[Asterisk-Users] Asterisk handling of SIP info

[Wei Su]

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:6002 at 192.168.10.90 SIP/2.0
        Method: INFO
        Resent Packet: False
    Message Header
        Call-ID: 60b8596c-4135c-c0a81e68 at 192.168.10.90
        From: Demo2<sip:6003 at 192.168.10.90;user=phone>;tag=221a0-a1cf
            SIP Display info: Demo2
            SIP from address: sip:6003 at 192.168.10.90
            SIP tag: 221a0-a1cf
        To: <sip:6002 at 192.168.10.90;user=phone>;tag=as6b294484
            SIP to address: sip:6002 at 192.168.10.90
            SIP tag: as6b294484
        CSeq: 102 INFO
        Via: SIP/2.0/UDP 192.168.10.164:5060
        Contact: Demo2<sip:6003 at 192.168.10.164:5060;user=phone>
        Max-Forwards: 70
        Supported: timer
        Proxy-Authorization: Digest
username="6003",realm="asterisk",uri="sip:6002 at 192.168.10.90",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:6003 at 192.168.10.90;user=phone>;tag=221a0-a1cf
            SIP Display info: Demo2
            SIP from address: sip:6003 at 192.168.10.90
            SIP tag: 221a0-a1cf
        To: <sip:6002 at 192.168.10.90;user=phone>;tag=as6b294484
            SIP to address: sip:6002 at 192.168.10.90
            SIP tag: as6b294484
        Call-ID: 60b8596c-4135c-c0a81e68 at 192.168.10.90
        CSeq: 102 INFO
        User-Agent: Asterisk PBX
        Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
        Contact: <sip:6002 at 192.168.10.90>
        Content-Length: 0

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