[Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 152

2005-03-22 Thread Wei Su
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: Demo2sip:[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: Demo2sip:[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: Demo2sip:[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
 
 


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Re: [Asterisk-Users] RE: Asterisk-Users Digest, Vol 8, Issue 152

2005-03-22 Thread Kevin P. Fleming
Wei Su wrote:
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?
It is not required to do anything with INFO requests at all, it can 
ignore them or reject them if it wants to.

It is possible to bridge the request to the other peer, but keep in mind 
that the other peer might not be SIP at all (IAX2, Zaptel, an 
application, H.323, etc.) Since Asterisk is acting as a PBX, it's free 
to choose what to do with them, if anything at all.

Right now Asterisk only handles one type of SIP INFO packet: the ones 
used for DTMF transmission.
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