Direct RTP, DTMF and voicemail issues

Asterisk Asterisk Support
1

I’m new to asterisk so be kind :smile:

I’m running into an issue with Asterisk in that when I make phone calls the RTP streams go through the Asterisk box. The whole concept with VoIP is that this doesn’t need to happen. What I’ve found is that this is because of the RFC2833 DTMF setting causing it. If you change the DTMF to ‘info’ the RTP streams go directly between the handsets. That works fine and dandy until you try to call voicemail. Voicemail does not detect DTMF when you have info set.

In the URL below it says that if RFC2833 is enabled that RTP will go through the Asterisk box. It says “When dtmfmode=rfc2833, asterisk will send the RTP stream through asterisk. With dtmfmode=info canreinvite works properly.”

voip-info.org/wiki/view/Aste … anreinvite

So I ask, why would anyone want relayed RTP in a pure VoIP environment that allows direct RTP streams? Yes I know the theory is different when NAT, gateways and other things come into the mix which might have different requirements/capabilities… I’ve worked on AVAYA, Cisco, NEC and other VoIP PBX’s and have not had this behavior.

Equipment:
Asterisk 1.8
Cisco 7960
X-Lite 4 software

I’ve not have much luck finding any resolution from Googling forums… Anyone else seen this?

2

I found some dial command options that caused it…

tr was coded…

t allows for the calling parting to be x-fer’ed by using a key that defined in extensions.conf, which is coded as # in mine.
r allows for ringback tone generation to the calling party

I removed both of these and still have ringback tone and can x-fer parties using the xfer button… I assume this is probably for older devices, essentially a compatibility mode??

3

“canreinvite” has been renamed “directmedia” and the old name may soon be removed, although it defaults to yes, so the wrong name wouldn’t disable it.

Quite a lot of people seem to use DTMF (features.conf) transfers, even though they are using SIP phones. I don’t know why.