Someone had a patch to retrieve the oldest call from the parking queue... maybe that could help
regards Martin On Thu, 4 Sep 2003, WipeOut . wrote: > 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" <[EMAIL PROTECTED]> > > To: <[EMAIL PROTECTED]> > > 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 > > > >[EMAIL PROTECTED] > > > >http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > Daniel ANDRE (mailto:[EMAIL PROTECTED]) > > > IRIS Technologies - http://www.iris-tech.com > > > Serveur kwartz - http://www.kwartz.com > > > > > > > > > _______________________________________________ > > > Asterisk-Users mailing list > > > [EMAIL PROTECTED] > > > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > _______________________________________________ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > ______________________________________________ > http://www.linuxmail.org/ > Now with e-mail forwarding for only US$5.95/yr > > Powered by Outblaze > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
