<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
Chad, You are right. tcpdump shows Asterisk sees 777 when the packet
arrived. <br>
<br>
It's truned out my router somehow modified the packet! I am using a
Asus RT-N16 router with TomatoUSB firmware. There is a setting "SIP
Helper". I disabled this "feature" on the router then everything
back to normal. <br>
<br>
There is one thing still puzzle me: It seems enable or disable this
"feature" doesn't effect other SIP thunks. It could be the Sippy
server use two different IP. The INVITE come from 208.65.xxx.xxx,
but in its packet it try to use 74.205.216.77 as contact address. Is
my guess correct? Why it does this? <br>
<br>
<div class="moz-signature"><b>Jian </b></div>
<br>
On 11-01-27 04:31 PM, Chad Wallace wrote:
<blockquote cite="mid:20110127163121.423aa25f@ws80.int.tlc"
type="cite">
<pre wrap="">On Thu, 27 Jan 2011 14:52:06 -0800
Jian Gao <a class="moz-txt-link-rfc2396E" href="mailto:jian.gao@sjgeophysics.com"><jian.gao@sjgeophysics.com></a> wrote:
</pre>
<blockquote type="cite">
<pre wrap="">Today I upgraded my Asterisk to the new 1.4.39.1. One of sip trunk
stop working after the upgrade. Here is the sip debug:
---------------------------------------------------------------------------
<--- SIP read from 208.65.xxx.xxx:5060 --->
</pre>
</blockquote>
<pre wrap="">
That packet is coming from the other end (Sippy). The problem is
probably there. However, it could be that the networking routines in
Asterisk have added a 7 at the end. You could compare a tcpdump of
that packet to what Asterisk sees. If the tcpdump shows .777 then the
problem is in Sippy. If it shows .77 then the problem is in Asterisk.
</pre>
<blockquote type="cite">
<pre wrap="">INVITE <a class="moz-txt-link-abbreviated" href="mailto:sip:1778xxxxxxx@10.11.22.77:5060">sip:1778xxxxxxx@10.11.22.77:5060</a> SIP/2.0
Via: SIP/2.0/UDP
208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport
Via: SIP/2.0/UDP
208.65.xxx.xxx:5061;branch=z9hG4bK-uhhmj2ir4ew6cn4p;rport=5061
Max-Forwards: 69
Record-Route: <sip:208.65.xxx.xxx;lr>
Contact: "Anonymous"<sip:208.65.xxx.xxx:5061>
To: <a class="moz-txt-link-rfc2396E" href="mailto:sip:1778xxxxxxx@208.65.xxx.xxx:5060"><sip:1778xxxxxxx@208.65.xxx.xxx:5060></a>
From: <a class="moz-txt-link-rfc2396E" href="mailto:sip:604xxxxxxx@208.65.xxx.xxx:5060"><sip:604xxxxxxx@208.65.xxx.xxx:5060></a>;tag=ixpa27sbhn3inu5x.o
Call-ID: <a class="moz-txt-link-abbreviated" href="mailto:550D37B3@208.72.xxx.xxx~o">550D37B3@208.72.xxx.xxx~o</a>
CSeq: 819 INVITE
Expires: 300
Content-Disposition: session
Content-Type: application/sdp
User-Agent: Sippy
cisco-GUID: 2851810672-711266784-2763915291-559912524
h323-conf-id: 2851810672-711266784-2763915291-559912524
Content-Length: 109
v=0
o=Sippy 223452192 0 IN IP4 74.205.216.77
s=-
t=0 0
m=audio 33830 RTP/AVP 0
c=IN IP4 74.205.216.777
<------------->
--- (17 headers 6 lines) ---
Sending to 208.65.xxx.xxx : 5060 (NAT)
Using INVITE request as basis request - <a class="moz-txt-link-abbreviated" href="mailto:550D37B3@208.72.xxx.xxx~o">550D37B3@208.72.xxx.xxx~o</a>
Found peer 'FreePhoneLine'
Found RTP audio format 0
[2011-01-27 14:35:18] WARNING[2911]: chan_sip.c:5948 process_sdp_c:
Unable to lookup RTP Audio host in c= line, 'IN IP4 74.205.216.777'
[2011-01-27 14:35:18] WARNING[2911]: chan_sip.c:5741 process_sdp:
Insufficient information in SDP (c=)...
-----------------------------------------------------------------------------------------------------------
It seems in the SIP INVITE, the IP 74.205.216.77 somehow changed to
74.205.216.777.
I am not sure this is a bug of Asterisk or not.
</pre>
</blockquote>
<pre wrap="">
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by <a class="moz-txt-link-freetext" href="http://www.api-digital.com">http://www.api-digital.com</a> --
New to Asterisk? Join us for a live introductory webinar every Thurs:
<a class="moz-txt-link-freetext" href="http://www.asterisk.org/hello">http://www.asterisk.org/hello</a>
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
<a class="moz-txt-link-freetext" href="http://lists.digium.com/mailman/listinfo/asterisk-users">http://lists.digium.com/mailman/listinfo/asterisk-users</a>
</pre>
</blockquote>
</body>
</html>