[asterisk-bugs] [Asterisk 0013523]: Trouble with Temporarily Moved

Tue Sep 23 10:27:16 CDT 2008

A NOTE has been added to this issue. 
====================================================================== 
http://bugs.digium.com/view.php?id=13523 
====================================================================== 
Reported By:                mattdarnell
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   13523
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     ready for testing
Asterisk Version:           1.6.0-rc6 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Disclaimer on File?:        N/A 
Request Review:              
====================================================================== 
Date Submitted:             2008-09-19 17:50 CDT
Last Modified:              2008-09-23 10:27 CDT
====================================================================== 
Summary:                    Trouble with Temporarily Moved
Description: 
Iam testing the SIP TCP and am having trouble with the Moved
Temporarily SIP Message.  I am using 1.6 RC6

Once Asterisk receives the message I get this in the console:
Unable to create local channel for call forward to
'SIP/::::TCP at 3451@10.10.20.31:5067' (cause = 20)

Below is the SIP trace.

*********************

<--- SIP read from TCP://10.10.20.31:5060 --->
SIP/2.0 100 Trying
FROM: "1000"<sip:1000 at wikitelcom.com>;tag=as74ab814b
TO: <sip:3451 at 10.10.20.31>
CSEQ: 102 INVITE
CALL-ID: 1989d15163e8597e31577007407992e8 at wikitelcom.com
VIA: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK5f356ae5;rport
CONTENT-LENGTH: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from TCP://10.10.20.31:5060 --->
SIP/2.0 302 Moved Temporarily
FROM: "1000"<sip:1000 at wikitelcom.com>;tag=as74ab814b
TO: <sip:3451 at 10.10.20.31>;tag=97bcfff0e7
CSEQ: 102 INVITE
CALL-ID: 1989d15163e8597e31577007407992e8 at wikitelcom.com
VIA: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK5f356ae5;rport
CONTACT: <sip:3451 at 10.10.20.31:5067;transport=TCP>
CONTENT-LENGTH: 0
SERVER: RTCC/3.0.0.0

<------------->
--- (9 headers 0 lines) ---
-- Got SIP response 302 "Moved Temporarily" back from 10.10.20.31

Transmitting (no NAT) to 10.10.20.31:5060:
ACK sip:3451 at 10.10.20.31 SIP/2.0
Via: SIP/2.0/TCP 10.10.20.50:5060;branch=z9hG4bK5f356ae5;rport
Max-Forwards: 70
From: "1000" <sip:1000 at wikitelcom.com>;tag=as74ab814b
To: <sip:3451 at 10.10.20.31>;tag=97bcfff0e7
Contact: <sip:1000 at 10.10.20.50:5060;transport=TCP>
Call-ID: 1989d15163e8597e31577007407992e8 at wikitelcom.com
CSeq: 102 ACK
User-Agent: Makai
Content-Length: 0

---
 -- Now forwarding SIP/1000-09648728 to
'SIP/::::TCP at 3451@10.10.20.31:5067' (thanks to SIP/exch-0964c710)

 == Using SIP RTP CoS mark 5

[Sep 19 11:42:27] WARNING[29048]: chan_sip.c:4181 create_addr: No such
host: 3451 at 10.10.20.31

Really destroying SIP dialog
'768961345131b4ac174866b0686596c2 at 10.10.20.50' Method: INVITE

[Sep 19 11:42:27] NOTICE[29048]: app_dial.c:505 do_forward: Unable to
create local channel for call forward to
'SIP/::::TCP at 3451@10.10.20.31:5067' (cause = 20)
 == Everyone is busy/congested at this time (1:0/0/1)
   -- Auto fallthrough, channel 'SIP/1000-09648728' status is
'CHANUNAVAIL'

<--- Reliably Transmitting (NAT) to 67.53.192.111:54240 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/TCP
67.53.191.111:38088;branch=z9hG4bK-d8754z-3b7d097a947cd95e-1---d8754z-;received=67.53.192.138;rport=54240
From: "1000"<sip:1000 at 64.75.211.160>;tag=c177381d
To: "1000"<sip:1000 at 64.75.211.160>;tag=as1f99c9d9
Call-ID: MDY2NDQ3YTczNWVlZDFmYzJlNDFkMGNiZDgyMjM2ODY.
CSeq: 2 INVITE
User-Agent: Makai
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Contact: <sip:1000 at 64.75.211.111:5060;transport=TCP>
Content-Length: 0

<------------>
<--- SIP read from TCP://67.53.192.111:54240 --->
ACK sip:1000 at 64.75.211.111 SIP/2.0
Via: SIP/2.0/TCP
67.53.191.111:38088;branch=z9hG4bK-d8754z-3b7d097a947cd95e-1---d8754z-;rport
To: "1000"<sip:1000 at 64.75.211.111>;tag=as1f99c9d9
From: "1000"<sip:1000 at 64.75.211.111>;tag=c177381d
Call-ID: MDY2NDQ3YTczNWVlZDFmYzJlNDFkMGNiZDgyMjM2ODY.
CSeq: 2 ACK
Content-Length: 0
====================================================================== 

---------------------------------------------------------------------- 
 (0092788) svnbot (reporter) - 2008-09-23 10:27
 http://bugs.digium.com/view.php?id=13523#c92788 
---------------------------------------------------------------------- 
Repository: asterisk
Revision: 144025

U   trunk/channels/chan_sip.c

------------------------------------------------------------------------
r144025 | mmichelson | 2008-09-23 10:27:14 -0500 (Tue, 23 Sep 2008) | 16
lines

When a promiscuous redirect contained both a user and
host portion in the Contact URI and specifies a 
transport, the parsing done in parse_moved_contact
resulted in a malformed URI.

This commit fixes the parsing so that a proper
Dial string may be formed when the forwarded
call is placed.

(closes issue http://bugs.digium.com/view.php?id=13523)
Reported by: mattdarnell
Patches:
      13523v2.patch uploaded by putnopvut (license 60)
Tested by: mattdarnell

------------------------------------------------------------------------

http://svn.digium.com/view/asterisk?view=rev&revision=144025 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-09-23 10:27 svnbot         Checkin                                      
2008-09-23 10:27 svnbot         Note Added: 0092788                          
======================================================================