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

[Steve Totaro]

Also, when you tried inband, did you set it on the phone as well as sip.conf?

Thanks,
Steve T

On Sun, Jun 22, 2008 at 10:35 AM, Steve Totaro
<stotaro at totarotechnologies.com> wrote:
> 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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