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<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>I am facing a couple different problems with DTMF on
Asterisk (G.729 codec and dtmfmode=rfc2833) as follows:</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-indent:-.25in'><font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>1)<font size=1
face="Times New Roman"><span style='font:7.0pt "Times New Roman"'>
</span></font></span></font><font size=2 face=Arial><span style='font-size:
10.0pt;font-family:Arial'>I come into Asterisk using a Voicepulse DID over IAX.
I use DTMF for authenticating the user and collecting the destination phone
number, which works without problems. However, when I connect the second leg of
the call, via SIP to my service provider (Level-3) to terminate a PSTN call, I
face a DTMF problem. e.g if I call into my voice mail and am prompted to enter
my password, it is not recognized. Asterisk is bridging the RTP at all times
and does not use reinvite. Not sure of the problem is at the IAX end or SIP end.
Using ethereal, for every key pressed I see a total of 4 DTMF events being sent
from Asterisk to level-3. 1 with “End of Event” set to “False”
and Event Duration of 0, and 3 more each with “End of Event” set to
“True” and Event Duration set to 800. The Volume for all the
4 DTMF events is set at 10. However, this is not received at the PSTN end. I am
pretty sure this is not a Level-3 problem and have discussed it with them. Is
the total of 4 DTMF events for every key pressed correct? Are the DTMF durations
of 0 and 800 correct? If the volume setting of 10 too low? How do I fix this
problem in Asterisk?</span></font></p>
<p class=MsoNormal style='margin-left:.25in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'> </span></font></p>
<p class=MsoNormal style='margin-left:.5in;text-indent:-.25in'><font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>2)<font size=1
face="Times New Roman"><span style='font:7.0pt "Times New Roman"'>
</span></font></span></font><font size=2 face=Arial><span style='font-size:
10.0pt;font-family:Arial'>When I come over SIP from Level-3, I use Asterisk to
play voice prompts and collect DTMF digits for authentication (using Authenticate()
dialplan command) and final destination number (using “background”).
The DTMF digits are correctly interpreted by Asterisk. However, immediately
after playing the respective voice prompts, Asterisk abruptly stops sending
Level-3 any RTP packets, until all the DTMF digits are received. This period of
no RTP messages from Asterisk can last up to 10 seconds, based on how fast the
user enters the DTMF digits. Level-3 does not like this and flags it as an
error as their media gateway complains of an NRS error, which is indicative of
no RTP packets from Asterisk during an active session for a prolonged period
(several seconds). How do I tell Asterisk to continue sending RTP packets,
while waiting on the DTMF input from the user? I do not know if silence suppression
is relevant to this but is currently disabled as Level-3..</span></font></p>
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font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>Thanks..</span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'> </span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span style='font-size:10.0pt;
font-family:Arial'>scm</span></font></p>
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