[asterisk-bugs] [JIRA] (ASTERISK-24839) CDR for Transfers not reporting as expected
Jonas Swiatek (JIRA)
noreply at issues.asterisk.org
Fri Feb 27 18:55:35 CST 2015
Jonas Swiatek created ASTERISK-24839:
----------------------------------------
Summary: CDR for Transfers not reporting as expected
Key: ASTERISK-24839
URL: https://issues.asterisk.org/jira/browse/ASTERISK-24839
Project: Asterisk
Issue Type: Bug
Security Level: None
Components: CDR/cdr_adaptive_odbc, CDR/General
Affects Versions: 13.2.0
Environment: Linux CentOS6 (Amazon Linux on AWS EC2)
Reporter: Jonas Swiatek
Severity: Blocker
Initial report made here: http://forums.asterisk.org/viewtopic.php?f=1&t=92599, and malcolmd has directed me to create this issue.
Asterisk doesn't generate CDR Data it should according to the Asterisk 12 CDR Specification for the "Attended Transfer to Channel"-scenario.
I'm using adaptive-odbc, connected to a PostgreSQL Database, with the following configuration:
[first]
connection=asterisk-pgsql
table=asterisk_cdr
alias start => calldate
cdr.conf is the default created when running make samples, but set to log both unanswered and congested calls.
This is my full configuration in extensions.conf:
[general]
static = yes
writeprotect = no
extenpatternmatchnew = yes ; Using new pattern matcher as a test
clearglobalvars = no
userscontext = default ; User context is where entries from users.conf are registered.
[account-shared-extensions-22575]
exten => 682,1,NoOp()
same => n,Set(CDR(userid)=9264)
same => n,Set(AccountID=22575)
same => n,Dial(SIP/swiatek,25)
same => n,Hangup()
exten => 850,1,NoOp()
same => n,Set(CDR(userid)=9574)
same => n,Set(AccountID=22575)
same => n,Dial(SIP/swiatek2,25)
same => n,Hangup()
exten => 851,1,NoOp()
same => n,Set(CDR(userid)=9598)
same => n,Set(AccountID=22575)
same => n,Dial(SIP/swiatek3,25)
same => n,Hangup()
[user-default-9264] ;swiatek
include => account-shared-extensions-22575
exten => *99,1,NoOp()
same => n,VoiceMailMain(swiatek at vm-22575,s)
same => n,Hangup()
[user-default-9574] ;swiatek2
include => account-shared-extensions-22575
exten => *99,1,NoOp()
same => n,VoiceMailMain(swiatek2 at vm-22575,s)
same => n,Hangup()
[user-default-9598] ;swiatek3
include => account-shared-extensions-22575
exten => *99,1,NoOp()
same => n,VoiceMailMain(swiatek3 at vm-22575,s)
same => n,Hangup()
This is my users configured at the end of sip.conf (rest of the file is unchanged except for allowing some IP ranges):
[user-template](!)
type = friend
host = dynamic
dtmfmode = auto
disallow = all
allow = ulaw
allow = alaw
;secret = testhest
context = outbound-init
mohsuggest = sound_1
alwaysauthreject=yes
[swiatek](user-template)
context=user-default-9264
callerid="Jonas Swiatek" <682>
mailbox=swiatek at vm-22575
accountcode=22575
mohsuggest=sound_1
[swiatek2](user-template)
context=user-default-9574
callerid="Jonas 2" <850>
mailbox=swiatek2 at vm-22575
accountcode=22575
[swiatek3](user-template)
context=user-default-9598
callerid="Jonas 3" <851>
mailbox=swiatek3 at vm-22575
accountcode=22575
Steps taken:
When testing this, I've got three phones (A(682), B(850) and C(851)) on my desk, one for each extension. I test is like this:
1) A, dial B
2) Pickup B, press Transfer, dial 851#
3) Pickip C, hangup B. C and A are connected.
4) Hangup A.
See the initial post (http://forums.asterisk.org/viewtopic.php?f=1&t=92599) for a formatted layout of the table data I'm getting, and what I think it should be when reading the CDR Spec for Asterisk 12 and later.
