<span style="font-family: Arial, Helvetica, sans-serif; font-size: 10pt">Hi all<br />
<br />
I have posted a question on the asterisk dev board about this issue but I want to see if any users have run up against this.<br />
<br />
This issue is that when calls are run through Broadvox and Level 3 the in-call rfc2833 dtmf is not reliable. This occured for me on asterisk version 1.6.1.18, 1.6.1.20 it appears to have been fixed when I went to 1.6.2.11 but broken again in 1.6.2.12-rc1.<br />
I have tested with Grandstream and SNOM phones and both fail 90% of the time Unidata phones fail 10% of the time Audiocodes and Grandstream ATA's appear to not suffer from the issue on any version of asterisk. <br />
<br />
What happens is when a caller trys to enter DTMF keys durring a call the far end routed through these carriers do not detect all of the digits. We did captures with broadvox and here is what they have said. <blockquote style="margin-right: 0px;" dir="ltr">
<div><br />
Hello,<br />
<br />
</div>
<p>Per our phone conversation I have attached our signaling capture. The issue is that after we receive a RTP packet, the RTP event that follows needs to be sent within 100 ms. Anything greater than 100 ms will not be received. <br />
Thank you,<br />
<br />
<span style="color: #1f497d;">Broadvox<br />
</span><span style="font-family: arial, sans-serif; color: #1f497d; font-size: 10pt;">Network Operations Center<br />
</span></p>
</blockquote><br />
Any one else seen this? Any ideas?<br />
<br />
Please note you must be being proxied directly to the carrier so your RTP flows direct other wise you will not see the issue.<br />
<br />
Thanks<br />
Bryant<br /></span>