[asterisk-bugs] [JIRA] (ASTERISK-24620) AGI GET VARIABLE ANSWEREDTIME gives ZERO on callback bridge using app

Rusty Newton (JIRA) noreply at issues.asterisk.org
Mon Dec 15 09:49:29 CST 2014


     [ https://issues.asterisk.org/jira/browse/ASTERISK-24620?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-24620:
------------------------------------

    Description: 
Below is the log of Asterisk Console during Callback request by 6012XXXXXXX
a) when call SIP/sipprovider/6012XXXXXXX is connected
b) Asterisk is dialing another sip channel SIP/ims_peer/60327XXXXXX
c) call duriation on SIP/ims_peer/60327XXXXXX is 00:00:12
d) GET VARIABLE ANSWEREDTIME report Wrong : 0
e) GET VARIABLE DIALEDTIME report Wrong : 28 

[Edit by Rusty - Remove inline debug - logs don't go in the description field. In the future, please use attachments for logs.]

  was:
Below is the log of Asterisk Console during Callback request by 6012XXXXXXX
a) when call SIP/sipprovider/6012XXXXXXX is connected
b) Asterisk is dialing another sip channel SIP/ims_peer/60327XXXXXX
c) call duriation on SIP/ims_peer/60327XXXXXX is 00:00:12
d) GET VARIABLE ANSWEREDTIME report Wrong : 0
e) GET VARIABLE DIALEDTIME report Wrong : 28 


