[asterisk-bugs] [JIRA] (ASTERISK-26895) AMI Redirect'd channel hangs up if created from AMI Originate

Bruce McIntosh (JIRA) noreply at issues.asterisk.org
Thu Mar 23 16:51:10 CDT 2017


Bruce McIntosh created ASTERISK-26895:
-----------------------------------------

             Summary: AMI Redirect'd channel hangs up if created from AMI Originate
                 Key: ASTERISK-26895
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26895
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Resources/res_pjsip
    Affects Versions: 14.3.0, 13.9.1
         Environment: Centos Linux 2.6.32-729.el6.x86_64
            Reporter: Bruce McIntosh


Using Asterisk 14.3

I use AMI Originate to create a call between two PJSIP endpoints:
 Action: Originate 
 Channel: PJSIP/88074 
 Application: Dial
 Data: PJSIP/5000
 ActionID: 12345

then, when I use AMI to Redirect the channel, it hangs up instead of entering the dial plan and calling the specified extension:
 Action: Redirect
 ActionID: 12346
 Channel: PJSIP/88074-00000009
 Context: DHFCS
 Priority: 1
 Exten: 227000

The log follows:
{code:title=Log|borderStyle=solid}
[Mar 23 05:33:38] DEBUG[19126] manager.c: Running action 'Redirect'
[Mar 23 05:33:38] DEBUG[19126] channel.c: Soft-Hanging (0x02) up channel 'PJSIP/88074-00000009'
[Mar 23 05:33:38] DEBUG[19303] bridge_channel.c: Setting 0x7f177c0cd6a8(PJSIP/88074-00000009) state from:0 to:1
[Mar 23 05:33:38] DEBUG[19303] bridge_channel.c: Bridge 4233b7a0-b405-44cc-a2c3-cb0091cda63f: pulling 0x7f177c0cd6a8(PJSIP/88074-00000009)
[Mar 23 05:33:38] VERBOSE[19303] bridge_channel.c: Channel PJSIP/88074-00000009 left 'native_rtp' basic-bridge <4233b7a0-b405-44cc-a2c3-cb0091cda63f>
[Mar 23 05:33:38] DEBUG[19303] bridge_channel.c: Bridge 4233b7a0-b405-44cc-a2c3-cb0091cda63f: 0x7f177c0cd6a8(PJSIP/88074-00000009) is leaving native_rtp technology
[Mar 23 05:33:38] DEBUG[19303] bridge_native_rtp.c: Discontinued RTP bridging of 'PJSIP/88074-00000009' and 'PJSIP/5000-0000000a' - media will flow through Asterisk core
[Mar 23 05:33:38] DEBUG[19303] bridge.c: Bridge 4233b7a0-b405-44cc-a2c3-cb0091cda63f: dissolving bridge with cause 16(Normal Clearing)
[Mar 23 05:33:38] DEBUG[19303] bridge_channel.c: Setting 0x7f177c048b08(PJSIP/5000-0000000a) state from:0 to:2
[Mar 23 05:33:38] DEBUG[19303] bridge.c: Bridge 4233b7a0-b405-44cc-a2c3-cb0091cda63f: queueing action type:13 sub:1001
[Mar 23 05:33:38] DEBUG[19303] bridge_channel.c: Channel PJSIP/88074-00000009 will survive this bridge; clearing outgoing (dialed) flag
[Mar 23 05:33:38] DEBUG[19303] bridge.c: Bridge 4233b7a0-b405-44cc-a2c3-cb0091cda63f is dissolved, not performing smart bridge operation.
[Mar 23 05:33:38] DEBUG[19303] app_dial.c: Exiting with DIALSTATUS=ANSWER.
[Mar 23 05:33:38] DEBUG[19303] channel.c: Hanging up channel 'PJSIP/88074-00000009'
[Mar 23 05:33:38] DEBUG[19303] chan_pjsip.c: AST hangup cause 16 (no match found in PJSIP)
[Mar 23 05:33:38] DEBUG[19073] manager.c: Mansession: 0x7f1768001308 refcount now 2
[Mar 23 05:33:38] DEBUG[19056] threadpool.c: Increasing threadpool stasis-core's size by 1
[Mar 23 05:33:38] DEBUG[19056] threadpool.c: Increasing threadpool stasis-core's size by 1
[Mar 23 05:33:38] DEBUG[19298] res_pjsip_session.c: Method is BYE
{code}

If I repeat the test but instead create the call using a VOIP phone to dial from 88074 to 5000 (not using AMI at all), then the AMI Redirect works fine.




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