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

[Chad Wallace]

On Thu, 27 Jan 2011 14:52:06 -0800
Jian Gao <jian.gao at sjgeophysics.com> wrote:

> 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 --->

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.


> 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.