Summary: | ASTERISK-28174: PJSIP issue with TEL (RFC 3966) | ||
Reporter: | ast (ast@nashdl.com) | Labels: | pjsip |
Date Opened: | 2018-11-20 06:21:51.000-0600 | Date Closed: | 2018-11-20 06:28:06.000-0600 |
Priority: | Critical | Regression? | Yes |
Status: | Closed/Complete | Components: | pjproject/pjsip Resources/res_pjsip_session |
Versions: | 16.0.0 | Frequency of Occurrence | Constant |
Related Issues: | |||
Environment: | Ubuntu 18.0.1 LT | Attachments: | |
Description: | Essentially set up pjsip.conf to register to ITSP and made an endpoint.
Received response advising that TEL is unknown or at least no SIP, actual SIP phrase is "416 Unsupported URI Scheme". PJSIP Logging enabled <--- Received SIP request (1697 bytes) from UDP:210.49.225.101:5060 ---> INVITE sip:+6173210321@192.168.1.100:5060;line={weirdnumbers} SIP/2.0 Max-Forwards: 65 Via: SIP/2.0/UDP 210.49.225.101:5060;branch={weirdnumbers} To: "SIPLineUser SIPLineUser" <tel:+6173210321> From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag={weirdnumbers} Call-ID: {weirdnumbers}@10.194.0.25 CSeq: 1 INVITE Contact: <sip:sgc_c@210.49.225.101;transport=udp> Record-Route: <sip:210.49.225.101;transport=udp;lr> Min-Se: 900 Privacy: id Session-Expires: 1800 Supported: com.nortelnetworks.firewall Supported: p-3rdpartycontrol Supported: nosec Supported: join Supported: x-nortel-sipvc Supported: gin Supported: com.nortelnetworks.im.encryption Supported: 100rel Supported: resource-priority Supported: replaces User-Agent: Nortel SESM 19.0.1.0 Content-Type: application/sdp Content-Length: 574 X-Nt-Service: brdplayed=yes X-Nt-Corr-Id: 40bc80137637c1f4413c5808d06ff1b42a3bacca3@10.194.0.25 X-Nortel-Profile: DEFAULT Allow: ACK, BYE, CANCEL, INVITE, OPTIONS, INFO, SUBSCRIBE, REFER, NOTIFY, PRACK, UPDATE v=0 o=- 3257871432 3257871432 IN IP4 210.49.225.101 s=- e=unknown@invalid.net c=IN IP4 210.49.123.41 t=0 0 m=audio 48426 RTP/AVP 8 0 18 101 110 111 b=AS:80 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=ptime:20 a=maxptime:20 a=3gOoBTC a=rtpmap:110 AMR/8000 a=fmtp:110 mode-change-period=2; mode-change-capability=2; mode-change-neighbor=1; max-red=0 a=rtpmap:111 AMR/8000 a=fmtp:111 octet-align=1; mode-change-period=2; mode-change-capability=2; mode-change-neighbor=1; max-red=0 <--- Transmitting SIP response (534 bytes) to UDP:210.49.225.101:5060 ---> SIP/2.0 416 Unsupported URI Scheme Via: SIP/2.0/UDP 210.49.225.101:5060;received=210.49.225.101;branch={weirdnumbers} Record-Route: <sip:210.49.225.101;transport=udp;lr> Call-ID: {weirdnumbers}@10.194.0.25 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag={weirdnumbers} To: "SIPLineUser SIPLineUser" <tel:+6173210321>;tag={weirdnumbers} CSeq: 1 INVITE Server: "{User_Agent}" Content-Length: 0 <--- Received SIP request (489 bytes) from UDP:210.49.225.101:5060 ---> ACK sip:+6173210321@192.168.2.22:5060;line={weirdnumbers} SIP/2.0 Max-Forwards: 70 Via: SIP/2.0/UDP 210.49.225.101:5060;received=210.49.225.101;branch={weirdnumbers} To: "SIPLineUser SIPLineUser" <tel:+6173210321>;tag={weirdnumbers} From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag={weirdnumbers} Call-ID: {weirdnumbers}@10.194.0.25 CSeq: 1 ACK Content-Length: 0 lappy*CLI> pjsip set logger off PJSIP Logging disabled | ||
Comments: | By: Asterisk Team (asteriskteam) 2018-11-20 06:21:53.919-0600 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: Joshua C. Colp (jcolp) 2018-11-20 06:28:07.110-0600 The tel URI is not currently supported in PJSIP and noone is actively working on it. If you'd like to provide a fully tested patch with test coverage, then this could be reopened otherwise this is considered a new feature request. By: ast (ast@nashdl.com) 2018-11-20 07:14:46.716-0600 Thank you, any suggestions on where to start in the code or where to look for some developer documentation so i can understand how the functions fit to gether? By: Asterisk Team (asteriskteam) 2018-11-20 07:14:46.869-0600 This issue has been reopened as a result of your commenting on it as the reporter. It will be triaged once again as applicable. By: ast (ast@nashdl.com) 2018-11-20 07:26:34.103-0600 Raised as a new feature request. |