[Asterisk-Users] SJphone, Asterisk and DTMF tones ...

[John Todd]

So, it seems a new bug has been found, which may or may not be at the 
root of this problem.

Let me describe it, and see if you agree with the synopsis:

   Asterisk, despite having dtmfmode= set to a particular value in 
sip.conf for a peer, will listen for SIP Info method transmissions 
even if RFC2833 is selected.  In some phones (Grandstream, in 
particular) this causes double-transmission of digits, since the 
phone sends both types of DTMF transmissions without blocking the 
other.  Asterisk should ignore the other two types of DTMF 
transmission when selected to do one type of reception to counter 
these types of equiment peculiarities which seem to prevent correct 
DTMF usage.


If I have described this correctly (I don't know - I don't have 
visibility into this problem) then can someone else (preferably 
someone with the problem) open a ticket?

JT


>I had the same problem with Grandsteam phones and *.  No other hard 
>or soft phones have the 'double digit' problem with *.  I don't 
>think Asterisk can do both RFC2833 and in-band DTMF at the same 
>time.  It does, however, do RFC2833 and SIP Info at the same time 
>(SIP Info method seems to be on all the time, even when RFC2833 is 
>selected in the sip.conf file).  Switching the Grandsteam to SIP 
>Info allowed it to talk to Asterisk and fixed the double digits 
>problem.
>
>- Jim
>
>Chris Albertson wrote:
>
>>I think this is a problem on the Asterisk side.  I'm seeing
>>the same problem using a Grandstream Budgetone 100.  And the GS
>>does have setting for both in-band and RFC2833.
>>
>>My guess is asterisk is accepting the DTMF tone __both__ ways
>>It is reading the RFC28833 stuff _and_ "hearing" the audio tones
>>as well. 
>>
>>--- Tilghman Lesher 
>><mailto:tilghman at mail.jeffandtilghman.com><tilghman at mail.jeffandtilghman.com> 
>>wrote:
>>
>>>On Sunday 21 December 2003 00:29, Darren Nickerson wrote:
>>>    
>>>>Folks,
>>>>
>>>>I can't seem to get DTMF signaling working properly using SJphone
>>>>connecting to Asterisk via a SIP connection. Here's an example of a
>>>>voicemail session where I entered 1234 for both the username and
>>>>the password:
>>>>
>>>>     -- Incorrect password '11223344' for user '11223f344' (context
>>>>      
>>>>
>><snip>
>>
>>>Changing the DTMF mode would indeed seem to be the logical
>>>solution.  However, it appears that SJphone does not support that
>>>option (after a quick perusal of their PDF).  You might want to file
>>>a bugtracker request on their website to implement that functionality.
>>>
>>
>>=====
>>Chris Albertson
>>   Home:   310-376-1029  
>><mailto:chrisalbertson90278 at yahoo.com>chrisalbertson90278 at yahoo.com
>>   Cell:   310-990-7550
>>   Office: 310-336-5189  
>><mailto:Christopher.J.Albertson at aero.org>Christopher.J.Albertson at aero.org
>   KG6OMK
>
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