  == Manager 'myasterisk' logged on from 127.0.0.1
  == Using SIP RTP TOS bits 8
  == Using SIP RTP CoS mark 5
    -- Called sipprovider/6012XXXXXXX
    -- SIP/sipprovider-00000043 is ringing
    -- SIP/sipprovider-00000043 is making progress
    -- SIP/sipprovider-00000043 answered
    -- Executing [60327XXXXXX at a2billing-callback:1] Answer("SIP/sipprovider-00000043", "") in new stack
    -- Executing [60327XXXXXX at a2billing-callback:2] Wait("SIP/sipprovider-00000043", "1") in new stack
    -- Executing [60327XXXXXX at a2billing-callback:3] AGI("SIP/sipprovider-00000043", "a2billing.php,2,callback") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/a2billing.php
 a2billing.php,2,callback: file:Class.RateEngine.php - line:1076 - app_callingcard: Dialing 'SIP/ims_peer/27XXXXXX,90,HgiL(2147483647:60000:30000)D(:1)' with timeout of '10243860'.
 a2billing.php,2,callback:
    -- AGI Script Executing Application: (DIAL) Options: (SIP/ims_peer/27XXXXXX,90,HgiL(2147483647:60000:30000)D(:1))
       > Limit Data for this call:
       > timelimit      = 2147483647 ms (2147483.647 s)
       > play_warning   = 60000 ms (60.000 s)
       > play_to_caller = yes
       > play_to_callee = no
       > warning_freq   = 30000 ms (30.000 s)
       > start_sound    =
       > warning_sound  = timeleft
       > end_sound      =
  == Using SIP RTP TOS bits 8
  == Using SIP RTP CoS mark 5
    -- Called SIP/ims_peer/27XXXXXX
    -- SIP/ims_peer-00000044 is ringing
    -- SIP/ims_peer-00000044 is making progress passing it to SIP/sipprovider-00000043
       > 0x7ffa7800f9d0 -- Probation passed - setting RTP source address to 10.1.2.131:14436
    -- SIP/ims_peer-00000044 answered SIP/sipprovider-00000043
    -- Sending DTMF '1' to the calling party.
    -- Channel SIP/sipprovider-00000043 joined 'simple_bridge' basic-bridge <390f5ee2-aff4-4d87-9b89-647c8c8728a6>
    -- Channel SIP/ims_peer-00000044 joined 'simple_bridge' basic-bridge <390f5ee2-aff4-4d87-9b89-647c8c8728a6>
callback01my*CLI> core show channels verbose
Channel              Context              Extension        Prio State   Application  Data                      CallerID        Duration Accountcode PeerAccount BridgeID
SIP/ims_peer-000000 from-unifi                               1 Up      AppDial      (Outgoing Line)           60327XXXXXX     00:00:12 9578581306  9578581306  390f5ee2-aff4-4d87-9
SIP/sipprovider-00000043   a2billing-callback   60327XXXXXX         3 Up      Dial         SIP/ims_peer/27XXXXXX,90 +6012XXXXXXX    00:00:35 9578581306  9578581306  390f5ee2-aff4-4d87-9
2 active channels
1 active call
30 calls processed
    -- Channel SIP/sipprovider-00000043 left 'simple_bridge' basic-bridge <390f5ee2-aff4-4d87-9b89-647c8c8728a6>
    -- Channel SIP/ims_peer-00000044 left 'simple_bridge' basic-bridge <390f5ee2-aff4-4d87-9b89-647c8c8728a6>
    -- ast_channel_answertime(chan): 1970-01-01 07:30:00
    -- ast_channel_get_up_time(1946159384): 0
    -- Successfully setting ANSWEREDTIME : 0
 <SIP/sipprovider-00000045>AGI Tx >> 200 result=1
<SIP/sipprovider-00000045>AGI Rx << GET VARIABLE ANSWEREDTIME
<SIP/sipprovider-00000045>AGI Tx >> 200 result=1 (0)
<SIP/sipprovider-00000045>AGI Rx << GET VARIABLE DIALEDTIME
<SIP/sipprovider-00000045>AGI Tx >> 200 result=1 (28)
<SIP/sipprovider-00000045>AGI Rx << GET VARIABLE CDR(billsec)
<SIP/sipprovider-00000045>AGI Tx >> 200 result=1 (21)
<SIP/sipprovider-00000045>AGI Rx << GET VARIABLE CDR(start)
<SIP/sipprovider-00000045>AGI Tx >> 200 result=1 (2014-12-15 21:58:26)
<SIP/sipprovider-00000045>AGI Rx << GET VARIABLE CDR(end)
<SIP/sipprovider-00000045>AGI Tx >> 200 result=1 (2014-12-15 21:58:48)
<SIP/sipprovider-00000045>AGI Rx << GET VARIABLE CDR(channel)
<SIP/sipprovider-00000045>AGI Tx >> 200 result=1 (SIP/sipprovider-00000045)
<SIP/sipprovider-00000045>AGI Rx << GET VARIABLE CDR(answer)
<SIP/sipprovider-00000045>AGI Tx >> 200 result=1 (2014-12-15 21:58:26)
<SIP/sipprovider-00000045>AGI Rx << GET VARIABLE AVAILSTATUS
<SIP/sipprovider-00000045>AGI Tx >> 200 result=0
<SIP/sipprovider-00000045>AGI Rx << GET VARIABLE DIALEDPEERNAME
<SIP/sipprovider-00000045>AGI Tx >> 200 result=1 (SIP/ims_peer-00000046)



> AGI GET VARIABLE ANSWEREDTIME gives ZERO on callback bridge using app
> ---------------------------------------------------------------------
>
>                 Key: ASTERISK-24620
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24620
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_dial
>    Affects Versions: 13.0.2
>         Environment: CentOS 6.5 64bit
>            Reporter: Sid Mason
>         Attachments: debug.txt
>
>
> Below is the log of Asterisk Console during Callback request by 6012XXXXXXX
> a) when call SIP/sipprovider/6012XXXXXXX is connected
> b) Asterisk is dialing another sip channel SIP/ims_peer/60327XXXXXX
> c) call duriation on SIP/ims_peer/60327XXXXXX is 00:00:12
> d) GET VARIABLE ANSWEREDTIME report Wrong : 0
> e) GET VARIABLE DIALEDTIME report Wrong : 28 
> [Edit by Rusty - Remove inline debug - logs don't go in the description field. In the future, please use attachments for logs.]



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