[asterisk-bugs] [JIRA] (ASTERISK-24526) Exec dial from AGI script does not hang up caller after called party hangs up

[Taylor Hawkes (JIRA)]

Taylor Hawkes created ASTERISK-24526:
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             Summary: Exec dial from AGI script does not hang up caller after called party hangs up
                 Key: ASTERISK-24526
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24526
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
    Affects Versions: 13.0.0, 12.7.0
         Environment: Linux/Ubuntu 13.04 (Raring Ringtail)
            Reporter: Taylor Hawkes


When executing a DIAL command from an AGI script the calling party will continue to the next extension after called party hangs up. This occurs when the "g" option in the dial command is not set. 

Behavior works as expected if Dial is called in the (extensions.conf), seems to be something with AGI &  dial application. 

This issue does not occur in Asterisk 11.7 and does occur in Asterisk 12.7. 

>From my AGI Script.
$agi->write("EXEC DIAL SIP/6001\n");

extensions.conf

[others]
exten => 12345,1,Answer
exten => 12345,2,AGI(/home/a_web/sip_load.php)
exten => 12345,3,SayDigits(12345))
exten => 12345,4,Hangup()


---------------------------------------------------------------------------
-- Executing [12345 at others:1] Answer("SIP/168.244.49.149-00000004", "") in new stack
[Nov 14 20:57:03] WARNING[31242]: config.c:2829 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available
[Nov 14 20:57:03] WARNING[31242]: config.c:2829 find_engine: Realtime mapping for 'sippeers' found to engine 'mysql', but the engine is not available
       > 0x7f3244053e50 -- Probation passed - setting RTP source address to 168.244.49.149:7078
    -- Executing [12345 at others:2] AGI("SIP/168.244.49.149-00000004", "/home/ivr_test/sip_load.php") in new stack
    -- Launched AGI Script /home/ivr_test/sip_load.php
    -- AGI Script Executing Application: (DIAL) Options: (SIP/6001)
  == Using SIP RTP CoS mark 5
    -- Called SIP/6001
    -- SIP/6001-00000005 is ringing
       > 0x7f325801d9f0 -- Probation passed - setting RTP source address to 168.244.49.149:8000
    -- SIP/6001-00000005 answered SIP/168.244.49.149-00000004
    -- Channel SIP/168.244.49.149-00000004 joined 'simple_bridge' basic-bridge <a1f53451-3bf1-4fde-9b23-fa865cfdf2a7>
    -- Channel SIP/6001-00000005 joined 'simple_bridge' basic-bridge <a1f53451-3bf1-4fde-9b23-fa865cfdf2a7>
       > Bridge a1f53451-3bf1-4fde-9b23-fa865cfdf2a7: switching from simple_bridge technology to native_rtp
       > 0x7f325801d9f0 -- Probation passed - setting RTP source address to 168.244.49.149:8000
    -- Channel SIP/6001-00000005 left 'native_rtp' basic-bridge <a1f53451-3bf1-4fde-9b23-fa865cfdf2a7>
    -- Channel SIP/168.244.49.149-00000004 left 'native_rtp' basic-bridge <a1f53451-3bf1-4fde-9b23-fa865cfdf2a7>
[Nov 14 20:57:10] ERROR[31321][C-00000002]: utils.c:1363 ast_carefulwrite: write() returned error: Broken pipe
    -- <SIP/168.244.49.149-00000004>AGI Script /home/ivr_test/sip_load.php completed, returning 0
    -- Executing [12345 at others:3] SayDigits("SIP/168.244.49.149-00000004", "12345)") in new stack
    -- <SIP/168.244.49.149-00000004> Playing 'digits/1.ulaw' (language 'en')
    -- <SIP/168.244.49.149-00000004> Playing 'digits/2.ulaw' (language 'en')
    -- <SIP/168.244.49.149-00000004> Playing 'digits/3.ulaw' (language 'en')
    -- <SIP/168.244.49.149-00000004> Playing 'digits/4.ulaw' (language 'en')
    -- <SIP/168.244.49.149-00000004> Playing 'digits/5.ulaw' (language 'en')
    -- Executing [12345 at others:4] Hangup("SIP/168.244.49.149-00000004", "") in new stack
  == Spawn extension (others, 12345, 4) exited non-zero on 'SIP/168.244.49.149-00000004'
    -- Executing [h at others:1] Answer("SIP/168.244.49.149-00000004", "") in new stack






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