[asterisk-bugs] [Asterisk 0013117]: [patch] multiple 'transport' on peer doesn't work, tcp port still open

noreply at bugs.digium.com noreply at bugs.digium.com
Tue Jul 22 15:03:35 CDT 2008


A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13117 
====================================================================== 
Reported By:                pj
Assigned To:                bbryant
====================================================================== 
Project:                    Asterisk
Issue ID:                   13117
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           SVN 
SVN Branch (only for SVN checkouts, not tarball releases):  trunk 
SVN Revision (number only!): 132312 
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             07-20-2008 15:28 CDT
Last Modified:              07-22-2008 15:03 CDT
====================================================================== 
Summary:                    [patch] multiple 'transport' on peer doesn't work,
tcp port still open
Description: 
When I specify eg. 'transport=udp,tcp,tls' in peer configuration, in effect
is only first transport type. 
Currently, I can't specify multiple transport for peers.
Separate to more rows, doesn't solve this, ie:
transport=udp
transport=tcp
transport=tls
still only first is used.

Second issue is, that even if 'sip show settings' displays tcp port
disabled (default), asterisk still have tcp/5060 open.
====================================================================== 

---------------------------------------------------------------------- 
 pj - 07-22-08 15:03  
---------------------------------------------------------------------- 
now it compile and working fine, except in case, when I have sip presence
in softphone turned on to monitor some extensions state. In that case, I'm
not able even to make calls when using tls transport. When have presence on
and with udp transport, I can make outgoing calls from softphone, but peer
seems to be unregistered, so unreachable for accept calls :(
sip debug attached 

Issue History 
Date Modified   Username       Field                    Change               
====================================================================== 
07-22-08 15:03  pj             Note Added: 0090574                          
======================================================================




More information about the asterisk-bugs mailing list