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

[WipeOut .]

> 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
> 
> 
> 
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