I understand Asterisk is more like a B2BUA. But when this INFO request is sent to asterisk, asterisk is supposed to bridge the request to the other endpoint, right? In what situation, it decides to send a reply; in what situation, it decides to bridge the request?
What is the role of gateway in SIP world, a proxy, a B2BUA or something else? Thank you, Wei Date: Fri, 18 Mar 2005 12:51:28 -0600 From: Eric Wieling <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] Asterisk handling of SIP info To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii; format=flowed Asterisk is not a SIP proxy. Wei Su wrote: > We encouter a situation where we need to use SIP info to convey infomation > for one end point to another endpoint. I use asterisk to do the test and > find asterisk does not forward the SIP info to another endpoint, but act as > UAS and returns a 4xx error message. I think asterisk is not right to handle > this SIP info message. > > In RFC 3261 Page 70 "This protocol is designed to be extended. Future > extensions may define new methods and header fields at any time. An element > MUST NOT refuse to proxy a request becasue it contains a method or header > field it does not know about". In this case, asterisk does not understand > this INFO message, so it acts as a UAS instead of proxy. > > How to let asterisk just forward this request to the other endpoint and > instead processing it as a UAS? > > Thank you, > > Wei > > > > > Here is the log from the asterisk server: > > Mar 17 12:01:31 WARNING[2804]: chan_sip.c:6134 receive_info: Unable to parse > INFO message > > > Here is the trace: > > > Frame 96 (808 bytes on wire, 808 bytes captured) > Session Initiation Protocol > Request-Line: INFO sip:[EMAIL PROTECTED] SIP/2.0 > Method: INFO > Resent Packet: False > Message Header > Call-ID: [EMAIL PROTECTED] > From: Demo2<sip:[EMAIL PROTECTED];user=phone>;tag=221a0-a1cf > SIP Display info: Demo2 > SIP from address: sip:[EMAIL PROTECTED] > SIP tag: 221a0-a1cf > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as6b294484 > SIP to address: sip:[EMAIL PROTECTED] > SIP tag: as6b294484 > CSeq: 102 INFO > Via: SIP/2.0/UDP 192.168.10.164:5060 > Contact: Demo2<sip:[EMAIL PROTECTED]:5060;user=phone> > Max-Forwards: 70 > Supported: timer > Proxy-Authorization: Digest > username="6003",realm="asterisk",uri="sip:[EMAIL PROTECTED]",response="034d > 6b15ec1b2fa91f59c55d51c0a8e7",nonce="70c7fe86" > Content-Type: application/media_control+xml > Content-Length: 195 > Message body > <?xml version="1.0" encoding="utf-8" ?>\n > <media_control>\n > <vc_primitive>\n > <to_encoder>\n > <picture_fast_update>\n > </picture_fast_update>\n > </to_encoder>\n > </vc_primitive>\n > </media_control> > > > Frame 97 (430 bytes on wire, 430 bytes captured) > Session Initiation Protocol > Status-Line: SIP/2.0 415 Unsupported media type > Status-Code: 415 > Resent Packet: False > Message Header > Via: SIP/2.0/UDP 192.168.10.164:5060 > From: Demo2<sip:[EMAIL PROTECTED];user=phone>;tag=221a0-a1cf > SIP Display info: Demo2 > SIP from address: sip:[EMAIL PROTECTED] > SIP tag: 221a0-a1cf > To: <sip:[EMAIL PROTECTED];user=phone>;tag=as6b294484 > SIP to address: sip:[EMAIL PROTECTED] > SIP tag: as6b294484 > Call-ID: [EMAIL PROTECTED] > CSeq: 102 INFO > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: <sip:[EMAIL PROTECTED]> > Content-Length: 0 > > _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users