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<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Hi,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Thanks very much for this excellent
explanation.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>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.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I just could not fine the logic as to why
I had to make it INFO.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>I hope Sipura/Lynksys/Cisco team come out
with that solution soon.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Thanks again,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Oswald<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'><o:p> </o:p></span></font></p>
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<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b>tracinet<br>
<b><span style='font-weight:bold'>Sent:</span></b> Monday, December 12, 2005
9:32 PM<br>
<b><span style='font-weight:bold'>To:</span></b> Asterisk Users Mailing List -
Non-Commercial Discussion<br>
<b><span style='font-weight:bold'>Subject:</span></b> Re: [Asterisk-Users]
Sipura + Asterisk 1.2 + dtmf</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal style='margin-bottom:12.0pt'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>Oswald,<br>
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.<br>
<br>
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.<br>
<br>
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.<br>
<br>
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.<br>
<br>
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.<o:p></o:p></span></font></p>
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<p class=MsoNormal><span class=gmailquote><font size=3 face="Times New Roman"><span
style='font-size:12.0pt'>On 12/12/05, <b><span style='font-weight:bold'>Asterisk
User</span></b> <<a href="mailto:asterisk@wondervoip.com">asterisk@wondervoip.com</a>>
wrote: </span></font></span><o:p></o:p></p>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'>Hi,<br>
I have a problem with Sipura and Asterisk 1.2... everything was working<br>
smoothly with 1.0.9 until I upgraded to 1.2.<br>
The DTMF tones are no longer working, I cannot access Voicemail or send DTMF<br>
digits anywhere.<br>
<br>
What changed in version 1.2??<br>
<br>
I've read many people with the same issue but with different phones, has <br>
anyone figure out what's wrong??<br>
<br>
Oswald<br>
<br>
<br>
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