[asterisk-bugs] [Asterisk 0011800]: asterisk 1.6-beta1 destroys cisco 7960 (sip firmware 7.4) outbound calls after 20sec due to no response to 200 OK

[noreply at bugs.digium.com]

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=11800 
====================================================================== 
Reported By:                hmodes
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   11800
Category:                   Channels/chan_sip/Interoperability
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.0-beta1 
SVN Branch (only for SVN checkouts, not tarball releases): 1.6 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             01-19-2008 23:04 CST
Last Modified:              01-25-2008 08:59 CST
====================================================================== 
Summary:                    asterisk 1.6-beta1 destroys cisco 7960 (sip firmware
7.4) outbound calls after 20sec due to no response to 200 OK
Description: 
Version tag on this report is incorrect as there is no 1.6-beta1 tag yet.

1.6-beta1 appears to have an issue with sip timers that causes cisco 7960
sip firmware 7.4 clients to be disconnected after 20 seconds for not
responding to 200 OK (marked as 'critical packet') once call setup is
complete.  See debug.  Incoming calls to the client do not experience the
disconnect.  This is reproducible on two phones both on the lan with
asterisk and at a remote site. 
====================================================================== 

---------------------------------------------------------------------- 
 hmodes - 01-25-08 08:59  
---------------------------------------------------------------------- 
sip.conf general section follows.

[general]
context=default                 ; Default context for incoming calls
recordhistory=yes               ; Record SIP history by default 
realm=matrix.gs         ; Realm for digest authentication
bindport=5066           ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes                   ; Enable DNS SRV lookups on outbound
calls
tos_audio=184
videosupport=no         ; Turn on support for SIP video
sdpsession=session
allow=all
useragent=matrix.gs asterisk            ; Allows you to change the user
agent string
externip = *my external ip here*
localnet=172.27.28.0/255.255.255.0; All RFC 1918 addresses are local
networks 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
01-25-08 08:59  hmodes         Note Added: 0081178                          
======================================================================