asterisk users - Aug 2003 - 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 .. 

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

> 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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> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> 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
> 
An echo at the begining of a call is normal as the * and phone
"trains" themselves but this should dissappear after about 30 seconds
to 1 min..

So my only suggesttions are..

First make sure you have echocancel=yes and echocancelwhenbridged=yes in your
zapata.conf..

If that doesn't help try lowering the volume on the sip handset and play
with the rxgain= and txgain= in zapata.conf for the X100P's..

Other than that I don't really know what else you can try..

Later..
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> could you send me the exact syntax for rxgain / txgain?
> I think that might help towards my problem
> becuase i'm having to turn the handset volume all the
> way up ..
> 
> thanks
> Dave
You can use either a percentage or a number IIRC..

Somthing like..

rxgain=5%
txgain=5%

or

rxgain=0.4
txgain=0.4

and I thing that you can use negative values as well..

I am not sure what the minimum and maximum values are I use percntages..

Hope that helps..
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Dave Alan Caruana wrote:
>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 ?
>  
>
You hit *8#  and you will pick up any call you have setup callgroups for.

sip.conf:

[6004]
type=friend
username=6004
canreinvite=no
callgroup=1
pickupgroup=1
host=dynamic

[6003]
type=friend
username=6003
canreinvite=no
callgroup=1
pickupgroup=1
host=dynamic

>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.
>
extensions.conf:

[local]
include => parkedcalls
[default]
exten => 701,1,ParkedCall(701)
exten => 702,1,ParkedCall(702)
exten => 703,1,ParkedCall(703)
exten => 704,1,ParkedCall(704)
exten => 705,1,ParkedCall(705)
exten => 706,1,ParkedCall(706)
exten => 707,1,ParkedCall(707)
exten => 708,1,ParkedCall(708)
exten => 709,1,ParkedCall(709)
exten => 710,1,ParkedCall(710)


parking.conf:

[general]
parkext => 700                          ; What ext. to dial to park
parkpos => 701-720                      ; What extensions to park calls on
context => parkedcalls                  ; Which context parked calls are in
parkingtime => 45 

>can anybody help please ?
>
>cheers
>Dave A Caruana
>
>
>
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>  
>