[asterisk-bugs] [Asterisk 0018540]: Problem with unistim on Asterisk 1.8.1.1

Asterisk Bug Tracker noreply at bugs.digium.com
Tue Dec 28 10:33:24 UTC 2010


The following issue has been SUBMITTED. 
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https://issues.asterisk.org/view.php?id=18540 
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Reported By:                tequ
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18540
Category:                   Channels/chan_unistim
Reproducibility:            have not tried
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.1.1 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-12-28 04:33 CST
Last Modified:              2010-12-28 04:33 CST
====================================================================== 
Summary:                    Problem with unistim on Asterisk 1.8.1.1
Description: 
I have a problem with unistim channel. I don't hear anything during the
call.
In addition, any calls to nortel phone causes asterisk crash.

*CLI> [Dec 28 11:12:25] WARNING[15177]: res_rtp_asterisk.c:422
create_new_socket: Unable to allocate RTP socket: Address family not
supported by protocol
[Dec 28 11:12:25] WARNING[15177]: chan_unistim.c:2073 start_rtp: Unable to
create RTP session: Address family not supported by protocol
binaddr=10.0.0.21
    -- Starting switch on '999 at 2002-0' to 123456
    -- Executing [123456 at incoming:1] Answer("999 at 2002-0", "") in new
stack
[Dec 28 11:12:25] WARNING[15187]: res_rtp_asterisk.c:422
create_new_socket: Unable to allocate RTP socket: Address family not
supported by protocol
[Dec 28 11:12:25] WARNING[15187]: chan_unistim.c:2073 start_rtp: Unable to
create RTP session: Address family not supported by protocol
binaddr=10.0.0.21
    -- Executing [123456 at incoming:2] Playback("999 at 2002-0", "beep") in new
stack
    -- <999 at 2002-0> Playing 'beep.gsm' (language 'en')
    -- Executing [123456 at incoming:3] Wait("999 at 2002-0", "4") in new stack
    -- Executing [123456 at incoming:4] Hangup("999 at 2002-0", "") in new
stack
  == Spawn extension (incoming, 123456, 4) exited non-zero on
'999 at 2002-0'
USTM(999 at 2002-0) channel already destroyed

# unistim.conf
[general]
port=4100
bindaddr=0.0.0.0

[2002]
device=000AE475AD4B
rtp_port=10000
rtp_method=3
status_method=0
titledefault=TEST
height=1
country=us
ringvolume=2
ringstyle=3
callhistory=1
callerid="Customer Support" <555-234-5678>
context=incoming
linelabel="Support"
extension=line
line => 999

====================================================================== 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-12-28 04:33 tequ           New Issue                                    
2010-12-28 04:33 tequ           Asterisk Version          => 1.8.1.1         
2010-12-28 04:33 tequ           Regression                => No              
2010-12-28 04:33 tequ           SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
======================================================================




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