Hello, I was interconnecting Asterisk (v1.0) with a strict router (ie, no ;lr in routes) and I think I found a bug in the way Asterisk prepare new requests inside a dialog. I'm sending some captures (ngrep) along with my comments. This is a 200 OK (INVITE) received by Asterisk ========================U 2005/02/10 16:41:55.065538 143.173.202.82:5060 -> 143.173.202.83:5070 SIP/2.0 200 OK..Via: SIP/2.0/UDP 143.173.202.83:5070;branch=z9hG4bK42c78895..Record-Route: <sip:001178612341106@143.173. 202.81;ftag=as182aa61c;lr>..Record-Route: <sip:001178612341106@143.173.202.82:5060>..From: "Call Center 1" <sip:55512@si p.trdc.telenova.com.br>;tag=as182aa61c..To: <sip:001178612341106@sip.com>;tag=281B1720-1D3C..Call-ID: 3 e3b393c407188e9311d01a03e43f144@sip.com..CSeq: 103 INVITE..Contact: <sip:747#1178612341106@143.173.194. 37:5060>..date: Thu, 10 Feb 2005 20:41:43 GMT..server: Cisco-SIPGateway/IOS-12.x..allow-events: telephone-event..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO..Content-Type: application/sdp..Conten t-Length: 257....v=0..o=CiscoSystemsSIP-GW-UserAgent 2591 2076 IN IP4 143.173.194.37..s=SIP Call..c=IN IP4 143.173.19 4.37..t=0 0..m=audio 19458 RTP/AVP 18 100..c=IN IP4 143.173.194.37..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpma p:100 X-NSE/8000..a=fmtp:100 192-194.. ================================ At this point Asterisk builds a route list for subsequent requests (record routes + contacts): sip:001178612341106@143.173.202.82:5060 sip:001178612341106@143.173.202.81;ftag=as182aa61c;lr sip:747#1178612341106@143.173.194.37:5060 But when Asterisk sends a BYE for this call: ======================================U 2005/02/10 16:41:57.400529 143.173.202.83:5070 -> 143.173.202.82:5060 BYE sip:747#1178612341106@143.173.194.37:5060 SIP/2.0..Via: SIP/2.0/UDP 143.173.202.83:5070;branch=z9hG4bK67ce783e..Rout e: <sip:001178612341106@143.173.202.81;ftag=as182aa61c;lr>,<sip:747#1178612341106@143.173.194.37:5060>..From: "Call Cente r 1" <sip:55512@sip.com>;tag=as182aa61c..To: <sip:001178612341106@sip.com>;tag=281B172 0-1D3C..Contact: <sip:55512@143.173.202.83:5070>..Call-ID: 3e3b393c407188e9311d01a03e43f144@sip.com..C Seq: 104 BYE..User-Agent: Asterisk PBX..Proxy-Authorization: Digest username="55512", realm="sip.com", algorithm=MD5, uri="sip:747#1178612341106@143.173.194.37:5060", nonce="420bc833181bca085f953885ace7fc72c107c8e0", respo nse="75fcd1bf185ae29becd0b4f715f5cb17", opaque=""..Content-Length: 0.... ========================================== As we can see it is using the contact as the URI of this request, which also appears as the last route header. As per RFC3261 (16.12.1) if the next hop is a strict router (no ;lr in route header) it should use that information as the R-URI, which is not the behaviour of Asterisk. Moreover, it should include all routes learned in the request Route header (including the one that will receive the request) when we are treating with loose routers. I have changed the source code and in my test bed (at least) is working fine. Am I missing something? Regards, Chuck. __________________________________ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250