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Summary:ASTERISK-27045: Issue with Parsing Contact Header without Brackets and with additional HeaderParameters seperated with semicolon
Reporter:balamurugan (balam)Labels:chan-sip-message-parsing
Date Opened:2017-06-08 17:07:16Date Closed:
Priority:MajorRegression?
Status:Open/NewComponents:Channels/chan_sip/General Resources/res_pjsip
Versions:13.18.4 Frequency of
Occurrence		Constant
Related
Issues:
is related toASTERISK-21726 Asterisk does not properly parse multiple allow: headers
is related toASTERISK-16954 Caller Name does not preserve multiple spaces
is related toASTERISK-17104 [patch] Uri encoded Refer-To fails to match callid, attended transfer fails
is related toASTERISK-17305 Caller name does not preserve "(" parenthese in the from field
is related toASTERISK-17944 Display names with '\' characters near quotes make Asterisk fail to parse the quotation marks correctly.
is related toASTERISK-24433 Diversion header is only partially parsed into REDIRECTING structure
is related toASTERISK-22932 [patch] - SIP Channel fails to parse refer_to_domain
is related toASTERISK-26461 chan_sip: SIP MESSAGE body parsed incorrectly (as headers) if first character is a space (0x20)
Environment:Attachments:
Description:when we handle or Parse Contact Header , if it is presented without brackets

I get a INVITE with
Contact:sip:p65549t0000000m112562c591000000@10.196.0.111:5089;+g.3gpp.accesstype="cellular";+sip.instance="<urn:gsma:imei:3561119000-996900-0>"

currently this is getting parsed incorrectly based on the closed brackets and we end up storing the fullcontact with incomplete URI (metnioned below) and same is sent in the BYE REQURI.

sip:p65549t0000000m112562c591000000@10.196.0.111:5089;+g.3gpp.accesstype="cellular";+sip.instance="<urn:gsma:imei:3561119000-996900-0
Comments:By: Asterisk Team (asteriskteam) 2017-06-08 17:07:17.369-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].
By: George Joseph (gjoseph) 2017-06-09 07:24:26.285-0500

This will affect pjsip as well