Recording log output from Asterisk during this test call:
== Using SIP RTP CoS mark 5
-- Executing [850 at user-default-9264:1] NoOp("SIP/swiatek-00000032", "") in new stack
-- Executing [850 at user-default-9264:2] Set("SIP/swiatek-00000032", "CDR(userid)=9574") in new stack
-- Executing [850 at user-default-9264:3] Set("SIP/swiatek-00000032", "AccountID=22575") in new stack
-- Executing [850 at user-default-9264:4] Dial("SIP/swiatek-00000032", "SIP/swiatek2,25") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/swiatek2
-- SIP/swiatek2-00000033 is ringing
-- SIP/swiatek2-00000033 answered SIP/swiatek-00000032
-- Channel SIP/swiatek-00000032 joined 'simple_bridge' basic-bridge <7910e64c-3f55-4153-b591-e092f1f61172>
-- Channel SIP/swiatek2-00000033 joined 'simple_bridge' basic-bridge <7910e64c-3f55-4153-b591-e092f1f61172>
-- Channel SIP/swiatek2-00000033 left 'native_rtp' basic-bridge <7910e64c-3f55-4153-b591-e092f1f61172>
-- Channel SIP/swiatek-00000032 left 'native_rtp' basic-bridge <7910e64c-3f55-4153-b591-e092f1f61172>
== Spawn extension (user-default-9264, 850, 4) exited non-zero on 'SIP/swiatek-00000032'
== Using SIP RTP CoS mark 5
-- Executing [850 at user-default-9264:1] NoOp("SIP/swiatek-00000034", "") in new stack
-- Executing [850 at user-default-9264:2] Set("SIP/swiatek-00000034", "CDR(userid)=9574") in new stack
-- Executing [850 at user-default-9264:3] Set("SIP/swiatek-00000034", "AccountID=22575") in new stack
-- Executing [850 at user-default-9264:4] Dial("SIP/swiatek-00000034", "SIP/swiatek2,25") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/swiatek2
-- SIP/swiatek2-00000035 is ringing
-- SIP/swiatek2-00000035 answered SIP/swiatek-00000034
-- Channel SIP/swiatek-00000034 joined 'simple_bridge' basic-bridge <8d1a7359-cda4-4b00-bd99-16227d0c05f8>
-- Channel SIP/swiatek2-00000035 joined 'simple_bridge' basic-bridge <8d1a7359-cda4-4b00-bd99-16227d0c05f8>
-- Music class sound_1 requested but no musiconhold loaded.
== Using SIP RTP CoS mark 5
-- Executing [851 at user-default-9574:1] NoOp("SIP/swiatek2-00000036", "") in new stack
-- Executing [851 at user-default-9574:2] Set("SIP/swiatek2-00000036", "CDR(userid)=9598") in new stack
-- Executing [851 at user-default-9574:3] Set("SIP/swiatek2-00000036", "AccountID=22575") in new stack
-- Executing [851 at user-default-9574:4] Dial("SIP/swiatek2-00000036", "SIP/swiatek3,25") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/swiatek3
-- SIP/swiatek3-00000037 is ringing
-- SIP/swiatek3-00000037 answered SIP/swiatek2-00000036
-- Channel SIP/swiatek2-00000036 joined 'simple_bridge' basic-bridge <2078600c-dd52-4005-8d67-4963b8b3dcf2>
-- Channel SIP/swiatek3-00000037 joined 'simple_bridge' basic-bridge <2078600c-dd52-4005-8d67-4963b8b3dcf2>
-- Channel SIP/swiatek-00000034 left 'native_rtp' basic-bridge <8d1a7359-cda4-4b00-bd99-16227d0c05f8>
-- Channel SIP/swiatek2-00000036 left 'native_rtp' basic-bridge <2078600c-dd52-4005-8d67-4963b8b3dcf2>
-- Channel SIP/swiatek-00000034 swapped with SIP/swiatek2-00000036 into 'native_rtp' basic-bridge <2078600c-dd52-4005-8d67-4963b8b3dcf2>
== Spawn extension (user-default-9574, 851, 4) exited non-zero on 'SIP/swiatek2-00000036'
-- Channel SIP/swiatek2-00000035 left 'simple_bridge' basic-bridge <8d1a7359-cda4-4b00-bd99-16227d0c05f8>
-- Channel SIP/swiatek3-00000037 left 'native_rtp' basic-bridge <2078600c-dd52-4005-8d67-4963b8b3dcf2>
-- Channel SIP/swiatek-00000034 left 'native_rtp' basic-bridge <2078600c-dd52-4005-8d67-4963b8b3dcf2>
== Spawn extension (user-default-9264, 850, 4) exited non-zero on 'SIP/swiatek-00000034'
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