[EMAIL PROTECTED] asterisk]# grep Duration= channels/*c channels/chan_sip.c: snprintf(tmp, sizeof(tmp), "Signal=%c\r\nDuration=250\r\n", digit); [EMAIL PROTECTED] asterisk]#
> -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-dev- > [EMAIL PROTECTED] On Behalf Of James Sizemore > Sent: Monday, 17 October 2005 2:27 PM > To: Asterisk Developers Mailing List > Subject: [Asterisk-Dev] INFO and Duration=250 > > I have a gateway using a Digium card to convert a PRI > call to a sip call then I transport the sip call to a Cisco > IAD where it is converted back to a PRI. This all works > well except DTMF is sent with a duration of .25sec. > PRI specs says this should be .25sec to .5sec so this > is with in spec, however the PBX on the other side of > the IAD does not reliable work with the DTMF tones > the minimum allowable length. I found in the INFO packets > where the DTMF is set to a duration or 250, I would like > to change this to 500. > > Which file and class would be the correct place to change > this value at? > > ============== > INFO packet options I would like to change > ============== > Content-Type: application/dtmf-relay > Content-Length: 24 > > Signal=5 > Duration=250 > ============== > _______________________________________________ > Asterisk-Dev mailing list > [email protected] > http://lists.digium.com/mailman/listinfo/asterisk-dev > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev _______________________________________________ Asterisk-Dev mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-dev To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
