| Summary: | ASTERISK-27895: chan_pjsip: 'tel' URI is unsupported | ||||
| Reporter: | Abhay Gupta (agupta) | Labels: | pjsip | ||
| Date Opened: | 2018-06-05 04:42:16 | Date Closed: | 2018-06-05 04:53:35 | ||
| Priority: | Minor | Regression? | |||
| Status: | Closed/Complete | Components: | Channels/chan_pjsip | ||
| Versions: | 15.3.0 | Frequency of Occurrence | |||
| Related Issues: |
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| Environment: | Ubuntu 16.04 with Asterisk 15.3.0 using PJSIP | Attachments: | |||
| Description: | This is the message showing issue in handling tel uri as per RFC 3966
<--- Received SIP request (1210 bytes) from UDP:10.232.130.170:5060 ---> INVITE sip:+911244310700@10.126.105.40:5060 SIP/2.0 Via: SIP/2.0/UDP 10.232.130.170:5060;branch=z9hG4bKuw6xxu92yr3uzzx25186w863w;Role=3;Hpt=8fb2_36;TRC=ffffffff-16b9 Call-ID: asbcjy71615rj11ffks55765ps15y5y8jf3s@B.5.203.ims.airtel.in From: "08802809405"<tel:8802809405;noa=subscriber;srvattri=national;phone-context=+91>;tag=c589dd9l To: <tel:+911244310700> CSeq: 1 INVITE Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE Contact: <sip:10.232.130.170:5060;Dpt=eeba-200;Hpt=8fb2_16;CxtId=4;TRC=ffffffff-16b9> Max-Forwards: 65 Supported: timer,100rel,histinfo Session-Expires: 1800 Min-SE: 600 P-Asserted-Identity: <tel:08802809405> P-Charging-Vector: icid-value=AE880F23FCFF6201865132030;orig-ioi=10.232.128.242;term-ioi=SIP_ZOMATO_4310700 Content-Length: 376 Content-Type: application/sdp v=0 o=- 58669091 58669091 IN IP4 10.232.130.179 s=SBC call c=IN IP4 10.232.130.179 t=0 0 m=audio 12422 RTP/AVP 108 102 8 0 18 97 a=rtpmap:108 AMR/8000 a=fmtp:108 mode-change-neighbor=1;mode-change-period=2 a=rtpmap:102 AMR/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:97 telephone-event/8000 a=ptime:20 a=maxptime:20 a=3gOoBTC <--- Transmitting SIP response (475 bytes) to UDP:10.232.130.170:5060 ---> SIP/2.0 416 Unsupported URI Scheme Via: SIP/2.0/UDP 10.232.130.170:5060;received=10.232.130.170;branch=z9hG4bKuw6xxu92yr3uzzx25186w863w;Role=3;Hpt=8fb2_36;TRC=ffffffff-16b9 Call-ID: asbcjy71615rj11ffks55765ps15y5y8jf3s@B.5.203.ims.airtel.in From: "08802809405" <tel:8802809405;phone-context=+91;noa=subscriber;srvattri=national>;tag=c589dd9l To: <tel:+911244310700>;tag=z9hG4bKuw6xxu92yr3uzzx25186w863w CSeq: 1 INVITE Server: Asterisk PBX 15.3.0 Content-Length: 0 | ||||
| Comments: | By: Asterisk Team (asteriskteam) 2018-06-05 04:42:18.236-0500 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: Joshua C. Colp (jcolp) 2018-06-05 04:43:19.396-0500 The chan_pjsip module does not support use of the 'tel' URI. No individual has gone through and audited everything and made sure it works everywhere, and adjusted the code accordingly. Is this something you plan on doing? By: Abhay Gupta (agupta) 2018-06-05 04:51:09.095-0500 As per RFC should the asterisk server support it or not ? This tel uri is coming from Telco and not something that we have created on our routers / switches . By: Joshua C. Colp (jcolp) 2018-06-05 04:52:52.849-0500 Asterisk doesn't implement or support that RFC, so no. I also know of noone actively working on tel URI support. | ||||