[Asterisk-Users] tdm (and x100p?) echo - fix is coming!
Wojciech Tryc
wojtek at tryc.ca
Thu Jun 24 17:20:30 MST 2004
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" <wojtek at tryc.ca>
To: <asterisk-users at lists.digium.com>
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" <mwless at mindworks.net>
> To: <asterisk-users at lists.digium.com>
> Sent: Thursday, June 24, 2004 7:16 PM
> Subject: RE: [Asterisk-Users] tdm (and x100p?) echo - fix is coming!
>
>
> >
> > > -----Original Message-----
> > > From: asterisk-users-admin at lists.digium.com [mailto:asterisk-users-
> > > admin at lists.digium.com] 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
> > Asterisk-Users at lists.digium.com
> > 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
> Asterisk-Users at lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
More information about the asterisk-users
mailing list