[Asterisk-Users] update re. Grandstream + SIP + Echo problems ..

[Dave Alan Caruana]

I was referring to the transfer button (sorry)

what again is the way you are using to transfer calls ?
so far what i'm doing is after accepting a call,
parking it .. then phoning the guy who wants the
call and telling him "the call is parked on 701"
for example ..

cheers
Dave

----- Original Message -----
From: "WipeOut ." <wipeout at linuxmail.org>
To: <asterisk-users at lists.digium.com>
Sent: Thursday, September 04, 2003 7:08 PM
Subject: Re: [Asterisk-Users] update re. Grandstream + SIP + Echo problems
..


> 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 at iris-tech.fr>
> > To: <asterisk-users at 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 écrit:
> > >
> > > >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 at lists.digium.com
> > > >http://lists.digium.com/mailman/listinfo/asterisk-users
> > > >
> > > >
> > > >
> > > >
> > > >
> > >
> > > --
> > > Daniel ANDRE (mailto:dandre at iris-tech.fr)
> > > IRIS Technologies - http://www.iris-tech.com
> > > Serveur kwartz - http://www.kwartz.com
> > >
> > >
> > > _______________________________________________
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> > > Asterisk-Users at lists.digium.com
> > > http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
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>
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