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

[Steve Davies]

I have found that you get good results by setting a per-device
GROUP_COUNT(), which prevents dialling if it is non-zero, and setting
call-limit to 999.

In Asterisk 1.0.x there were separate in- and out-bound call limits,
but IIRC this was pretty broken, and was removed.

See http://www.voip-info.org/wiki/view/Asterisk+func+group

Hope that helps.
Steve

2009/3/31 carl Lougher <c_lougher at yahoo.co.uk>:
>
> 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
>> >