Re: [asterisk-dev] [Code Review] 3349: Implement RFC-3966 TEL URI incoming INVITE
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/ --- (Updated March 22, 2014, 2:08 p.m.) Review request for Asterisk Developers, Corey Farrell, lmadensen, Matt Jordan, and wdoekes. Changes --- When an incoming RFC 3966 TEL URI INVITE call does not contain a global number nor a phone-context, we still return the local number, but we issue a warning message. This could incur that local calls could be possible, but outgoing return calls might fail because there might be no route to the (original) caller. Bugs: ASTERISK-17179 https://issues.asterisk.org/jira/browse/ASTERISK-17179 Repository: Asterisk Description --- 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) - /trunk/channels/sip/reqresp_parser.c 410429 /trunk/channels/chan_sip.c 410429 Diff: https://reviewboard.asterisk.org/r/3349/diff/ Testing --- 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 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3349: Implement RFC-3966 TEL URI incoming INVITE
--- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/#review11323 --- I don't see what happens with the phone-context argument. Shouldn't we pass that on as a channel variable? - Olle E Johansson On March 22, 2014, 2:08 p.m., Geert Van Pamel wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/ --- (Updated March 22, 2014, 2:08 p.m.) Review request for Asterisk Developers, Corey Farrell, lmadensen, Matt Jordan, and wdoekes. Bugs: ASTERISK-17179 https://issues.asterisk.org/jira/browse/ASTERISK-17179 Repository: Asterisk Description --- 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 - /trunk/channels/sip/reqresp_parser.c 410429 /trunk/channels/chan_sip.c 410429 Diff: https://reviewboard.asterisk.org/r/3349/diff/ Testing --- 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 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3349: Implement RFC-3966 TEL URI incoming INVITE
On March 21, 2014, 7:01 p.m., Corey Farrell wrote: /trunk/channels/sip/reqresp_parser.c, line 130 https://reviewboard.asterisk.org/r/3349/diff/7-8/?file=56285#file56285line130 This needs to blank both variables: userinfo = uri = ; Geert Van Pamel wrote: We return the local number anyway when an incoming RFC 3966 TEL URI INVITE call does not contain a global number nor a phone-context. Corey Farrell wrote: First sentence of 3rd paragraph of section 5.1.5: Local numbers MUST have a 'phone-context' parameter that identifies the scope of their validity. Note the word MUST, this has specific meaning in RFC's. I will not approve this review if it's going to contradict the RFC it's claiming to implement. You have to be strict in what you send, but open for receiving stuff that doesn't always follow the RFC. We can add an option that sets strictness. I haven't seen many implementations of Tel: uri's sadly, but many of the few did not follow the RFC. - Olle E --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/#review11314 --- On March 22, 2014, 2:08 p.m., Geert Van Pamel wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/ --- (Updated March 22, 2014, 2:08 p.m.) Review request for Asterisk Developers, Corey Farrell, lmadensen, Matt Jordan, and wdoekes. Bugs: ASTERISK-17179 https://issues.asterisk.org/jira/browse/ASTERISK-17179 Repository: Asterisk Description --- 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 - /trunk/channels/sip/reqresp_parser.c 410429 /trunk/channels/chan_sip.c 410429 Diff: https://reviewboard.asterisk.org/r/3349/diff/ Testing --- 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 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3349: Implement RFC-3966 TEL URI incoming INVITE
On March 21, 2014, 2:01 p.m., Corey Farrell wrote: /trunk/channels/sip/reqresp_parser.c, line 130 https://reviewboard.asterisk.org/r/3349/diff/7-8/?file=56285#file56285line130 This needs to blank both variables: userinfo = uri = ; Geert Van Pamel wrote: We return the local number anyway when an incoming RFC 3966 TEL URI INVITE call does not contain a global number nor a phone-context. Corey Farrell wrote: First sentence of 3rd paragraph of section 5.1.5: Local numbers MUST have a 'phone-context' parameter that identifies the scope of their validity. Note the word MUST, this has specific meaning in RFC's. I will not approve this review if it's going to contradict the RFC it's claiming to implement. Olle E Johansson wrote: You have to be strict in what you send, but open for receiving stuff that doesn't