[asterisk-users] Call-limit=1 breaks attended transfer

[Mike]

I think the comment was more along the lines of "use call-limit, but put a
number higher than 1".

Mike

> -----Original Message-----
> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-
> bounces at lists.digium.com] On Behalf Of carl Lougher
> Sent: Monday, March 30, 2009 21:21
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Call-limit=1 breaks attended transfer
> 
> 
> 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 <mmichelson at digium.com> wrote:
> 
> > From: Mark Michelson <mmichelson at digium.com>
> > Subject: Re: [asterisk-users] Call-limit=1 breaks attended transfer
> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-
> users at lists.digium.com>
> > 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
> >
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> 
> 
> 
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