asterisk users - Mar 2009 - Call-limit=1 breaks attended transfer

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..

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

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
> 
> _______________________________________________
> -- 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
>

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
> > 
> > _______________________________________________
> > -- 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
> > 
> 
> 
> ? ? ? 
> 
> _______________________________________________
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> 
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
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>