[Asterisk-Dev] DTMF bugs in Asterisk ?

[scm-j at nuntius.com]

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.

 

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.

 

 

I am facing a couple different problems with DTMF on Asterisk as
follows:

 

1)       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?

 

2)       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..

 

Thanks..

 

scm

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