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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'>I did post this on asterisk-users earlier.
Is the problem mentioned in Item 2 a bug in Asterisk? Any
assistance is appreciated as I need resolution to them to complete my SIP interop
testing with Level-3 successfully.</span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'> </span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'>Just a clarification that the item 1 problem
is specific to G.729 and rfc2833 combination. It works fine when I use G.711-ulaw
codec and inband DTMF. However, item 2 problem exists for both codes and DTMF
types.</span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'> </span></font></p>
<p class=MsoNormal><font size=2 color=navy face=Arial><span style='font-size:
10.0pt;font-family:Arial;color:navy'> </span></font></p>
<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 as follows:</span></font></p>
<p class=MsoNormal style='margin-left:.5in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'> </span></font></p>
<p class=MsoNormal style='margin-left:.25in;text-indent:-.25in'><font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>1)</span></font><font
size=1><span style='font-size:7.0pt'> </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:.75in'><font size=2 face=Arial><span
style='font-size:10.0pt;font-family:Arial'> </span></font></p>
<p class=MsoNormal style='margin-left:.25in;text-indent:-.25in'><font size=2
face=Arial><span style='font-size:10.0pt;font-family:Arial'>2)</span></font><font
size=1><span style='font-size:7.0pt'> </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>
<p class=MsoNormal style='margin-left:.5in'><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'>Thanks..</span></font></p>
<p class=MsoNormal style='margin-left:.5in'><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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