always follow the RFC. We can add an option that sets strictness. I haven't seen many implementations of Tel: uri's sadly, but many of the few did not follow the RFC. If that is the case then should we not return error = -1? As for optional strictness maybe use sip_settings.pedanticsipchecking? - Corey --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/#review11314 --- On March 22, 2014, 9:08 a.m., Geert Van Pamel wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/ --- (Updated March 22, 2014, 9:08 a.m.) Review request for Asterisk Developers, Corey Farrell, lmadensen, Matt Jordan, and wdoekes. Bugs: ASTERISK-17179 https://issues.asterisk.org/jira/browse/ASTERISK-17179 Repository: Asterisk Description --- 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 - /trunk/channels/sip/reqresp_parser.c 410429 /trunk/channels/chan_sip.c 410429 Diff: https://reviewboard.asterisk.org/r/3349/diff/ Testing --- 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 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3349: Implement RFC-3966 TEL URI incoming INVITE
On March 22, 2014, 4:39 p.m., Olle E Johansson wrote: I don't see what happens with the phone-context argument. Shouldn't we pass that on as a channel variable? We return this into the hostport. - Geert --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/#review11323 --- On March 22, 2014, 2:08 p.m., Geert Van Pamel wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/ --- (Updated March 22, 2014, 2:08 p.m.) Review request for Asterisk Developers, Corey Farrell, lmadensen, Matt Jordan, and wdoekes. Bugs: ASTERISK-17179 https://issues.asterisk.org/jira/browse/ASTERISK-17179 Repository: Asterisk Description --- 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 - /trunk/channels/sip/reqresp_parser.c 410429 /trunk/channels/chan_sip.c 410429 Diff: https://reviewboard.asterisk.org/r/3349/diff/ Testing --- 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 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3349: Implement RFC-3966 TEL URI incoming INVITE
On March 22, 2014, 4:39 p.m., Olle E Johansson wrote: I don't see what happens with the phone-context argument. Shouldn't we pass that on as a channel variable? Geert Van Pamel wrote: We return this into the hostport. According to RFC 3966 phone-context is either a domain-name, or (part of) an international telephone number (indicated with +prefix). It is used by a gateway to know how to dial the local number... the local number must be unique within its context... - Geert --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/#review11323 --- On March 22, 2014, 2:08 p.m., Geert Van Pamel wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/ --- (Updated March 22, 2014, 2:08 p.m.) Review request for Asterisk Developers, Corey Farrell, lmadensen, Matt Jordan, and wdoekes. Bugs: ASTERISK-17179 https://issues.asterisk.org/jira/browse/ASTERISK-17179 Repository: Asterisk Description --- 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 - /trunk/channels/sip/reqresp_parser.c 410429 /trunk/channels/chan_sip.c 410429 Diff: https://reviewboard.asterisk.org/r/3349/diff/ Testing --- 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 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
Re: [asterisk-dev] [Code Review] 3349: Implement RFC-3966 TEL URI incoming INVITE
On March 15, 2014, 7:04 p.m., Corey Farrell wrote: /trunk/channels/sip/reqresp_parser.c, line 111 https://reviewboard.asterisk.org/r/3349/diff/3/?file=56070#file56070line111 scheme is the input parameter listing acceptable schemes, we don't need to see it here. The other ast_debug included scheme since the problem was a failure to match the uri with any scheme. I fixed this - Geert --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/#review11227 --- On March 22, 2014, 2:08 p.m., Geert Van Pamel wrote: --- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3349/ --- (Updated March 22, 2014, 2:08 p.m.) Review request for Asterisk Developers, Corey Farrell, lmadensen, Matt Jordan, and wdoekes. Bugs: ASTERISK-17179 https://issues.asterisk.org/jira/browse/ASTERISK-17179 Repository: Asterisk Description --- 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 - /trunk/channels/sip/reqresp_parser.c 410429 /trunk/channels/chan_sip.c 410429 Diff: https://reviewboard.asterisk.org/r/3349/diff/ Testing --- 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 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 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
