I checked and it was version 1.4.21.2 which I got working correctly (all aastra hardware, mostly 57i) after applying a patch for rtp.c. Please see the link. The release notes for 1.4 said something about this being fixed in 1.4.38, but I do know that the above worked well for me if you want to try that route first.
http://pbxinaflash.com/forum/showpost.php?p=17794&postcount=8 On Tue, May 3, 2011 at 15:30, Simon P. Ditner <si...@uc.org> 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, <dyzsil...@gmail.com> 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" <spdit...@gmail.com> > > Date: Thu, 28 Apr 2011 22:47:19 > > To: dyzsil...@gmail.com<dyzsil...@gmail.com> > > Cc: mgam...@mgamble.ca<mgam...@mgamble.ca>; Simon P. Ditner<si...@uc.org>; > Asterisk Users Group<asterisk@uc.org> > > 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, dyzsil...@gmail.com wrote: > > > >> While we're at it, which device are you using? I've found some handle > better than others. > >> ------Original Message------ > >> From: mgam...@mgamble.ca > >> To: Daniel Silver > >> To: Simon P. Ditner > >> To: spdit...@gmail.com > >> To: 'Asterisk Users Group' > >> ReplyTo: mgam...@mgamble.ca > >> 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: dyzsil...@gmail.com > >> To: Simon P. Ditner > >> To: spdit...@gmail.com > >> To: 'Asterisk Users Group' > >> ReplyTo: dyzsil...@gmail.com > >> 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: spdit...@gmail.com > >> 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: asterisk-unsubscr...@uc.org > >> For additional commands, e-mail: asterisk-h...@uc.org > >> > >> > >> > >> > >> --------------------------------------------------------------------- > >> To unsubscribe, e-mail: asterisk-unsubscr...@uc.org > >> For additional commands, e-mail: asterisk-h...@uc.org > >> > >> > >> Sent from my BlackBerry device on the Rogers Wireless Network > > > > >
