[asterisk-bugs] [JIRA] (ASTERISK-23846) Unistim multilines. Loss of voice after second call drops (on a second line).

[Igor Goncharovsky (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-23846?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=222607#comment-222607 ] 

Igor Goncharovsky commented on ASTERISK-23846:
----------------------------------------------

I was not able yet to work on this. Please left it open

> 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: Igor Goncharovsky
>
> 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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