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

[carl Lougher]

Yeah but doesnt help for extensions that have or require call-limit=1.

--- On Tue, 31/3/09, carl Lougher <c_lougher at yahoo.co.uk> wrote:

> From: carl Lougher <c_lougher at yahoo.co.uk>
> 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: 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 <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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