Re: [asterisk-users] conferenced transfers
On Feb 21, 2012, at 17:01 , Phil Frost wrote: On Feb 14, 2012, at 17:13 , isr...@gmail.com wrote: On the snom too Create a conferance and then press the transfer button. That will join the parties and release the receptionist Hmm...You can do that with just hitting the transfer button, or is there more? I'm using a Snom 870 with firmware 8.4.35. I set up the conference, but when I hit transfer, it presents me with a transfer party dialog. It's a bit confusing, because it's not really clear which party is being transferred. Are you using a different phone or firmware? Maybe there's a setting somewhere? Well, I happened to figure it out. On the Snom 870, when you have the conference going, both parties are in the context area. Drag one and drop it on the other and bam, they are connected, and now you can hang up while they continue to talk. Pretty obvious once you figure it out, but rather unconventional, and not documented anywhere best I can tell. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferenced transfers
On Feb 14, 2012, at 17:13 , isr...@gmail.com wrote: On the snom too Create a conferance and then press the transfer button. That will join the parties and release the receptionist Hmm...You can do that with just hitting the transfer button, or is there more? I'm using a Snom 870 with firmware 8.4.35. I set up the conference, but when I hit transfer, it presents me with a transfer party dialog. It's a bit confusing, because it's not really clear which party is being transferred. Are you using a different phone or firmware? Maybe there's a setting somewhere? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferenced transfers
As I read this, this is a regular attended transfer. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Phil Frost Sent: Tuesday, February 14, 2012 2:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] conferenced transfers I'm wondering how one might implement a transfer where a receptionist introduces a caller to the recipient in a 3-way conference before hanging up, leaving the other two parties connected. Something like this, from the perspective of the customer: Customer: Hi. I'd like to buy a widget. Receptionist: Great. Let me connect you with someone in sales. (Customer on hold) Receptionist: Hello customer. I have John here with me. John: Hello. Receptionist: John can sell you a widget. Have a great day. (Receptionist hangs up) (John and Customer continue the discussion) The problem is that in most systems I've seen, the 3-way is accomplished by the handset that initiates the conference mixing the two legs of the call. When that party (in this case, the receptionist) hangs up, the conference is over, and the other two parties are either disconnected or put on hold. I know there are ways to do server-side conferences - the challenge is making this no harder than a regular transfer, so the receptionist can do it comfortably. The usual model of dialing a number, entering a conference number, passcode, etc, is far too heavy for something as common as a transfer. In particular, I'm playing with a new Snom 870, and its drag-and-drop conference functionality is really great. However, I'm looking for any suggestions on how people skin this problem from a user interface perspective, and keep it friendly instead of frustrating for receptionists. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferenced transfers
On Feb 14, 2012, at 15:34 , Danny Nicholas wrote: As I read this, this is a regular attended transfer. No, as I understand an attended transfer, there is no 3-way period where the receptionist introduces the caller to someone else. In an attended transfer, from the caller's perspective, he's talking to the receptionist, then he's on hold, then he's talking to someone else. No different from a blind transfer, from the caller perspective. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferenced transfers
No, as I understand an attended transfer, there is no 3-way period where the receptionist introduces the caller to someone else. In an attended transfer, from the caller's perspective, he's talking to the receptionist, then he's on hold, then he's talking to someone else. No different from a blind transfer, from the caller perspective. using the Cisco-Linksys SPA Phones you would: 1) Receptionist Answers Call and hits 'Conf' button. 2) Receptionist makes call and when answered hits 'Conf' again. 3) Now everybody is talking 4) Receptions hits 'Join' button. This releases the Receptionist from the call and the other 2 parties are joined directly. -- Technical Support http://www.telesip.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferenced transfers
I think you can do the same thing with most Polycom phones. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andres Sent: Tuesday, February 14, 2012 4:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] conferenced transfers No, as I understand an attended transfer, there is no 3-way period where the receptionist introduces the caller to someone else. In an attended transfer, from the caller's perspective, he's talking to the receptionist, then he's on hold, then he's talking to someone else. No different from a blind transfer, from the caller perspective. using the Cisco-Linksys SPA Phones you would: 1) Receptionist Answers Call and hits 'Conf' button. 2) Receptionist makes call and when answered hits 'Conf' again. 3) Now everybody is talking 4) Receptions hits 'Join' button. This releases the Receptionist from the call and the other 2 parties are joined directly. -- Technical Support http://www.telesip.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferenced transfers
On the snom too Create a conferance and then press the transfer button. That will join the parties and release the receptionist -Original Message- From: Andres and...@telesip.net Sender: asterisk-users-boun...@lists.digium.com Date: Tue, 14 Feb 2012 17:10:38 To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Reply-To: and...@telesip.net, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] conferenced transfers No, as I understand an attended transfer, there is no 3-way period where the receptionist introduces the caller to someone else. In an attended transfer, from the caller's perspective, he's talking to the receptionist, then he's on hold, then he's talking to someone else. No different from a blind transfer, from the caller perspective. using the Cisco-Linksys SPA Phones you would: 1) Receptionist Answers Call and hits 'Conf' button. 2) Receptionist makes call and when answered hits 'Conf' again. 3) Now everybody is talking 4) Receptions hits 'Join' button. This releases the Receptionist from the call and the other 2 parties are joined directly. -- Technical Support http://www.telesip.net -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conferenced transfers
On Tue, Feb 14, 2012 at 3:10 PM, Andres and...@telesip.net wrote: using the Cisco-Linksys SPA Phones you would: 1) Receptionist Answers Call and hits 'Conf' button. 2) Receptionist makes call and when answered hits 'Conf' again. 3) Now everybody is talking 4) Receptions hits 'Join' button. This releases the Receptionist from the call and the other 2 parties are joined directly. I was just about to post exactly that. This is how we've taught our customers to do it, the feedback is positive. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
