TEL URIs test results on Asterisk version 18.15.0-rc1

Hi guys, would like to provide the test result for Tel URIs on sip trunk from my local telco. There are 2 parts here, one is working sample and the other is not. The one which is not working seems like the “Invite” portion having sip uri and the “To” having tel, system response with 401 Unauthorized. I’m not sure why local telco sending 2 different type of invite uri. Below is the screen shot of the sip trace.

One may be associated with an endpoint that is challenging for authentication, or not matching at all. You’d need to provide the associated PJSIP configuration as well (outbound registration, endpoint, identify).

I don’t think this is strictly due to tel URI, but more likely your configuration.

Alright, attach the configuration files. For your information, the not working sample number is the actual registration number, those working one are the DIDs.
pjsip.endpoint.txt (668 Bytes)
pjsip.identify.txt (137 Bytes)
pjsip.registration.txt (471 Bytes)


From the endpoint. You are trying to authenticate them, and it is unlikely they support that.

I’m also questioning whether your identify is actually working and if it’s actually landing as an anonymous call instead. A console log would confirm what actually happened with the working case.

After removed the authentication from the endpoint, it’s working right now! Thanks for that :wink:
Apparently the telco send caller id using the Contact header which i already mentioned in another post, that’s the only issue I have right now still didn’t find any solution to it.

@david551 gave you the “solution”. Contact isn’t meant for callerid, the fact it’s there is just an implementation detail so it’s up to you to use the available dialplan stuff to get it and treat it as callerid.

Sure will look into that solution, thanks again for helping :handshake:

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