration 140 ms
[2010-12-29 10:08:39] DTMF[21357]: channel.c:3876 __ast_read: DTMF end accepted with begin '*' on SIP/111-000062c0
[2010-12-29 10:08:39] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '*' on SIP/111-000062c0
[2010-12-29 10:08:39] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '*' received on SIP/111-000062c0
[2010-12-29 10:08:39] DTMF[21357]: channel.c:3931 __ast_read: DTMF begin passthrough '*' on SIP/111-000062c0
[2010-12-29 10:08:39] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '*' received on SIP/111-000062c0, duration 200 ms
[2010-12-29 10:08:39] DTMF[21357]: channel.c:3876 __ast_read: DTMF end accepted with begin '*' on SIP/111-000062c0
[2010-12-29 10:08:39] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '*' on SIP/111-000062c0
[2010-12-29 10:08:39]     -- Started music on hold, class 'default', on SIP/chelngn54-000062b2
[2010-12-29 10:08:39]     -- <SIP/111-000062c0> Playing 'pbx-transfer.gsm' (language 'ru')
[2010-12-29 10:08:39] WARNING[31437]: mp3/interface.c:216 decodeMP3: Junk at the beginning of frame 49443304
[2010-12-29 10:08:41] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '1' received on SIP/111-000062c0
[2010-12-29 10:08:41] DTMF[21357]: channel.c:3925 __ast_read: DTMF begin ignored '1' on SIP/111-000062c0
[2010-12-29 10:08:41] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '1' received on SIP/111-000062c0, duration 200 ms
[2010-12-29 10:08:41] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '1' on SIP/111-000062c0
[2010-12-29 10:08:41] DTMF[21357]: channel.c:3921 __ast_read: DTMF begin '3' received on SIP/111-000062c0
[2010-12-29 10:08:41] DTMF[21357]: channel.c:3925 __ast_read: DTMF begin ignored '3' on SIP/111-000062c0
[2010-12-29 10:08:41] DTMF[21357]: channel.c:3836 __ast_read: DTMF end '3' received on SIP/111-000062c0, duration 160 ms
[2010-12-29 10:08:41] DTMF[21357]: channel.c:3905 __ast_read: DTMF end passthrough '3' on SIP/111-000062c0
[2010-12-29 10:08:41] WARNING[21357]: features.c:1861 builtin_atxfer: Did not read data.
By: Leif Madsen (lmadsen) 2011-01-19 10:34:13.000-0600

Please test the latest 1.8 branch (svn co http://svn.asterisk.org/svn/asterisk/branches/1.8) because some work has been done here to resolve these types of transfers.
By: Digium Subversion (svnbot) 2011-01-19 15:21:58.000-0600

Repository: asterisk
Revision: 302671

U   branches/1.4/res/res_features.c

------------------------------------------------------------------------
r302671 | rmudgett | 2011-01-19 15:21:57 -0600 (Wed, 19 Jan 2011) | 15 lines

DTMF transfer plays the wrong sounds for wrong number or other call failure.

* Set the default for features.conf.sample xferfailsound option to "beeperr"
as documented instead of "pbx-invalid" and corrected the use of it in DTMF
blind transfer (#1).

* Improved DTMF blind transfer handling of wrong numbers.

Most of the concerns in this issue were taken care of by the patch for
issue 17999: Issues with DTMF triggered attended transfers.

(closes issue ASTERISK-17026)
Reported by: gincantalupo
Tested by: rmudgett

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

http://svn.digium.com/view/asterisk?view=rev&revision=302671
By: Digium Subversion (svnbot) 2011-01-19 15:25:42.000-0600

Repository: asterisk
Revision: 302693

_U  branches/1.6.2/
U   branches/1.6.2/main/features.c

------------------------------------------------------------------------
r302693 | rmudgett | 2011-01-19 15:25:42 -0600 (Wed, 19 Jan 2011) | 22 lines

Merged revisions 302671 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
 r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) | 15 lines
 
 DTMF transfer plays the wrong sounds for wrong number or other call failure.
 
 * Set the default for features.conf.sample xferfailsound option to "beeperr"
 as documented instead of "pbx-invalid" and corrected the use of it in DTMF
 blind transfer (#1).
 
 * Improved DTMF blind transfer handling of wrong numbers.
 
 Most of the concerns in this issue were taken care of by the patch for
 issue 17999: Issues with DTMF triggered attended transfers.
 
 (closes issue ASTERISK-17026)
 Reported by: gincantalupo
 Tested by: rmudgett
........

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

http://svn.digium.com/view/asterisk?view=rev&revision=302693
By: Digium Subversion (svnbot) 2011-01-19 15:29:23.000-0600

Repository: asterisk
Revision: 302713

_U  branches/1.8/
U   branches/1.8/main/features.c

------------------------------------------------------------------------
r302713 | rmudgett | 2011-01-19 15:29:23 -0600 (Wed, 19 Jan 2011) | 29 lines

Merged revisions 302693 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.6.2

................
 r302693 | rmudgett | 2011-01-19 15:25:41 -0600 (Wed, 19 Jan 2011) | 22 lines
 
 Merged revisions 302671 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.4
 
 ........
   r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) | 15 lines
   
   DTMF transfer plays the wrong sounds for wrong number or other call failure.
   
   * Set the default for features.conf.sample xferfailsound option to "beeperr"
   as documented instead of "pbx-invalid" and corrected the use of it in DTMF
   blind transfer (#1).
   
   * Improved DTMF blind transfer handling of wrong numbers.
   
   Most of the concerns in this issue were taken care of by the patch for
   issue 17999: Issues with DTMF triggered attended transfers.
   
   (closes issue ASTERISK-17026)
   Reported by: gincantalupo
   Tested by: rmudgett
 ........
................

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

http://svn.digium.com/view/asterisk?view=rev&revision=302713
By: Digium Subversion (svnbot) 2011-01-19 15:35:29.000-0600

Repository: asterisk
Revision: 302732

_U  trunk/
U   trunk/main/features.c

------------------------------------------------------------------------
r302732 | rmudgett | 2011-01-19 15:35:29 -0600 (Wed, 19 Jan 2011) | 36 lines

Merged revisions 302713 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.8

................
 r302713 | rmudgett | 2011-01-19 15:29:22 -0600 (Wed, 19 Jan 2011) | 29 lines
 
 Merged revisions 302693 via svnmerge from
 https://origsvn.digium.com/svn/asterisk/branches/1.6.2
 
 ................
   r302693 | rmudgett | 2011-01-19 15:25:41 -0600 (Wed, 19 Jan 2011) | 22 lines
   
   Merged revisions 302671 via svnmerge from
   https://origsvn.digium.com/svn/asterisk/branches/1.4
   
   ........
     r302671 | rmudgett | 2011-01-19 15:21:56 -0600 (Wed, 19 Jan 2011) | 15 lines
     
     DTMF transfer plays the wrong sounds for wrong number or other call failure.
     
     * Set the default for features.conf.sample xferfailsound option to "beeperr"
     as documented instead of "pbx-invalid" and corrected the use of it in DTMF
     blind transfer (#1).
     
     * Improved DTMF blind transfer handling of wrong numbers.
     
     Most of the concerns in this issue were taken care of by the patch for
     issue 17999: Issues with DTMF triggered attended transfers.
     
     (closes issue ASTERISK-17026)
     Reported by: gincantalupo
     Tested by: rmudgett
   ........
 ................
................

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

http://svn.digium.com/view/asterisk?view=rev&revision=302732