[asterisk-users] DTMF not reproduced towards ZAP T1 Port after connection when arrives as SIP

[Barton Fisher]

OK, tried changing DTMF tone as described on URL and no difference

Bart

Steve, I fooled with dtmf mode and it was 2833 - However, got stranger
results with inband, seems it would take digits, but audio goes to 1 way
afterwards first push. 

As far as changing the code per the URL, I think I get what's it doing, but
wonder what else would be effected afterwards - I guess I could switch back
if it turns out to be a bad idea

Bart


On Sat, Jun 21, 2008 at 12:11 PM, Barton Fisher <bart at icpage.com> wrote:
> I place SIP DID call towards ZAP (TE410P). ZAP uses e&m signaling to an
> external IVR system. I can hear the asterisk sending the DTMFs properly
> toward ZAP at call setup. After the call connects, any further DTMF digits
> from SIP is very short sounding or distorted (barely audible)  on the ZAP
> and ignored. ZAP to ZAP connections work perfect.
>
> Just so you know, with 1.2 this is not an issue and this issue is keeping
me
> from moving to 1.4.
>
> I have a test system setup with a SIP DID to a test IVR system to
> demonstrate this problem. I can provide full access to these systems for
> testing. I've placed on Digium bugs but have not received any responses
yet.
> There is nothing in the logs or cli that provides anything meaningful -
> Below is a call where I press '*' and it is ignored.

Hello, here are a few pointers that might help.  Are you using
RFC2833, inband, info?  My guess is 2833, you might want to give
inband a try unless you are using a lossy codec.

This is pretty interesting and might solve your issue.  It seems that
by doing this, Asterisk just passes the DTMF as regular audio instead
of trying to interpret it.  Bookmarked for when I run into this same
issue.....
http://astrecipes.net/index.php?n=248

Linked from this page on the wiki that has more info on your issue.
http://www.voip-info.org/wiki/view/Asterisk+DTMF

Thanks,
Steve Totaro





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