[asterisk-bugs] [Asterisk 0016083]: CLI incorrectly delivered when there is no calling number

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=16083 
====================================================================== 
Reported By:                mrmrmrmr
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   16083
Category:                   Channels/chan_sip/Transfers
Reproducibility:            always
Severity:                   tweak
Priority:                   normal
Status:                     acknowledged
Asterisk Version:           Older 1.6.0 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2009-10-15 14:46 CDT
Last Modified:              2009-10-16 09:36 CDT
====================================================================== 
Summary:                    CLI incorrectly delivered when there is no calling
number
Description: 
I am using a SPA 3000 as a PSTN gateway. Incoming PSTN calls are connected

to Asterisk through SPA 3000 (it has a fxo port) via SIP.

Everything is fine with this call scenario, but if the incoming PSTN call

has no caller ID, then Asterisk receives the call with contact header and

from header as "sip:192.168.254.5"

When it sends the same call to an internal extension Asterisk adds number
"192168254254" as caller ID to both from and contact fields. 
"sip:192168254254 at 192.168.254.5"

I  checked all configuration files but couldn't find a way to remove this
caller ID. These calls don't have a real calling number, so they should be
delivered to the extension without any calling number.

I will add the SIP trace captured on Astersik console and the network
trace including the SIP packets to the issue.
====================================================================== 

---------------------------------------------------------------------- 
 (0112366) jsmith (administrator) - 2009-10-16 09:36
 https://issues.asterisk.org/view.php?id=16083#c112366 
---------------------------------------------------------------------- 
I'm not sure this has any relevance, but I find this line interesting:

Found peer 'pstn sip' for '192.168.254.254' from 192.168.254.5:5061

Do you really have a peer section in sip.conf with a space in it, like
this?

[pstn sip]

If so, you may want to try removing the spaces from your peer name and see
if that makes a difference.  As far as I know, it's not valid to have
spaces in section names in any of the Asterisk configuration files. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-10-16 09:36 jsmith         Note Added: 0112366                          
======================================================================