asterisk users - Sep 2003 - update re. Grandstream + SIP + Echo problems ..

well .. good news :)

i've just put in
txgain=1.0
rxgain=1.0
in my zapata.conf

and upgraded the Grandstream Budgettones i'm using to version 81
of the software and all seems fine .. there is still an echo but after
the first couple of seconds of call it vanishes, as the echocancelling
kicks in .. so far my client is happy :)

now .. i have one slight problem left .. although most of my SIP
phones are on a LAN connection with the asterisk server,
there are two phones which are at a remote office bridged to
my LAN via a 128k point to point ADSL .. these do not seem
to be working well, you do hear speech but the remote person
(dialled over PSTN through an X100P) hears it low and garbled ..
I am assuming it's due to the delays in stuffing 64kbits (of g711)
over a 128k link and was thinking of switching to G729.

I already have the G729 codec installed, and configured with 1
license. Can anyone give me the correct sip.conf commands 
(or whatever I need) to get the budgettones working over G729?

many thanks
Dave

Hello,

Have you succeded to use flash key to do call transfert?

Regards,

Daniel


Dave Alan Caruana a ?crit:
>well .. good news :)
>
>i've just put in
>txgain=1.0
>rxgain=1.0
>in my zapata.conf
>
>and upgraded the Grandstream Budgettones i'm using to version 81
>of the software and all seems fine .. there is still an echo but after
>the first couple of seconds of call it vanishes, as the echocancelling
>kicks in .. so far my client is happy :)
>
>now .. i have one slight problem left .. although most of my SIP
>phones are on a LAN connection with the asterisk server,
>there are two phones which are at a remote office bridged to
>my LAN via a 128k point to point ADSL .. these do not seem
>to be working well, you do hear speech but the remote person
>(dialled over PSTN through an X100P) hears it low and garbled ..
>I am assuming it's due to the delays in stuffing 64kbits (of g711)
>over a 128k link and was thinking of switching to G729.
>
>I already have the G729 codec installed, and configured with 1
>license. Can anyone give me the correct sip.conf commands 
>(or whatever I need) to get the budgettones working over G729?
>
>many thanks
>Dave
>
>
>_______________________________________________
>Asterisk-Users mailing list
>Asterisk-Users@lists.digium.com
>http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
>
>  
>
-- 
Daniel ANDRE (mailto:dandre@iris-tech.fr)
IRIS Technologies - http://www.iris-tech.com
Serveur kwartz - http://www.kwartz.com

Parking the call is a problem becasue you will not hear the parked call location
(because its a blind transfer into the parked call)..

The only solution I could think of is to call the person you want to transfer to
on the second line, then go back to the first line and blind transfer the call..
(the person you are transfering to will have to hang up after you have spoken to
them)

What is the process for transfering with the flash button??

I have always used the transfer button and the redial/send button..
> no ..
> 
> flash key can do a blind transfer, and that's about it.
> the only way to do a consultative transfer
> (ie. speak to the person you are transferring to, and then transfer)
> is by parking the call ..
> 
> i've heard that this is pretty much the definitive situation
> from what i've been reading on this list.
> 
> if anyone knows better, i'd be happy to know!
> 
> cheers
> Dave
> 
> ----- Original Message -----
> From: "Daniel ANDRE" <dandre@iris-tech.fr>
> To: <asterisk-users@lists.digium.com>
> Sent: Thursday, September 04, 2003 5:53 PM
> Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems
> ..
> 
> 
> > Hello,
> >
> > Have you succeded to use flash key to do call transfert?
> >
> > Regards,
> >
> > Daniel
> >
> >
> > Dave Alan Caruana a ?crit:
> >
> > >well .. good news :)
> > >
> > >i've just put in
> > >txgain=1.0
> > >rxgain=1.0
> > >in my zapata.conf
> > >
> > >and upgraded the Grandstream Budgettones i'm using to version
81
> > >of the software and all seems fine .. there is still an echo but
after
> > >the first couple of seconds of call it vanishes, as the
echocancelling
> > >kicks in .. so far my client is happy :)
> > >
> > >now .. i have one slight problem left .. although most of my SIP
> > >phones are on a LAN connection with the asterisk server,
> > >there are two phones which are at a remote office bridged to
> > >my LAN via a 128k point to point ADSL .. these do not seem
> > >to be working well, you do hear speech but the remote person
> > >(dialled over PSTN through an X100P) hears it low and garbled ..
> > >I am assuming it's due to the delays in stuffing 64kbits (of
g711)
> > >over a 128k link and was thinking of switching to G729.
> > >
> > >I already have the G729 codec installed, and configured with 1
> > >license. Can anyone give me the correct sip.conf commands
> > >(or whatever I need) to get the budgettones working over G729?
> > >
> > >many thanks
> > >Dave
> > >
> > >
> > >_______________________________________________
> > >Asterisk-Users mailing list
> > >Asterisk-Users@lists.digium.com
> > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > >
> > >
> > >
> > >
> > >
> >
> > --
> > Daniel ANDRE (mailto:dandre@iris-tech.fr)
> > IRIS Technologies - http://www.iris-tech.com
> > Serveur kwartz - http://www.kwartz.com
> >
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> now .. i have one slight problem left .. although most of my SIP
> phones are on a LAN connection with the asterisk server,
> there are two phones which are at a remote office bridged to
> my LAN via a 128k point to point ADSL .. these do not seem
> to be working well, you do hear speech but the remote person
> (dialled over PSTN through an X100P) hears it low and garbled ..
> I am assuming it's due to the delays in stuffing 64kbits (of g711)
> over a 128k link and was thinking of switching to G729.
> 
Rememeber G.711 is 64Kbps without overhead.. Actual bandwidth required is
somewhere around 87Kbps.. On a 128Kbps line if anything else uses the line you
will get clicks and breaks in the transmission..
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You should be able to simply use either

allow=all

in sip.conf and then change the codec order on the phone..

or if you want a little more control you can put

disallow=all
allow=g729
allow=...
 
in the sip.conf and selectivly allow the codecs.. you will probably still want
to set the order on the phone as well..

You may need to make sure on the "allow=g729" syntax.. I typed this
from memory so it could be wrong..

> has anyone got G729 and SIP working together?
> some config examples would help :)
> since I need to do this at a client where I don't
> really have internet access, or the will to root
> around mailing lists with the client breathing down
> my neck!
> 
> thsnk
> Dave
> 
> ----- Original Message -----
> From: "WipeOut ." <wipeout@linuxmail.org>
> To: <asterisk-users@lists.digium.com>
> Sent: Thursday, September 04, 2003 7:13 PM
> Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems
> ..
> 
> 
> > > now .. i have one slight problem left .. although most of my SIP
> > > phones are on a LAN connection with the asterisk server,
> > > there are two phones which are at a remote office bridged to
> > > my LAN via a 128k point to point ADSL .. these do not seem
> > > to be working well, you do hear speech but the remote person
> > > (dialled over PSTN through an X100P) hears it low and garbled ..
> > > I am assuming it's due to the delays in stuffing 64kbits (of
g711)
> > > over a 128k link and was thinking of switching to G729.
> > >
> >
> > Rememeber G.711 is 64Kbps without overhead.. Actual bandwidth required
is
> somewhere around 87Kbps.. On a 128Kbps line if anything else uses the line
> you will get clicks and breaks in the transmission..
> > --
> > ______________________________________________
> > http://www.linuxmail.org/
> > Now with e-mail forwarding for only US$5.95/yr
> >
> > Powered by Outblaze
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> 
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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