Hello, we’re using Debian’s Asterisk package with PJSIP (16.2.1~dfsg-1+deb10u2). Lately we receive some INVITE messages from the ITSP containing tel URIs in the TO header. Asterisk replies with “416 Unsupported URI Scheme”. Using tel URIs seems to be okay in terms of the RFC. But is it really c…

It’s not that common. It’s technically both Asterisk and PJSIP that handle it, but Asterisk doesn’t have the support for it. There is no debugging really required - they’re just not supported in chan_pjsip.

This was discussed recently: [image] Why are chan_sip and res_pjsip with the same access data results in an error message Asterisk SIP The following problem in the PBX I have noticed and it is the res_pjsip which outputs a error in the case of incoming calls with the follo…

Thanks for your hint. In fact, it’s the same ITSP we’re having this issue with. This came up, when they changed some of their hardware a couple weeks ago. They might fix it.

You described on the other thread why it’s complicated to fix this issue. Just to make sure I understood: It’s not just an Asterisk issue, it’s rather an issue within “upstream” PJSIP. Right? So their claim of supporting RFC3966 is misleading? Switching to chan_sip should do the trick? Or is it the…

It’s an Asterisk issue. Asterisk uses the URIs, and because of that it has to be aware so it can treat them differently. As this isn’t done requests with tel URI are blocked. The chan_sip may or may not work with tel URIs, it may throw warnings or errors. Requests don’t show up in logs for this, the…

I totally agree, if that tel URI schema isn’t used a lot, then it’s probably not worth screwing up Asterisk for that. But still I’d appreciate if the logging would be extended. Usually you don’t trace the SIP messages, as far as I know for performance reasons. But then there is no other way of tell…

Asterisk 13 can handle such calls using chan_sip channel driver. From: <tel:0701XXXXXX;phone-context=+40>;tag=xgjix183-CC-147 To: sip:+40213XXXXXX@as1.romtelecom.net;user=phone At the time of installation, it was the only version that had support for tel:URI. The version used was asterisk-certifi…

Hi Even asterisk asterisk-certified-13.18-cert2. did not help to accept incoming call from Huawei SoftX3000 PBX. The invite was: <--- SIP read from UDP:192.168.11.2:5060 ---> INVITE sip:+123456789@myid.telecom.au SIP/2.0 Via: SIP/2.0/UDP 192.168.11.2:5060;branch=z9hG4bKk013kj2osqkfpihfgsoo3…

The first two messages look like reasonable reactions to tel:. You should look to see if the authentication is simply failing before blaming this on tel: Note that cert versions should be avoided by open source users. In particular, they do not load chan_sip, because it is not included in Sangoma…

Hi David. Yes, of cause I’ve looked into authentication before blaming on tel: its all about implementation of RFC-3966, which were never completed in asterisk untill now, https://raw.githubusercontent.com/asterisk/asterisk/master/channels/chan_sip.c , ‘Empty domain name in FROM header’ messa…

Handling of tel URIs

Asterisk Asterisk SIP

david551 February 15, 2021, 6:07pm 3

This was discussed recently:

Topic		Replies	Views
Uri tel sheme not supported Asterisk SIP	9	497	July 16, 2022
PJSIP issue with TEL (RFC 3966) Asterisk SIP	8	3408	November 20, 2018
TEl URI on Asterisk Asterisk SIP	3	1069	December 16, 2021
Why are chan_sip and res_pjsip with the same access data results in an error message Asterisk SIP	13	1430	February 28, 2021
Reception of a tel URI Asterisk Support	1	349	January 15, 2013

Handling of tel URIs

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