Iam newbie in asterisk
I wanna asking something after fresh install freepbx 13 asterisk 13 in debian 8.1
whats the meaning about this error on debugging message
i have re-download codec g729 from asteriskhostinglv for my server
and now my trunk is establish but still can’t call via outbound routes

That codec implementation is not supported here.

Moreover, it doesn’t have valid licence documents. I suspect that, if the missing licence document from Intel was tracked down, it would have a no commercial use restriction, meaning the code is not open source. The original “free” g.729 implementation, which I suspect this may be, had a void GPL licence because of this. Given that history I’m not going to attempt to support this codec implementation, even from first principles.

(Also note that we don’t support FreePBX here, although nothing in your report appears FreePBX specific.)

Having said that, your screen shot shows an incomplete trace of a, so far, successful INVITE transaction.

Also note that screen shot are generally not useful. Much more than one screenful is needed, to the extent that one really needs to be able to do text searches.

Paste the log dont, use images please, also there is an authentication issue 401 response to the INVITE is note related to codec

Issue implies a problem, but 401 is completely normal.

I know 401 it is normal but it will depend on what scenario you get a 401 for a example on response to an unauthenticated REGISTER request, but no body is expecting a 401 response on an INVITE request

You should be expecting 401 on INVITEs from local devices. Most ITSPs won’t reverse authenticate, so you have to use inescure=invite, or remotesecret (for chan_sip) for that to be suppressed, and one should not be going insecure when it isn’t necessary.

Dont know the scenario here but mybe the from header is matching with a local device, and this is why the 401

That’s quite common, and the result of wrongly using type=friend.

However, as this is WebRTC, it is also likely that the OP just failed to configure any credentials for the realm that Asterisk is using.

I didnt know he was using Webrtc as I dont see any referece about in on the thread

My mistake. I thought he’d said someting about it, but it looks like I was wrong.

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