[Asterisk-Users] VoicePulse DTMF Problems Anyone?
Rich Adamson
radamson at routers.com
Sat Jun 11 00:09:27 MST 2005
> We are developing an IVR application and when I am testing locally on
> my machine using a softphone (iaxcomm) the digits I press for GET DATA
> work every time. I am testing with a local extension that goes right
> into my routine. However when I try to call in to the system using an
> analog or cell phone GET DATA drops some digits that are pressed.
> There doesn't seem to be a pattern to which digits get dropped either.
> Digits in the beginning middle or end gets dropped equally.
>
> I am wondering if anyone else is experiencing similar issues. I
> believe the problem lies with VoicePulse because we are using them for
> IAX connections. I don't believe its a bandwidth problem on my
> network (cable) because I have tried the same exact system/config
> everything on another network (T1) and the same digit dropping
> continues to happen. This is happening with a load of 1 call.
>
> Is this problem with VoicePulse? Is anyone else experiencing it? Can
> anyone recommend a more reliable company?
In most previous cases, dtmf issues have been related to how you
define your interfaces. For sip definitions, use dtmfmode=rfc2833.
Some itsp's have an issue with asterisk in that a completed "iax" call
to an asterisk IVR is considered an "answered" call, and therefore
expect dtmf tones to be passed to the endpoints. In this case, the
dtmf tones are expected to be generated by the phone and passed
to the IVR as inband audio tones. I'm not a voicepulse user, so don't
know if they have some particular problem or not.
If the dtmf digits are expected to be passed as inband audio tones,
then a reasonable codec would need to be specified. Might try ulaw
if you are using something different now.
My system has iax trunks from multiple itsp providers, multiple
iax links to other companies that we work with, a variety of sip
phones (each defined with rfc2833), and multiple analog pstn lines.
We don't have a problem (cvs-head) with an IVR that starts out as:
[bus-ivr-main]
exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,Set(TIMEOUT(digit)=5)
exten => s,4,Set(TIMEOUT(response)=15)
exten => s,5,Background(abc-greeting) ; "Thanks for calling press 1 for"
exten => s,6,Hangup
More information about the asterisk-users
mailing list