Ivan Ullmann created ASTERISK-25649:
---------------------------------------
Summary: 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
Severity: Minor
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
== 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'
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.
--
This message was sent by Atlassian JIRA
(v6.2#6252)