[Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of local echo, & questions about call transfers

Dave Alan Caruana david at melita.net
Tue Aug 5 00:06:41 MST 2003


my error .. the cards are X100P which is why I wrote FXO.

The Grandstream phones are on a LAN, the * server connects to the phonelines
via the X100P cards. When I call from the Grandstream phones onto the PSTN
there is a VERY big amount of echo, ie. I can hear myself in the earpiece.

cheers
Dave

----- Original Message -----
From: "WipeOut ." <wipeout at linuxmail.org>
To: <asterisk-users at lists.digium.com>
Sent: Tuesday, August 05, 2003 8:50 AM
Subject: Re: [Asterisk-Users] SIP + Grandstream 100 + TDM100P = lots of
local echo, & questions about call transfers


> > hi ..
> >
> > I have an asterisk system with three TDM100P (single port FXO) cards
> > and 10 Grandstream 100 phones connected to it ..
>
> The TDMx00P cards are FXS cards.. :)
>
> >
> > 1st question:
> > when i phone out
> > or receive a call from one of the SIP phones onto the PSTN, there is
> > a LOT of local echo in the handset .. the PSTN end of the call does not
> > here this echo, but it's VERY annoying on the SIP end of things ..
> > the echo seems to be about 0.3 seconds delayed to the speech ..
> > there is no echo on incoming voice, just an echo of my own voice
> > as I speak.
>
> What are you using to connect to the PSTN?? X100P, T100P, E100P, I4L,
Chan_Capi....
>
> >
> > 2nd question:
> > using a grandstream phone & asterisk, if I hear another phone ringing,
> > how can answer it from the phone infront of me? eg. if extension 6003
> > is ringing, and i have phone number 6004, how can I answer it ?
>
> You need to setup call groups, search through the archives cos I rememeber
a thread on this a short while ago..
>
> >
> > 3rd question:
> > can someone give me some "starter hints" to configure call parking ?
> > I haven't managed to find a direct way to transfer a call from phone
> > to phone except using blind transfer and I want the person initiating
> > the transfer to speak to the receiving person before actually passing
> > the call.
>
> As far as I know there is no facility to do a consultative transfer on the
GS phones.. Only a blind transfer.. Maybe it will come later..
>
> >
> > can anybody help please ?
> >
> > cheers
> > Dave A Caruana
> >
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users at lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
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