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

Asterisk Bug Tracker noreply at bugs.digium.com
Mon Sep 22 16:15:55 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:                     new
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:              2008-09-22 16:15 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
====================================================================== 

---------------------------------------------------------------------- 
 (0092767) putnopvut (administrator) - 2008-09-22 16:15
 http://bugs.digium.com/view.php?id=13523#c92767 
---------------------------------------------------------------------- 
I've made a patch which should correct the parsing error that was occurring
before. I am operating under the assumption that you have promiscredir set
to yes in your sip.conf file. If that assumption is wrong, please let me
know.

Please test 13523.patch and report if it has fixed the problem. Thanks! 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2008-09-22 16:15 putnopvut      Note Added: 0092767                          
======================================================================




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