Editing the way the invite is sent to the SIP trunk

I need to change the sip option we send when a phone does not answer.

Currently FreePBX shows option code 603 decline, I need this to be 486 busy here - is there any way to change the way this option is sent?


Assuming that your incoming call is routed to an extension or ring group and voicemail is not enabled, you could set Destination if no answer to Terminate Call -> Busy.

However, what you are asking for seems strange. It’s not normal to hear a busy signal after 20 or 30 seconds of ringing and callers may assume that e.g. voicemail is malfunctioning and retry several times or call your main number to report trouble. I recommend setting Ring Time to e.g. 120 seconds and let the caller decide when there is no answer. If someone hangs on for 120 seconds, perhaps route to an announcement “The extension you are calling is not answering” then hang up.

Otherwise, please describe the desired behavior.

And which side is showing that? The endpoint/phone? Or the PBX’s reply to the provider?

Just so we’re clear, when a phone doesn’t answer, the correct code is 603. The call was declined - the extension was available, but no one answered it. The 486 code should only be used if the phone is actually busy (DND, Off-hook, etc.).

Setting the “No answer destination” to “Terminate with busy” is the simplest way to accomplish what you are trying to do.

It would technically be a 408 Timeout. There are two types, time out before provisional response (100/18X) or after provisional response, A 603 means the other side did not want the call or could not accept the call and there are not other options like voicemail or CF for it to use.

The reason that I asked which side was sending the 603 is because the PBX would most likely send a 603 back to the provider when the other channel (phone) sends back a 408. There are two sides to this call and Asterisk is taking the response from the phone, processing it and present a response back to the original caller channel.

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