Tel: instead of sip: at the Request URI is handled with 416 unsupported URi message

FreePBX Providers
1

Hello to all.

I have a pjsip trunk with a provider and I receive the invite message with tel: instead of sip before the called number. I reply with 416 message (Unsupported URI).

The From, To and P-asserted Uris have the sip:.

I’m using asterisk 16.20.0 version.

Is there a way that I can bypass the Request Uri or transcode it to sip: so that I can proceed the call.

Thank you in advance for any information.

Best Regards,
Kyriakos

2

There is nothing built into Asterisk to do that, no.

3

Thank you for your reply.

I will try to use the chan_sip trunk configuration to test it.
I can see from older posts that may work.

The final step is to use a media gateway to transcode the request uri.
I’ll come back with the results.

4

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