Yeah but doesnt help for extensions that have or require call-limit=1. --- On Tue, 31/3/09, carl Lougher <[email protected]> wrote:
> From: carl Lougher <[email protected]> > Subject: Re: [asterisk-users] Call-limit=1 breaks attended transfer > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Date: Tuesday, 31 March, 2009, 2:20 AM > > 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 > _______________________________________________ -- 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
