[Asterisk-Users] Sipura + Asterisk 1.2 + dtmf

[Asterisk User]

Hi,

Thanks very much for this excellent explanation.

 

I had played with it and the option that best worked for me was to set
dtmf=INFO in the Sipura and dtmfmode=auto or dtmfmode=rfc2833 in sip.conf.

I just could not fine the logic as to why I had to make it INFO.

 

I hope Sipura/Lynksys/Cisco team come out with that solution soon.

 

Thanks again,

 

Oswald

 

  _____  

From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of tracinet
Sent: Monday, December 12, 2005 9:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Sipura + Asterisk 1.2 + dtmf

 

Oswald,
I have had the same issues with Sipura devices since moving to Asterisk 1.2
as well.  We use rfc2833 exclusively in our network and the Sipura devices
just stopped working with regard to DTMF.  After MANY packet captures
comparing Sipura devices which did not work to Cisco devices that did work,
it was found that the Sipura implementation of rfc2833 was not to spec
(Sipura calls it AVT).  Specifically, when the Sipura device sends the RTP
packets for each DTMF digit, the mark bit is set to "1" for each packet
instead of just the first packet's mark bit being set to "1".  Previous
versions of asterisk were not as strict with regard to rfc2833 which is
probably why you did not have issues before.

I have confirmed this issue with our Sipura vendor as well as some of the
developers in the asterisk-dev IRC channel.  Our vendor has taken all the
packet captures and notified Linksys of this bug in hopes that updated
firmware will be released soon as it appears to affect Sipura/Linksys phones
and ATAs.

In the meantime, there is a "workaround" that you can use to get *somewhat*
accurate DTMF tones.  If you set the device DTMF settings to INFO and you
specify rfc2833 as the dtmfmode in sip.conf, the phone *should* pass DTMF
digits as long as you are not using a speakerphone.  If you are using a
speakerphone, either pickup the handset when pressing tones or hit the mute
button while pressing tones to avoid the tones getting duplicated by
microphone pickup.

Hopefully Linksys/Cisco/Sipura gets this fix out soon since the whole point
of using rfc2833 for DTMF is to avoid getting duplicate and inaccurate tones
sent as a result of microphone pickup and to pass the digits in their own
RTP stream.

Anyone on this list who is using Sipura devices and is having this
AVT/rfc2833 DTMF issue, please contact your Sipura vendor and make them
aware of this issue and ask them to notify Linksys (as Linksys will only
deal with their resellers).  Hopefully if enough "noise" is made, they will
sense the urgency in getting this fixed and can put out a bug fix in the
form of updated firmware.

On 12/12/05, Asterisk User <asterisk at wondervoip.com> wrote: 

Hi,
I have a problem with Sipura and Asterisk 1.2... everything was working
smoothly with 1.0.9 until I upgraded to 1.2.
The DTMF tones are no longer working, I cannot access Voicemail or send DTMF
digits anywhere.

What changed in version 1.2??

I've read many people with the same issue but with different phones, has 
anyone figure out what's wrong??

Oswald


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