Basically outgoing calls through zap channels doesn't detect that the other end answered. In my cdr I see hang-up no answer, plus the console shows that the channel is ringing..while I am actually talking to someone. Incoming calls seems to be fine. Wojtek ----- Original Message ----- From: "Wojciech Tryc" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Thursday, June 24, 2004 7:27 PM Subject: Re: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
> I have similar problem with outbound calls... > Wojtek > ----- Original Message ----- > From: "Brent Franks" <[EMAIL PROTECTED]> > To: <[EMAIL PROTECTED]> > Sent: Thursday, June 24, 2004 7:16 PM > Subject: RE: [Asterisk-Users] tdm (and x100p?) echo - fix is coming! > > > > > > > -----Original Message----- > > > From: [EMAIL PROTECTED] [mailto:asterisk-users- > > > [EMAIL PROTECTED] On Behalf Of Rich Adamson > > > Sent: Thursday, June 24, 2004 5:01 PM > > > Be careful with that thought... here's the three lines that were > > > manually changed for testing purposes only (these would have been > > prior to > > > yesterday's change to chan_zap.c): > > > ~1195: x = 800; > > > ~1636: strcpy(p->echorest, "ww"); > > > ~1637: strcpy(p->echorest + 2, > > > > > > Changing x = 400 to x = 800 fixed the echo problem, but caused > > outbound > > > dialing to totally fail. The pstn line would be seized, but the dtmf > > > sent to the CO was less then acceptable. > > > > > > Changing lines 1636 (from "w" to "ww") and line 1637 (from "1" to "2") > > > brought the outbound dialing back into a functional state. Since I'm > > > not a programmer, I don't really know what those lines are doing. > > > > > > Mark then used that info to write the code for implementing > > > echotrainging=800 as a configurable option. > > > > > > Does today's code support changing all three values? (Since the > > example > > > in the config files suggest two specific choices, I'd bet that using > > > a value of 600 or 1200 or whatever does cause an issue with the > > outbound > > > dialing, etc.) > > > > My report.... > > > > With our current setup we have an Adtran TotalAccess 750 connected to a > > T100P. There are 5 incoming FXO lines from Verizon. > > > > We use about 15 Polycom SIP IP500 phones. > > > > I updated to today's CVS and still noticed an echo in the middle of > > nearly every call. The echo would come in after 2 or 3 minutes, last > > for 30 seconds and then disappear. I will report on our user's > > experiences tomorrow. > > > > Regards, > > > > - Brent > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
