[asterisk-bugs] [JIRA] (ASTERISK-23846) Unistim multilines. Loss of voice after second call drops (on a second line).
Rusty Newton (JIRA)
noreply at issues.asterisk.org
Wed Aug 6 09:15:01 CDT 2014
[ https://issues.asterisk.org/jira/browse/ASTERISK-23846?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Rusty Newton updated ASTERISK-23846:
------------------------------------
Assignee: Rusty Newton (was: Igor Goncharovsky)
Status: Triage (was: Waiting for Feedback)
> Unistim multilines. Loss of voice after second call drops (on a second line).
> -----------------------------------------------------------------------------
>
> Key: ASTERISK-23846
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-23846
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_unistim
> Affects Versions: 11.7.0, 11.10.0
> Environment: jail on FreeBSD 9.2-RELEASE-p3
> Reporter: Rustam Khankishyiev
> Assignee: Rusty Newton
>
> unistim.conf:
> {noformat}
> [general]
> port=5000
> public_ip=192.168.99.85
> [7098]
> device=000ae4762c49
> rtp_port=10000
> rtp_method=3
> status_method=0
> height=1
> maintext0="Company Name"
> dateformat=1
> timeformat=2
> contrast=8
> country=us
> ringvolume=2
> ringstyle=3
> cwvolume=2
> cwstyle=3
> sharpdial=1
> interdigit_timer=4000
> callhistory=1
> callerid="7098" <7098>
> context=kiev
> callgroup=5
> pickupgroup=5
> linelabel="7098"
> extension=none
> line => 7098
> line => 7098
> bookmark=4 at 1234567@1234567
> {noformat}
> extensions.conf:
> {noformat}
> exten => _7XXX,1,Dial(USTM/${EXTEN}@${EXTEN},120,TtKk)
> exten => _7XXX,2,HangUp()
> {noformat}
> 1. Incoming call on the first line.
> {noformat}
> -- Executing [7098 at kiev:1] Dial("SIP/7028-00000032", "USTM/7098 at 7098,120,TtKk") in new stack
> -- Called USTM/7098 at 7098
> -- USTM/7098 at 7098-0x803d38280 is ringing
> > 0x828f5b000 -- Probation passed - setting RTP source address to 192.168.192.156:10000
> {noformat}
> 2. Pick up the phone and talk.
> {noformat}
> -- USTM/7098 at 7098-0x803d38280 answered SIP/7028-00000032
> > 0x82928a000 -- Probation passed - setting RTP source address to 192.168.192.150:16412
> {noformat}
> 3. If at that time someone calls to the second line and immediately hang up, the voice in the first line stops.
> {noformat}
> -- Executing [7098 at kiev:1] Dial("SIP/7052-00000033", "USTM/7098 at 7098,120,TtKk") in new stack
> -- Called USTM/7098 at 7098
> -- USTM/7098 at 7098-0x828973dc0 is ringing
> == Spawn extension (kiev, 7098, 1) exited non-zero on 'SIP/7052-00000033'
> {noformat}
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