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

[Dave Alan Caruana]

hi ..

I have an asterisk system with three TDM100P (single port FXO) cards
and 10 Grandstream 100 phones connected to it .. 

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.

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 ?

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.

can anybody help please ?

cheers
Dave A Caruana