[asterisk-bugs] [JIRA] (ASTERISK-25649) VoiceMail exitcontext not able to use transfer function if local channel engaged
Ivan Ullmann (JIRA)
noreply at issues.asterisk.org
Mon Jan 4 10:05:33 CST 2016
[ https://issues.asterisk.org/jira/browse/ASTERISK-25649?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=228814#comment-228814 ]
Ivan Ullmann commented on ASTERISK-25649:
-----------------------------------------
Rusty,
You're not being dense. I think my explanation doesn't have enough context.
Calls are incoming on the SIP channel from one of my SIP Servers (not Asterisk). The problem I was experiencing is that sometimes my initiating SIP Server would not receive a 18X progress message or a 200 OK within a 15 second timeout. To rectify this, I attempted to transfer the incoming SIP channel call to a local channel using the Dial() function. This allowed me to set a timeout (See extensions.conf line 70). The problem isn't that the local channel doesn't answer or function. The problem is that if I press 0 for Operator, the Transfer() function (See line 52) does not send the call back to the SIP channel. It sends the call to the local channel instead.
Example from the log:
[Jan 4 10:49:12] DEBUG[29631][C-00000001] pbx.c: Launching 'Transfer'
[Jan 4 10:49:12] VERBOSE[29631][C-00000001] pbx.c: -- Executing [o at toSvcCenter:2] Transfer("Local/6022823054 at Leave_VoiceMail-00000001;2", "SIP/ToSvcCenter at 10.93.118.12") in new stack
[Jan 4 10:49:12] DEBUG[29631][C-00000001] pbx.c: Launching 'Hangup'
[Jan 4 10:49:12] VERBOSE[29631][C-00000001] pbx.c: -- Executing [o at toSvcCenter:3] Hangup("Local/6022823054 at Leave_VoiceMail-00000001;2", "") in new stack
[Jan 4 10:49:12] DEBUG[29631][C-00000001] channel.c: Soft-Hanging up channel 'Local/6022823054 at Leave_VoiceMail-00000001;2'
[Jan 4 10:49:12] DEBUG[29631][C-00000001] pbx.c: Spawn extension (toSvcCenter,o,3) exited non-zero on 'Local/6022823054 at Leave_VoiceMail-00000001;2'
[Jan 4 10:49:12] VERBOSE[29631][C-00000001] pbx.c: == Spawn extension (toSvcCenter, o, 3) exited non-zero on 'Local/6022823054 at Leave_VoiceMail-00000001;2'
I am not using a features.conf transfer to the best of my knowledge. I don't have anything configured there to do so. Additionally, I have had this operating previously. It was once I added Dial() that it stopped performing the transfer.
Additionally, I have provided the features.conf file for your perusal.
Thank you,
-Ivan
> 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: Unassigned
> Severity: Minor
> Attachments: extensions.conf, full.asterisk-25649.txt, 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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