[asterisk-dev] [Code Review] 3349: Implement RFC-3966 TEL URI incoming INVITE

[Geert Van Pamel]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3349/
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(Updated March 20, 2014, 10:59 p.m.)


Review request for Asterisk Developers, Corey Farrell, lmadensen, Matt Jordan, and wdoekes.


Changes
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* Enhanced the description of the TEL URI parameter parsing algorithm
* Corrected regression tests:
- Added a 3rd (complex) TEL URI invite with parameters and headers
- Rolled back to test case name "sip_uri_parse_test"
- Rolled back to ast_strlen_zero check on invalid TEL URI check


Bugs: ASTERISK-17179
    https://issues.asterisk.org/jira/browse/ASTERISK-17179


Repository: Asterisk


Description
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Implements RFC-3966 TEL URI incoming INVITE.

See https://issues.asterisk.org/jira/browse/ASTERISK-17179 for a description of the original isssue.

I have been patching all versions since Asterisk 1.6. I would like to include the code into the main trunk for version 13.

Previously Asterisk was failing with error on incoming IMS call:

Nov 13 17:52:05 NOTICE[27459]: chan_sip.c:6973 check_user_full: From address missing 'sip:', using it anyway

Nov 13 17:52:05 WARNING[27459]: chan_sip.c:6525 get_destination: Huh? Not a SIP header (tel:0987654321;phone-context=+32987654321)?

Reason: tel: protocol was not recognized.


Diffs (updated)
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  /trunk/channels/sip/reqresp_parser.c 410429 
  /trunk/channels/chan_sip.c 410429 

Diff: https://reviewboard.asterisk.org/r/3349/diff/


Testing
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Executed an incoming TEL URI INVITE connection.
CLI was present on the display and in the CDR file.
No errors on SIP debug output.


File Attachments
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RFC-3966 tel URI patch
  https://reviewboard.asterisk.org/media/uploaded/files/2014/03/13/cad7a996-88c1-47fe-a2a9-cc6987af3b75__rfc-3966-tel-uri-patch-diff.txt


Thanks,

Geert Van Pamel

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