On Thu, 27 Jan 2011 14:52:06 -0800 Jian Gao <jian....@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@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:1778xxxx...@208.65.xxx.xxx:5060> > From: <sip:604xxxx...@208.65.xxx.xxx:5060>;tag=ixpa27sbhn3inu5x.o > Call-ID: 550d3...@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 - 550d3...@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. -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users