[Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly

[Dave Fullerton]

I have reverted back to 1.2.6 and set my sipuras to tx dtmf as info so I 
can see them with sip debug. I'll see if there is a difference and 
report on my findings in a couple days.

-Dave

Bryan Boatright wrote:
> 
> I too am experiencing DTMF problems with 1.2.7.1 that I did not 
> experience with recent prior versions.  I've backed up to version 1.2.6 
> and so far DTMF detection is working reliably (but that's only with 
> about 10 calls worth of testing).
> 
> I've only had problems over SIP channels.  Zap channels did not have 
> problems with 1.2.7.1.  I do not have any IAX channels, so cannot 
> comment on that.
> 
> I know others tend to discount DTMF problems because of "known problems" 
> with how Asterisk handles DTMF, but there does seem to be enough 
> anecdotal evidence that something bad has recently happened to make 
> things worse.
> 
> Dave, would you mind trying version 1.2.6 to see if that also resolves 
> your problems?
> 
> Dave Fullerton wrote:
>>
>> Greetings,
>>
>> I'm using asterisk to connect our three locations together with a sort 
>> of inter-company auto attendant connected like this:
>>
>> PBX (fxs) <-> Sipura 3k (fxo) <-> Asterisk <-IAX-> remote asterisk
>>
>> It works like this: Person picks up their phone and dials a number to 
>> get to the auto attendant (I don't have any FXO ports available on our 
>> PBX to do it the "right" way). The attendant answers and asks them the 
>> remote extension they want to dial. This setup has worked very well 
>> for several months. Last week I upgraded to 1.2.7.1 from 1.2.4 (I 
>> think). Since then I've been having trouble with the auto-attendant 
>> correctly detecting DTMF (missing digits). Some times it works 
>> flawlessly, others I have to try over and over before it is detected 
>> correctly. It isn't even consistently dropping the same digit from 
>> what I can see on the console. The only thing I've found is that I 
>> have a better chance of it working if I wait for the prompt to finish 
>> before dialing. I have changed the DTMF method from rfc2833 to info 
>> and finally inband with only a little change (inband seems to work the 
>> best).
>>
>> Has anyone else run into similar problems or have any more suggestions 
>> to try?
>>
>> This is the attendant portion of my extensions.conf:
>>
>> [inter-attendant]
>> exten => s,1,Answer
>> exten => s,2,Wait(1)
>> exten => s,3,Set(TIMEOUT(response)=10)
>> exten => s,4,Background(enter-ext-of-person)
>>
>> exten => i,1,Playback(invalid)
>> exten => i,2,Goto(s,4)
>> exten => i,3,Hangup
>>
>> exten => t,1,Playback(goodbye)
>> exten => t,2,Hangup
>>
>> include => tests
>> include => fullertonpbx
>> include => intercompany
>>
>>
>>
>> Thank you for any insight you can provide.
>>
>> Dave Fullerton
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