Summary:	ASTERISK-30187: chan_sip: Unsupported URI(416)
Reporter:	abhi (abhinav94)	Labels:
Date Opened:	2022-08-23 03:04:00	Date Closed:
Priority:	Major	Regression?	No
Status:	Open/New	Components:	Channels/chan_sip/General
Versions:	16.26.0	Frequency of Occurrence	Frequent
Related Issues:
Environment:	Debian 11	Attachments:
Description:	Hi Team, i am getting unsupported uri on asterisk (416 for incoming call) when calling from landline number without E164 format if we add country code then works fine but when we try to call without  country code the we get this issue. This is very critical issue now can you please suggest what changes i have to do in chan_sip.c for handling this or any other solution for this issue .below i am sharing the asterisk cli log.       > 0x55af3bcc5270 -- Strict RTP learning after remote address set to: 10.238.70.67:36646 [Aug 19 15:43:26] NOTICE[916][C-00010096]: chan_sip.c:18547 get_destination: debug [<tel:01111>] [Aug 19 15:43:26] NOTICE[916][C-00010096]: chan_sip.c:18549 get_destination: PATCH: Using P-Asserted-Identity: [tel:01111] [Aug 19 15:43:26] WARNING[916][C-00010096]: chan_sip.c:18554 get_destination: Not a SIP header (001111)? [Aug 19 15:43:29] NOTICE[916][C-00010097]: chan_sip.c:19487 check_user_full: PATCH tel:SCHEMA : USE  P-Asserted-Identity, Instead [From <tel:>] [Aug 19 16:05:40] NOTICE[916][C-00012e86]: chan_sip.c:19528 check_user_full: From address missing 'sip:', we try to use P-Asserted-Identity instead Please guide asap.
Comments:	By: Asterisk Team (asteriskteam) 2022-08-23 03:04:04.897-0500 The severity of this issue has been automatically downgraded from "Blocker" to "Major". The "Blocker" severity is reserved for issues which have been determined to block the next release of Asterisk. This severity can only be set by privileged users. If this issue is deemed to block the next release it will be updated accordingly during the triage process. By: Asterisk Team (asteriskteam) 2022-08-23 03:04:09.681-0500 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. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. 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]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/]. By: Joshua C. Colp (jcolp) 2022-08-23 03:54:12.558-0500 The chan_sip channel driver is in 'extended' support status and is supported only by community members.  Your issue is in the queue. Your patience is appreciated as a community developer may work the issue when time and resources become available. Asterisk is an open source project and community members work the issues on a voluntary basis. You are welcome to develop your own patches and submit them to the project.[1] If you are not a programmer and you are in a hurry to see a patch provided then you might try rallying support on the Asterisk users mailing list or forums.[2] Another alternative is offering a bug bounty on the asterisk-dev mailing list.[3] Often a little incentive can go a long way. [1]: https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process [2]: http://www.asterisk.org/community/discuss [3]: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Bug+Bounties By: abhi (abhinav94) 2022-08-23 04:30:07.292-0500 Hi Team, For this 416 and incoming call disconnect issue, where i can make changes for add PAI header with tel uri in chan_sip.c code? By: N A (InterLinked) 2022-08-23 06:53:19.306-0500 abhi: I don't think is really a bug, it would be a new feature, since I don't think chan_sip has ever supported tel URIs. As such, such a feature would never go into chan_sip mainstream since only bugs and security fixes are accepted for chan_sip. It does not sound like you are familiar with the Asterisk source so you would probably need to put a bounty on this for someone to add it for you, and there is no guarantee either way. Basic tel URI support is currently being added to PJSIP. You should investigate whether this suits your needs and, if so, use PJSIP instead of SIP. By: abhi (abhinav94) 2022-08-23 07:06:55.172-0500 Dear Sir/Mam, I have already check with pjsip and tel uri doesn't supported with pjsip that's why i shifted sip from pjsip and i have added patch for supporting tel uri but in some landline number i am facing below 416 unsupported uri error on asterisk while other request from tel uri is perfectly working with same asterisk version. > 0x55af3bcc5270 – Strict RTP learning after remote address set to: 10.238.70.67:36646 [Aug 19 15:43:26] NOTICE[916][C-00010096]: chan_sip.c:18547 get_destination: debug [<tel:01111>] [Aug 19 15:43:26] NOTICE[916][C-00010096]: chan_sip.c:18549 get_destination: PATCH: Using P-Asserted-Identity: [tel:01111] [Aug 19 15:43:26] WARNING[916][C-00010096]: chan_sip.c:18554 get_destination: Not a SIP header (001111)? [Aug 19 15:43:29] NOTICE[916][C-00010097]: chan_sip.c:19487 check_user_full: PATCH tel:SCHEMA : USE P-Asserted-Identity, Instead [From <tel:>] [Aug 19 16:05:40] NOTICE[916][C-00012e86]: chan_sip.c:19528 check_user_full: From address missing 'sip:', we try to use P-Asserted-Identity instead By: abhi (abhinav94) 2022-08-23 07:10:02.466-0500 Dear Sir/mam, Is it possible to solve this issue with asterisk ( add PAI header with tel uri from header)?