We use call-limit set to 1 for hints. I guess i'll look into the dtmf method and debug the linksys phone to see what it uses for attended transfers.
Cheers!!!! --- On Mon, 30/3/09, Mark Michelson <[email protected]> wrote: > From: Mark Michelson <[email protected]> > Subject: Re: [asterisk-users] Call-limit=1 breaks attended transfer > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Date: Monday, 30 March, 2009, 10:50 PM > carl Lougher wrote: > > Howdy, > > Was there ever a fix for this? > > > > I have Trix 2.6 running asterisk 1.4 and have to set > an extension with call-limit=1. However that user can no > longer do attended transfers from Linkys 962 ip phone. > > > > Is there anyway around this? > > > > Cheers, > > Taff.. > > > > Yes, set call-limit to something else :P > > Seriously though, there's no "fix" for that since it is > behaving exactly as it > should. When attempting to transfer the call, Asterisk has > no way of knowing > that the new SIP INVITE it receives (in order to call the > transfer target) is an > attempt to transfer the call. It appears that the same SIP > peer is attempting to > make a second call. Since the call-limit is set to 1, > Asterisk rejects the > second call attempt. > > I haven't tried this yet, but it may actually be possible > to use DTMF transfers > when the call limit is that low since Asterisk is the one > that actually > initiates the new call to the transfer target instead of > the transferer's phone. > To use DTMF transfers, you need to set a DTMF sequence in > features.conf and use > the 't' or 'T' flag (depending on which party should have > the ability to > transfer the call) in your calls to Dial() or Queue(). > > Why do you have the call-limit set to 1, anyway? > > Mark Michelson > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
