[asterisk-users] Transfers on AgentLogin()

[Matt Riddell]

What do you get when you type show features?

On 9/6/08, Mark Hamilton <mark.h at cage151.com> wrote:
> Hi James,
>
> Thank you very much for a detailed reply. (Matt, sorry about earlier, I
> totally missed the part you said about the t option)
> To answer, yes the Queue command does have t and T passed to it. This is how
> I tested it. Agent1 is on eyeBeam and he's the one who will need to do an
> attended transfer to a queue. So, let's say the shortcode to the queue is 3.
> Agent1 gets a call, presses the # (even though the transfer sequence is set
> to #2.. immediately, Agent1 heard "Transfer", which means just the # was
> enough to put it in the transfer mode) and the minute Agent1 presses 3, it's
> a blind transfer.
>
> canreinvite=no and so dtmf=auto. It doesn't seem to be picking up the
> feature codes set in features.conf for some reason. So # is doing the
> transfer, even though the only thing uncommented in features.conf was
> atndxfer, which was set to *2 and then to #2 since *2 was doing a hangup
> (the hangup sequence for agentlogin). dtmfmode couldn't be set to info
> because eyeBeam is used by Agent1 and DTMF wasn't being recognized when the
> agent was trying to login to the queue.
>
>
> [1013]
> type=friend
> qualify=yes
> nat=yes
> host=dynamic
> dtmfmode=auto
> context=manila
> canreinvite=no
> callerid=Agent <1013>
> call-limit=10
>
> Please help
> Thanks!
>
>
>> -------- Original Message --------
>> Subject: Re: [asterisk-users] Transfers on AgentLogin()
>> From: "James Sneeringer" <jsneerin at gmail.com>
>> Date: Fri, September 05, 2008 10:57 pm
>> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>> <asterisk-users at lists.digium.com>
>>
>> Since AgentLogin() essentially keeps a channel to the agent open all
>> the time, a normal SIP transfer will do exactly as you say. That is,
>> it will try to send the agent's login session into queue, which isn't
>> what you want.
>>
>> As Matt suggested, you need to pass the "t" option to the Queue()
>> application. This will let your agents perform a DTMF transfer using
>> the codes defined in features.conf. The agent basically dials a short
>> code while talking to the caller. Asterisk intercepts it, and then
>> prompts the agent for the extension to transfer the call to. Look in
>> features.conf for more information.
>>
>> Fair warning, I have never needed to use this feature, so I can't
>> attest to exactly how it behaves. We use dynamic agent logins, so
>> we've never had to deal with AgentLogin(). This allows us to do normal
>> SIP transfers.
>>
>> Also, you will probably have to do one of two things in your sip.conf.
>> One, set "canreinvite" to "no" to keep Asterisk in the call path, that
>> way it can intercept the DTMF tones. Or, two, set "dtmfmode" to
>> "info", so that DTMF tones are converted to SIP INFO messages, which
>> Asterisk will see.
>>
>> At least, that's how I think it works. :)
>>
>> -James
>>
>>
>> On Sun, Aug 31, 2008 at 3:15 PM, Mark Hamilton <mark.h at cage151.com> wrote:
>> > I've tried the regular, xfer button on xlite, dial 100 (to transfer to
>> > the
>> > queue), and hit go back to line 1 and hit xfer again. But it's
>> > AgentLogin(),
>> > so it transfers the full persistent connection to the queue instead of
>> > the
>> > call itself and this causes the transferring agent to logout.
>> >
>> > Either that, or I'm doing something wrong. There is no documentation out
>> > there so I don't know how it would work for AgentLogin().
>> >
>> > -----Original Message-----
>> > From: asterisk-users-bounces at lists.digium.com
>> > [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matt
>> > Riddell
>> > Sent: August 30, 2008 6:18 PM
>> > To: Asterisk Users Mailing List - Non-Commercial Discussion
>> > Subject: Re: [asterisk-users] Transfers on AgentLogin()
>> >
>> > What did you try and how did it fail? Are you using the t option in
>> > queue?
>> >
>>
>> _______________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-users mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-users
>

-- 
Sent from Gmail for mobile | mobile.google.com

Matt Riddell
Director
VentureVoIP