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

[Barton Fisher]

Yeah, it gets a bit confusing with all the scenario possible - Regardless,
you are right I should stay on 1.2 until 1.4 is ready for prime time, but
now that 1.6 is out, I'm sure I'm in for a long wait. I reposted my bug
again, since I think I may have listed it wrong - it's now
http://bugs.digium.com/view.php?id=12913 - Maybe now someone might notice :)

Thanks, Steve for your inputs

Bart


Asterisk has never been good at catching DTMF in rapid succession.  I
have read in many places that asterisk 1.4 had many changes to DTMF.

You contradict yourself below.  "The bad effect of inband mode was
> audio went one way after first press" and "One note: if I press say
'1111111' fast, it might hear '11', but not all digits sadly"

I suppose that you were using different methods.  Try pressing the
keys a little slower.

Personally, I would just go back to 1.2.X if you cannot get anyone to
acknowledge your issue.  What features do you need in 1.4 anyways?
Maybe if the DTMF bugs you opened get resolved then 1.4.X could be
revisited.

Thanks,
Steve T

On Sun, Jun 22, 2008 at 11:30 AM, Barton Fisher <bart at icpage.com> wrote:
> 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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