---------- Forwarded message ---------- From: Henry Coleman <[email protected]> Date: Thu, May 12, 2011 at 12:48 PM Subject: Re: [on-asterisk] DTMF, will it ever work? To: [email protected]
In general settings (freepbx) the default is no t/T but their is no k/K mentioned. IMHO I would assume that it is not used. H On Thu, May 12, 2011 at 12:01 PM, Roger <[email protected]> wrote: > Thanks, simon. So you use purely asterisk. I was wondering how to remove > them in gui, for example, freepbx or elastix. > > Yajie > Sent from my BlackBerry device > > -----Original Message----- > From: "Simon P. Ditner" <[email protected]> > Sender: [email protected] > Date: Thu, 12 May 2011 11:12:45 > To: <[email protected]> > Subject: Re: [on-asterisk] DTMF, will it ever work? > 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. > >>>> > >>>> --------------------------------------------------------------------- > >>>> To unsubscribe, e-mail: [email protected] > >>>> For additional commands, e-mail: [email protected] > >>>> > >>>> > >>>> > >>>> > >>>> --------------------------------------------------------------------- > >>>> To unsubscribe, e-mail: [email protected] > >>>> For additional commands, e-mail: [email protected] > >>>> > >>>> > >>>> Sent from my BlackBerry device on the Rogers Wireless Network > >>> > >>> > >> > >>--------------------------------------------------------------------- > >>To unsubscribe, e-mail: [email protected] > >>For additional commands, e-mail: [email protected] > >> > > > > --------------------------------------------------------------------- > To unsubscribe, e-mail: [email protected] > For additional commands, e-mail: [email protected] > > -- * Henry Coleman* [image: Picture] -- * Henry Coleman* [image: Picture]
