hi simon,
  sorry, can tell me how did remove the t and k option? and how did you confirm 
you are now on packet2packet mode? on Wiki page, those are defined
T -Allow the calling party to transfer the called party by sending the DTMF 
sequence defined in features.conf. This setting does not perform policy 
enforcement on transfers initiated by other methods.
K -Allow the calling party to enable parking of the call by sending the DTMF 
sequence defined for call parking infeatures.conf.

"Simon P. Ditner" <[email protected]> wrote:

>I've done this test on 3 different platforms -- AMD Geode, Intel Atom
>270, and VirtualBox on x86 and AMD64. The virtualization had no impact
>on the outcome.
>
>When I remove the 'T' and 'K' options to Dial(), and the Packet2Packet
>bridging kicks in, I can dial as fast as my fingers will move.
>Otherwise, I'm limited to about 2 digits / second. My captures from
>wireshark using asterisk 1.8 show every other digit being missed when
>dialing at a normal rate with T or K present with a single active
>call, so Asterisk's in-line dtmf monitoring seems to muck things up
>pretty badly.
>
>On 3 May 2011 13:07,  <[email protected]> wrote:
>> I had this with 1.4 a while back, version 1.4.21 or so fixed this if I 
>> recall the version correctly, but I'll double check. Also, there was a patch 
>> for the RTP.c (I think), but again I'll double check. We were having the 
>> DTMF issue with calling voicemail and ivr's at various other offices.
>>
>> By any chance is your system virtualised?
>> -----Original Message-----
>> From: "Simon P. Ditner" <[email protected]>
>> Date: Thu, 28 Apr 2011 22:47:19
>> To: [email protected]<[email protected]>
>> Cc: [email protected]<[email protected]>; Simon P. Ditner<[email protected]>; 
>> Asterisk Users Group<[email protected]>
>> Subject: Re: [on-asterisk] DTMF, will it ever work?
>>
>> I see this across a wide range of conditions, asterisk 1.4 thru 1.8, sip 
>> with rfc2833, using snom, polycom, and aastra handsets.
>>
>> When I sniff the traffic, I see the rfc2833 rtp go into asterisk in the 
>> right order, but it comes out of asterisk in the wrong order.
>>
>> ---
>> Simon P. Ditner
>>
>> On 2011-04-28, at 10:29 PM, [email protected] wrote:
>>
>>> While we're at it, which device are you using? I've found some handle 
>>> better than others.
>>> ------Original Message------
>>> From: [email protected]
>>> To: Daniel Silver
>>> To: Simon P. Ditner
>>> To: [email protected]
>>> To: 'Asterisk Users Group'
>>> ReplyTo: [email protected]
>>> Subject: Re: [on-asterisk] DTMF, will it ever work?
>>> Sent: 28 Apr 2011 22:28
>>>
>>> And what DTMF mode are you trying to use on both the SIP Trunk and the 
>>> device?  With RFC2833 on both side and Packet to Packet bridging you 
>>> shouldn't have any issues.
>>>
>>>
>>> ------Original Message------
>>> From: [email protected]
>>> To: Simon P. Ditner
>>> To: [email protected]
>>> To: 'Asterisk Users Group'
>>> ReplyTo: [email protected]
>>> Subject: Re: [on-asterisk] DTMF, will it ever work?
>>> Sent: Apr 28, 2011 22:27
>>>
>>> Which version of asterisk are you using?
>>> ------Original Message------
>>> From: Simon P. Ditner
>>> Sender: [email protected]
>>> To: 'Asterisk Users Group'
>>> Subject: [on-asterisk] DTMF, will it ever work?
>>> Sent: 28 Apr 2011 22:22
>>>
>>> Is there some trick to getting asterisk working reliably when calling
>>> into conference bridges and voicemail? No end of trouble with DTMF on
>>> my home system regardless of which provider I'm using. Drives my wife
>>> nuts when she's working from home.
>>>
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>>>
>>>
>>>
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>>>
>>> Sent from my BlackBerry device on the Rogers Wireless Network
>>
>>
>
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