[asterisk-bugs] [Asterisk 0018379]: attended transfer weird behaviour

Asterisk Bug Tracker noreply at bugs.digium.com
Wed Dec 29 08:23:09 UTC 2010


A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18379 
====================================================================== 
Reported By:                gincantalupo
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18379
Category:                   Applications/app_dial
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.8.1-rc1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-11-25 11:35 CST
Last Modified:              2010-12-29 02:23 CST
====================================================================== 
Summary:                    attended transfer weird behaviour
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

======================================================================
Relationships       ID      Summary
----------------------------------------------------------------------
related to          0018254 Attended transfer failure
related to          0017999 Issues with DTMF triggered attended tra...
====================================================================== 

---------------------------------------------------------------------- 
 (0130024) Docent (reporter) - 2010-12-29 02:23
 https://issues.asterisk.org/view.php?id=18379#c130024 
---------------------------------------------------------------------- 
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. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-12-29 02:23 Docent         Note Added: 0130024                          
======================================================================




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