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

[Barton Fisher]

Yep - tried both and combination thereof - The bad effect of inband mode was
audio went one way after first press
My test app reads back the ANI & DNIS at answer (which works), then prompts
for more digits.  
It's suppose to read back whatever is heard. I can see it reading back
something, back I don't hear anything.

One note: if I press say '1111111' fast, it might hear '11', but not all
digits sadly
I'm sure this is a 'bug' as it work perfectly on 1.2, but so far there is no
acknowledgement from Developers yet.
Not sure how long it should take :(

Bart


-----Original Message-----
From: Steve Totaro [mailto:stotaro at totarotechnologies.com] 
Sent: Sunday, June 22, 2008 7:36 AM
To: bart at icpage.com; Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] DTMF not reproduced towards ZAP T1 Port after
connection when arrives as SIP

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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