[Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly

[Bryan Boatright]

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
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users