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.<br><br><div><span class="gmail_quote">On 12/12/05, <b class="gmail_sendername">Asterisk User</b> <<a href="mailto:asterisk@wondervoip.com">asterisk@wondervoip.com</a>> wrote:
</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">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>_______________________________________________<br>--Bandwidth and Colocation provided by <a href="http://Easynews.com">Easynews.com</a> --<br><br>Asterisk-Users mailing list
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