FYI: Asterisk puts URIs in messages which violates the SIP spec and
can't be accepted by URI parsers: username includes a whitespace.
See for example the From header field. Attached is example of an
incorrect message and related parts of RFC3261 specification.
(Who doesn't want to dig into parser details may want to realize
that whitespaces are used as uri delimitors in first request
line and can't thus be a uri part.)
I would recommend that the stack generally validates URIs for
such glitches and uses other word for "no callId".
"anonymous"
is in frequent use by other software.
-jiri
OPTIONS sip:195.37.77.101 SIP/2.0
Via: SIP/2.0/UDP 24.172.18.166:5060;branch=z9hG4bK03be4cf3
From: "No CallID" <sip:No CallID@24.172.18.166>;tag=as2746f4f3
To: <sip:195.37.77.101>
Contact: <sip:No CallID@24.172.18.166>
Call-ID: 72b6aaf63319c64e4a96a6cd42245f7e@24.172.18.166
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0
3261:
From->name_addr|addr_spec
addr_spec->SIP_URI
SIP_URI->userinfo
user_info->user
user->1*( unreserved / escaped / user-unreserved
user-unreserved = "&" / "=" / "+" /
"$" / "," / ";" / "?" / "/"
unreserved = alphanum / mark
mark = "-" / "_" / "." / "!" /
"~" / "*" / "'"
/ "(" / ")"
--
Jiri Kuthan http://iptel.org/~jiri/
iptel.org -- creaters of the fastest SIP server
