I was explicitly using the tk/TK options in the Dial() command, so I
removed them and live without the features

On 4 May 2011 23:36, Roger <[email protected]> wrote:
> 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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