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

[Steve Totaro]

Bart,

Did you try the method of inband along with changing the frequencies
at the same time?

Thanks,
Steve T

On Sat, Jun 21, 2008 at 3:29 PM, Barton Fisher <bart at icpage.com> wrote:
> 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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