Dan Zhou
2005-Feb-06 14:01 UTC
[Asterisk-Users] SIP URI modified unexpectedly! Is that a router problem?
Hi, I set up an asterisk server at my home computer, and the * box is configured as the DMZ of my ADSL modem/router. I found the SIP URI in an INVITE message has been changed before it reaches the * server. In my setup, I have a SJPhone software installed in with a public IP yy.yy.yy.yy. (ph no. 10930) I have a unknown brand SIP phone in the same network as the * sever is in. The public IP is xx.xx.xx.xx, phone no =10916. Here is the SJphone Log of an INVITE message sent from 10930 to 10916. 10:42:05 INFO Initiating SIP call to sip:10916@xx.xx.xx.xx:5060 10:42:05 DEBUG 2005-02-05 21:42:05.937 UDP LOCAL->xx.xx.xx.xx:5060 INVITE sip:10916@xx.xx.xx.xx:5060 SIP/2.0 Content-Length: 341 Contact: <sip:10930@yy.yy.yy.yy:5060> Call-ID: C64037DF-A3BA-4F1D-8FE9-84FA9D3DBD85@yy.yy.yy.yy Content-Type: application/sdp From: "Dan"<sip:10930@xx.xx.xx.xx:5060>;tag=588746825471 CSeq: 1 INVITE Max-Forwards: 70 To: <sip:10916@xx.xx.xx.xx:5060> Via: SIP/2.0/UDP yy.yy.yy.yy;rport;branch=z9hG4bKcb617ac20131c9b142053dad0000798300000032 User-Agent: SJLabs-SJphone/1.40.258 --snip--- Here is the corresponding message I received in my server, the output of "ngrep 10930 port 5060 -d eth0" . ### U yy.yy.yy.yy:5060 -> 192.168.1.2:5060 INVITE sip:192.168.1.2@xx.xx.xx.xx:5060 SIP/2.0..Content-Length: 341..Co ntact: <sip:10930@yy.yy.yy.yy:5060>..Call-ID: C64037DF-A3BA-4F1D-8FE9-84 FA9D3DBD85@yy.yy.yy.yy..Content-Type: application/sdp..From: "Dan"<sip:1 0930@xx.xx.xx.xx:5060>;tag=588746825471..CSeq: 1 INVITE..Max-Forwards: 7 0..To: <sip:192.168.1.2@xx.xx.xx.xx:5060>..Via: SIP/2.0/UDP 203.97.122.1 94;rport;branch=z9hG4bKcb617ac20131c9b142053dad0000798300000032..User-Agent : SJLabs-SJphone/1.40.258....v=0..o=- 3316628525 3316628525 IN IP4 203.97.1 22.194..s=SJphone..c=IN IP4 yy.yy.yy.yy..t=0 0..a=direction:active..m=au dio 16386 RTP/AVP 0 8 3 97 98 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/80 00..a=rtpmap:3 GSM/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:98 iLBC/8000..a=fm tp:98 mode=20..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-11,16.. # Obviously, they are two places in the received messages changed (the INVITE tag and the "To" header), from <sip:10916@xx.xx.xx.xx:5060> to sip:192.168.1.2@xx.xx.xx.xx:5060 As a result, I got a 404 not found error. Before the error, there is one line in the CLI output saying: Looking for 192.169.1.2 in local ... I think it should have looked for 10916 in local (the context defined in extension.conf). Has anyone here experienced similar problem? Can I say my router is not VoIP friendly? Cheers, Dan _________________________________________________________________ Don’t just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/