[asterisk-bugs] [JIRA] (ASTERISK-25649) VoiceMail exitcontext not able to use transfer function if local channel engaged

[Rusty Newton (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-25649?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Rusty Newton updated ASTERISK-25649:
------------------------------------

    Assignee: Ivan Ullmann
      Status: Waiting for Feedback  (was: Triage)

I think I'm missing something here, or being dense.

How do you initiate the transfer and which channel initiates it?

I see the transfer sound byte being played. I'm presuming you are using a features.conf transfer?

Can you post a debug log.. I'll add a comment with the link.

> VoiceMail exitcontext not able to use transfer function if local channel engaged
> --------------------------------------------------------------------------------
>
>                 Key: ASTERISK-25649
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25649
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_voicemail/ODBC, Channels/chan_local, Channels/chan_sip/General
>    Affects Versions: 11.2.2
>         Environment: Development
>            Reporter: Ivan Ullmann
>            Assignee: Ivan Ullmann
>            Severity: Minor
>         Attachments: extensions.conf, sip.conf, voicemail.conf
>
>
> When triggering exitcontext logic inside of the VoiceMail application, calls sent to the local channel cannot transfer.
> Call Flow:
>      1. Incoming call to Asterisk server via SIP
>      2. Call is processed appropriately to VoiceMail application via a Dial function to a local channel
>      3. Press 0
>      4. Call triggers 'toSvcCenter' dialplan logic
>      5. Transfer function triggered
>      6. Extension exits non-zero on Local channel
> {noformat}
>   == Using SIP RTP TOS bits 184
>   == Using SIP RTP CoS mark 5
>     -- Executing [9999999999 at sip:1] Wait("SIP/Asterisk_CLE-00000024", "1") in new stack
>     -- Executing [9999999999 at sip:2] Set("SIP/Asterisk_CLE-00000024", "SIP_CODEC=ulaw") in new stack
>     -- Executing [9999999999 at sip:3] Set("SIP/Asterisk_CLE-00000024", "GLOBAL(INITIAL_CHANNEL)=SIP/Asterisk_CLE-00000024") in new stack
>   == Setting global variable 'INITIAL_CHANNEL' to 'SIP/Asterisk_CLE-00000024'
>     -- Executing [9999999999 at sip:4] Set("SIP/Asterisk_CLE-00000024", "GLOBAL(INCOMING_SIP_PEER)=10.93.118.12") in new stack
>   == Setting global variable 'INCOMING_SIP_PEER' to '10.93.118.12'
>     -- Executing [9999999999 at sip:5] GotoIf("SIP/Asterisk_CLE-00000024", "0?internal:next1") in new stack
>     -- Goto (sip,9999999999,14)
>     -- Executing [9999999999 at sip:14] GotoIf("SIP/Asterisk_CLE-00000024", "0?external:customer") in new stack
>     -- Goto (sip,9999999999,17)
>     -- Executing [9999999999 at sip:17] Dial("SIP/Asterisk_CLE-00000024", "local/9999999999 at Leave_VoiceMail/b,8,r") in new stack
>     -- Called local/9999999999 at Leave_VoiceMail/b
>     -- Executing [9999999999 at Leave_VoiceMail:1] VoiceMail("Local/9999999999 at Leave_VoiceMail-00000004;2", "9999999999 at GVMA_DN,su") in new stack
>     -- Local/9999999999 at Leave_VoiceMail-00000004;1 answered SIP/Asterisk_CLE-00000024
> [Dec 29 18:06:42] NOTICE[19515][C-00000026]: chan_sip.c:7238 try_suggested_sip_codec: Changing codec to 'ulaw' for this call because of ${SIP_CODEC} variable
> [Dec 29 18:06:42] NOTICE[19515][C-00000026]: chan_sip.c:7238 try_suggested_sip_codec: Changing codec to 'ulaw' for this call because of ${SIP_CODEC} variable
>        > 0x189d30a0 -- Probation passed - setting RTP source address to 10.93.107.65:3794
>     -- <Local/9999999999 at Leave_VoiceMail-00000004;2> Playing '/var/spool/asterisk/voicemail/GVMA_DN/9999999999/unavail.slin' (language 'en')
>     -- <Local/9999999999 at Leave_VoiceMail-00000004;2> Playing 'transfer.ulaw' (language 'en')
>     -- Executing [o at toSvcCenter:1] BackGround("Local/9999999999 at Leave_VoiceMail-00000004;2", "one-moment-please") in new stack
>     -- <Local/9999999999 at Leave_VoiceMail-00000004;2> Playing 'one-moment-please.ulaw' (language 'en')
>     -- Executing [o at toSvcCenter:2] Transfer("Local/9999999999 at Leave_VoiceMail-00000004;2", "SIP/ToSvcCenter at 10.93.118.12") in new stack
>     -- Executing [o at toSvcCenter:3] Hangup("Local/9999999999 at Leave_VoiceMail-00000004;2", "") in new stack
>   == Spawn extension (toSvcCenter, o, 3) exited non-zero on 'Local/9999999999 at Leave_VoiceMail-00000004;2'
>   == Spawn extension (sip, 9999999999, 17) exited non-zero on 'SIP/Asterisk_CLE-00000024'
> {noformat}
> I am unable to perform a Transfer() back on the SIP/Asterisk_CLE-00000024 channel.  My call flow requires a SIP REFER in order to remove this server from call flow.  Dial() works, but the upstream server is expecting to produce a REFER to the originating SIP Server as well, and this results in calls being REFER'd back to Asterisk rather than upstream.



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