Direct rtp setup and call tranfer option

hello,

by removing the “t” from Asterisk dial command option in General settings, and changing some settings, i was able to make the media traffic between the calls go directly between the clients and not pass through the asterisk server.

But by removing the “t” options i lost the call transfer option, so i a looking for a way to keep the transfer option and keep the direct media calls.

Any thoughts?

P.S:I really dunno why they are related, why keeping the call transfer option forces the calls to go through the asterisk servers, and removing the call transfer option, allow direct media calls.

I guess you have to pay a few bucks for the [email protected]

Yes, you can use the SIP transfer function of the phone without the t. The phones sends out the reinvite.

Even if it was a softphone ? like i am using x-lite?

I would think x-lite works fine. Does the free version have a transfer button?

No it seems in all the free versions of the softphone the transfer function is disabled!

Because Asterisk has to “listen” to the media stream for the in call transfer sequence.

You should still be able to transfer from and endpoint that has a local SIP transfer key.

What do you mean by “You should still be able to transfer from and endpoint that has a local SIP transfer key.”

Do you mean after removing the “t” option?