[asterisk-bugs] [JIRA] (ASTERISK-24839) CDRs: LinkedID field in Attended Transfer not reporting as expected
Matt Jordan (JIRA)
noreply at issues.asterisk.org
Fri Feb 27 21:41:34 CST 2015
[ https://issues.asterisk.org/jira/browse/ASTERISK-24839?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Matt Jordan updated ASTERISK-24839:
-----------------------------------
Summary: CDRs: LinkedID field in Attended Transfer not reporting as expected (was: CDR for Transfers not reporting as expected)
> CDRs: LinkedID field in Attended Transfer 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
> Assignee: Jonas Swiatek
> Severity: Critical
>
> 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:
> {noformat}
> [first]
> connection=asterisk-pgsql
> table=asterisk_cdr
> alias start => calldate
> {noformat}
> {{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:
> {noformat}
> [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()
> {noformat}
> This is my users configured at the end of sip.conf (rest of the file is unchanged except for allowing some IP ranges):
> {noformat}
> [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
> {noformat}
> 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:
> # A, dial B
> # Pickup B, press Transfer, dial 851#
> # Pickip C, hangup B. C and A are connected.
> # 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:
> {noformat}
> == 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'
> {noformat}
--
This message was sent by Atlassian JIRA
(v6.2#6252)
More information about the asterisk-bugs
mailing list