[Asterisk-Users] Asterisk 1.2.7.1 DTMF anomaly (UPDATE)

Dave Fullerton dfullertasterisk at shorelinecontainer.com
Tue Apr 25 06:19:04 MST 2006


I think my DTMF problems are solved, but the solution isn't crystal 
clear. I reverted back to 1.2.6 and then had the idea to have asterisk 
email me every time someone hit the invalid extension. The email 
contained the number they dialed and the channel (read sipura box) they 
came in on. After a day and a half I noticed 99.9% of my invalid 
extensions were all coming from one sipura box that happened to be 
running version 3.1.10 firmware. I downgraded to 3.1.3 and the DTMF 
problems disappeared. Thing is, I *know* I was getting invalid 
extensions on calls from the other sipura box (which is running 3.1.7) 
when I was running asterisk 1.2.7.1 but I don't know how many and I'm 
not going back to find out just yet. (Need to wait for my users to stop 
thinking its a useless POS first).

So there *may* still be an issue with 1.2.7.1, but I definitely had an 
issue with sipura firmware 3.1.10 and DTMF detection.

Thanks to everyone who submitted ideas.

-Dave




Dave Fullerton wrote:
> 
> 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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