What do you get when you type show features? On 9/6/08, Mark Hamilton <[EMAIL PROTECTED]> 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" <[EMAIL PROTECTED]> >> Date: Fri, September 05, 2008 10:57 pm >> To: "Asterisk Users Mailing List - Non-Commercial Discussion" >> <[email protected]> >> >> 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 <[EMAIL PROTECTED]> 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: [EMAIL PROTECTED] >> > [mailto:[EMAIL PROTECTED] 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 _______________________________________________ -- 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
