| Summary: | ASTERISK-00174: SIP From: header not RFC compliant | ||
| Reporter: | John Todd (jtodd) | Labels: | |
| Date Opened: | 2003-08-26 12:39:03 | Date Closed: | 2011-06-07 14:10:31 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Core/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Whitespaces in the From: name in the non-quoted section of the line cause some proxies or phones to report invalid caller ID name. Use of the word "Anonymous" or even "No-CallerID" would solve the problem. ****** ADDITIONAL INFORMATION ****** To: asterisk-users@lists.digium.com From: Jiri Kuthan <jiri@iptel.org> Subject: [Asterisk-Users] bug report: whitespaces in uris Reply-To: asterisk-users@lists.digium.com Date: Tue, 26 Aug 2003 14:04:11 +0200 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 | ||
| Comments: | By: Mark Spencer (markster) 2003-08-27 10:32:05 Can you provide an example of such a call flow? By: Thorsten Lockert (tholo) 2003-08-29 17:32:40 I think he refers to the "sip:No CallID@ip-address" that is generated. Specifically, the space in "No CallID"... Maybe use underline or dash there instead? By: John Todd (jtodd) 2003-09-08 15:23:51 A simple "grep" through the * source tree doesn't find any instances of "No CallID" or even "CallID" (except in comments) Where is this string being generated? By: Thorsten Lockert (tholo) 2003-09-08 21:25:46 I cannot appear to reproduce this problem anymore. At a guess I would say this was probably just a configuration problem. By: John Todd (jtodd) 2003-09-08 22:56:36 This may be due to the Cisco sending the non-RFC compliant string. Dig through your 7960 configuration and see if "No CallerID" is specified somewhere in the configs... I seem to recall seeing that in some sample file... By: Mark Spencer (markster) 2003-09-26 21:18:34 Not an issue with Asterisk | ||