Re: [asterisk-users] chan_sip bug? (Asterisk 1.4)
Chad, You are right. tcpdump shows Asterisk sees 777 when the packet arrived. 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. 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? *Jian * On 11-01-27 04:31 PM, Chad Wallace wrote: On Thu, 27 Jan 2011 14:52:06 -0800 Jian Gaojian@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:1778xxx@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: Anonymoussip:208.65.xxx.xxx:5061 To:sip:1778...@208.65.xxx.xxx:5060 From:sip:604...@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 -- _ -- 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
[asterisk-users] chan_sip bug? (Asterisk 1.4)
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:1778xxx@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: Anonymoussip:208.65.xxx.xxx:5061 To: sip:1778...@208.65.xxx.xxx:5060 From: sip:604...@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. Regards, Jian -- _ -- 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
Re: [asterisk-users] chan_sip bug? (Asterisk 1.4)
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:1778xxx@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: Anonymoussip:208.65.xxx.xxx:5061 To: sip:1778...@208.65.xxx.xxx:5060 From: sip:604...@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