[asterisk-bugs] [Asterisk 0016238]: sip calls drop because of BYE's

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=16238 
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Reported By:                seandarcy
Assigned To:                
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Project:                    Asterisk
Issue ID:                   16238
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.6.0.18-rc2 
JIRA:                        
Regression:                 Yes 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-11-12 16:30 CST
Last Modified:              2009-11-13 12:36 CST
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Summary:                    sip calls drop because of BYE's
Description: 
On 1.6.0.16, 1.6.0.18-rc1 and 2, SIP calls die because * sends a BYE. We
have 12 internal SIP phones. The server connects them to sip, iax and dahdi
over libpri. iax and dahdi work fine. The internal SIP phones work fine.
But outgoing SIP calls are terminated with a BYE. We've tried sip over both
Teliax and Junction. Same result. Teliax support told us they could see the
BYE.
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 (0113803) seandarcy (reporter) - 2009-11-13 12:36
 https://issues.asterisk.org/view.php?id=16238#c113803 
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Nope. The call drops when it's picked up. Here's CLI and sip.conf context.
I'll upload sip debug.

    -- Executing [1XXXYYY4299 at internal:1] Answer("DAHDI/1-1", "") in new
stack
    -- Executing [1XXXYYY4299 at internal:2] NoOp("DAHDI/1-1", "Context:
outbound-long-distance") in new stack
    -- Executing [1XXXYYY4299 at internal:3] Dial("DAHDI/1-1",
"SIP/teliax-sip/XXXYYY4299") in new stack
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
  == Using SIP VRTP TOS bits 136
  == Using SIP VRTP CoS mark 4
  == Using UDPTL TOS bits 184
  == Using UDPTL CoS mark 5
    -- Called teliax-sip/XXXYYY4299
    -- SIP/teliax-sip-00000003 is making progress passing it to DAHDI/1-1
    -- SIP/teliax-sip-00000003 answered DAHDI/1-1
    -- fixed jitterbuffer created on channel DAHDI/1-1
WARNING[8250]: chan_sip.c:3587 __sip_autodestruct: Autodestruct on dialog
'1ea0b25d06cdf9d36417a52e65d52f1a at 76.248.146.19' with owner in place
(Method: INVITE)
  == Spawn extension (internal, 1XXXYYY4299, 3) exited non-zero on
'DAHDI/1-1'
    -- Hungup 'DAHDI/1-1'
    -- fixed jitterbuffer destroyed on channel DAHDI/1-1


[teliax-sip]
type=peer
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=gsm
dtmfmode=rfc2833
insecure=port,invite
context=from-teliax-sip ; no such context. There should be no calls from
teliax.
defaultuser=USERNAME(Sip)
secret=PASSWORD
host=nyc.teliax.net
qualify=yes
session-timers=refuse 

Issue History 
Date Modified    Username       Field                    Change               
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2009-11-13 12:36 seandarcy      Note Added: 0113803                          
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