[asterisk-users] chan_sip bug? (Asterisk 1.4)

[Jian Gao]

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 --->
INVITE sip:1778xxxxxxx at 10.11.22.77:5060 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: <sip:1778xxxxxxx at 208.65.xxx.xxx:5060>
From: <sip:604xxxxxxx at 208.65.xxx.xxx:5060>;tag=ixpa27sbhn3inu5x.o
Call-ID: 550D37B3 at 208.72.xxx.xxx~o
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 - 550D37B3 at 208.72.xxx.xxx~o
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.

Regards,

Jian