[Asterisk-Users] Re: Livevoip still no DTMF?

[Joel Jn-Francois]

> > I read in the archives a number of discussions about livevoip, DID,
> > and DTMF not working.
> >
> > However, no resolutions.
> >
> > I just setup a livevoip DID and indeed the DTMF does not work.
> >
> > The same asterisk context works via broadvoice and via
> > direct dialing in to the asterisk server via SIP.
> >
> > Just no DTMF with calls via livevoip.
> >
> > I'm running Asterisk CVS-v1-0-03/06/05-23:15:12
>
>Its been working fine here for about a month now. Currently using
>CVS-HEAD-03/31/05, however it worked fine with several previous
>cvs-head versions as well.

>Below are the pieces I'm using for incoming calls. Might want to
>review and compare to whatever you're using. The iax.conf section
>is a very basic type=user with a context referring incoming calls
>to the liveviop800 section of extensions.conf shown below.
>
>[livevoip800]
>include=>bus-ivr-main
>exten=>8001234567,1,Dial(${PHONE6}&${PHONE7},10)
>exten=>8001234567,2,Goto(bus-ivr-main|s|1)
>
>[bus-ivr-main]
>exten => s,1,Wait,1
>exten => s,2,Answer
>exten => s,3,DigitTimeout,5
>exten => s,4,ResponseTimeout,20
>exten => s,5,Background(npi-greeting)  ; "Thanks for calling press 1 for"

This DTMF and livevoip issue seems quite interesting and really mystifies 
me. The fact that some livevoip customers have this issue and others don't, 
makes this all the more confusing.  I love livevoip service and support, I 
think they are great, but this one issue is creating a nightmare for me 
during.

I am currently testing a calling card asterisk application I developed.  I 
have about 30 people presently testing the system for me and the DTMF issue 
has been everyone's main complaint.  It's either the pin number which they 
know they entered correctly is wrong, or the destination number is 
invalid.  Sometimes we get someone on the other line that we did not 
call.   I am really scared of the ramification if I proceed with the 
launching of this service before this issue is resolved.

If Livevoip cannot seem to resolve this problem on their own, is there any 
way that we can put our heads together and get to the bottom of this 
problem?  I am not only referring to those who are experiencing this 
problem, but also to those who have a similar setup and have not 
encountered this problem.  We can compare our settings and any thing else 
we have noticed during testing.  I am going to get the ball rolling.

Here is a summary of what I am doing:

I developed an asterisk calling card application written in C.  This 
application simply prompts for a PIN number, checks to see if the PIN 
number is valid and whether there are sufficient funds to place a call.  If 
there are sufficient funds the application would then prompt for the 
destination number. At least 30%-50% of the time (can be more sometimes), 
one or both of these prompts would be wrong.  Yesterday was a frustrating 
day for me.  After switching over my DIDs from IAX to SIP with the hope 
that this may resolve the problem, I was still consistently able to 
duplicate the problem.  I would immediately switch to a different DID 
provider and get through 100% of the time without any DTMF issues.

Here is a list of things I have setup, what I am doing and what I have 
noticed trying to troubleshoot this problem.

1. I am using a stable version of asterisk CVS-v1-0-03/26/05-16:54:47.

2. I have tested this problem on CVS-HEAD-03/10/05 and CVS-HEAD-03/26/05 
and I am able to duplicate the problem every time.

3. I am using 1800 DID numbers from LiveVOIP.

4. I am using local DID numbers from SixTel

5. I do NOT use any DTMF settings in IAX.conf nor SIP.conf

6. I am using ast_app_getdata to play the requested prompt and store the 
number entered into a variable. I would then display the content of that 
variable in asterisk command line so that I know exactly what values the 
system is receiving and compare it to what was entered.
      Eg. For the PIN Number: reslt = ast_app_getdata(channel, pinprompt, 
pinnum, 10,0);
            For the Destination number: reslt = ast_app_getdata(channel, 
destprompt, destinationno, 16,0);

7. I think (not 100% sure) that the default DigitTimeout is 6 seconds and 
the default ResponseTime is 12 seconds for the ast_app_getdata.


8. When getting data for the PIN number and the destination number I have 
noticed the DTMF issue is more consistent with the destination number 
rather than the PIN number.  My assumption is that the PIN number is always 
10 digits, so there is no pause after the last digit has been 
entered.  Once the system gets 10 digits, it proceeds to the next step 
ignoring any other digits that may have been received by the system.

There is no fixed length for phone numbers; therefore, depending on where 
you are calling, the length of the phone number tends to vary.  After 
entering a 10 digit number the system may receive added data making the 
number invalid.

9.  When I take a look at the content of the variables compared with the 
digits entered, the problem is either one of two things:
A. Double digits in the number, eg. 4076831234 might be 40076831234 or 
40776831234 etc.
B.  Some of the numbers spill over into the next prompt for data.

10. You can call using a local number from Sixtel or a 1800 number from 
LiveVOIP.  We presently do NOT experience any problems with our local DID 
numbers from Sixtel.

11. The problem happens on both IAX and SIP based DID service.

Thanks everyone, that's all I have for now. I really hope we can resolve this.