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

[Bruce McIntosh (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-26895?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Bruce McIntosh closed ASTERISK-26895.
-------------------------------------

    Resolution: Not A Bug

> 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: 13.9.1, 14.3.0
>         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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