[asterisk-bugs] [Asterisk 0018172]: Peer goes unreachable when placing a call from it.

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18172 
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Reported By:                jordankirby
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18172
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.8.0-rc5 
JIRA:                        
Regression:                 No 
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:             2010-10-20 06:23 CDT
Last Modified:              2010-10-20 06:43 CDT
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Summary:                    Peer goes unreachable when placing a call from it.
Description: 
We are using 1.8RC5, Asterisk RealTime (MySQL) and realtime caching.
Both the server and the phones are behind separate NAT (phone in office,
server in data centre).

In this example, SIP/3001 calls SIP/3002 (via a gosub). The call proceeds
fine but asterisk sees SIP/3001 as unavailable for a period of time after
the initial invite.

This looks to be because asterisk starts sending OPTIONS and NOTIFIES to
the LAN IP address of the phone rather than it's NATed IP address.

In the attached trace:
  10.50.0.47 is the server's LAN IP.
  212.62.4.230 is the server's NATed IP.
  192.168.1.231 is 3001's LAN IP.
  195.59.152.66 is 3001's NATed IP.

Asterisk output:

  == Using SIP RTP CoS mark 5
    -- Executing [3002 at DLPN_All:1] Gosub("SIP/3001-000000aa",
"internal,3002,1")
    -- Executing [3002 at internal:1] Dial("SIP/3001-000000aa", "SIP/3002")
  == Using UDPTL CoS mark 5
  == Using SIP RTP CoS mark 5
    -- Called 3002
    -- SIP/3002-000000ab is ringing
[Oct 20 12:11:05] NOTICE[24099]: chan_sip.c:24694 sip_poke_noanswer: Peer
'3001' is now UNREACHABLE!  Last qualify: 37
  == Spawn extension (internal, 3002, 1) exited non-zero on
'SIP/3001-000000aa'
    -- Registered SIP '3001' at 195.59.152.66:17261
[Oct 20 12:11:55] NOTICE[24099]: chan_sip.c:19523
handle_response_peerpoke: Peer '3001' is now Reachable. (34ms / 2000ms)

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---------------------------------------------------------------------- 
 (0128224) schmidts (manager) - 2010-10-20 06:43
 https://issues.asterisk.org/view.php?id=18172#c128224 
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We require a complete debug log to help triage the issue.

This document will provide instructions on how to collect debugging logs
from an Asterisk machine for the purpose of helping bug marshals
troubleshoot an issue:

http://svn.digium.com/svn/asterisk/trunk/doc/HOWTO_collect_debug_information.txt

could you also please attach the output of sip show peer before and after
the invite when the options stop working.

what i see and this is strange in your invite the c= row looks like this:
c=IN IP4 192.168.1.231 normaly the external ip should be used there not
the internal. Further the sip response of this call itself is sent back to
the right address but somehow the contact ip for this peer is changed.

Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
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2010-10-20 06:43 schmidts       Note Added: 0128224                          
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