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

[carl Lougher]

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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