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]
