Hi All, Does any one know if attended call transfer has been added into the STABLE release of asterisk yet? Potentially using a mix of phones would create confusion in a user base, any ideas on attended transfer or how to achieve this / mods to dial plan etc would be greatly appreciated. I have been on an almost vertical learning curve with Asterisk and Linux for 6 months this is just about my last challenge (for now - haha). Many thanks Chris Blunt -- SIP: 248189@fwd.pulver.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/c8aecdda/attachment.htm
> > >Does any one know if attended call transfer has been added into the STABLE >release of asterisk yet? >Any news? I am also looking for #-Transfers for asterisk-stable. Thomas
Hi, We're trying to setup attended call transfer, but we have not been able to find the required configuration. Blind transfer works fine using the # key, but we don't like the fact that the transferring extension does not have any info on what happened to the call. Any pointers to setting up proper attended transfer? Thanks, Denis The information contained in this email is confidential and may be privileged. It is intended for the addressee only, if you are not the intended recipient please notify the sender and delete the email immediately. The contents of this email must not be disclosed or copied without the senders consent. We cannot accept any responsibility for viruses. Any views expressed in this message are those of the individual sender, except where the sender specifically states them to be the view of Philip Toledo Limited -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051015/598062b8/attachment.htm
Hi! I am new with asterisk and I have my first problem with the attended call transfer feature. When a call comes in, i take the call and i would like to transfer it. So I press the * button (mapped for the attended transfer in features.conf) and the number for the receiving extension. The receiving extension rings and the call can be taken there. So far so good. Now to my problem: If I hook on the handset BEFORE the receiving extension take the call, the caller from outside will be disconnected and the receiving extension stops ringing. Shouldn't the receiving extension keep on ringing until the call is taken? Independent of hooking on the handset or not! (as it is with the blind transfer feature) The incoming line and all of the extensions are POTS, connected on a tdm400p card. I use asterisk 1.2.4 and zaptel 1.2.3 Hope someone could help me. Thx, Tom
Hi all, I had problem running MySQL on FC3 and what I found from googling was that SELinux should be disabled to make MySQL work n FC3. Now I am concerned about Asterisk, is it a good idea to disable SELinux. Or is there any other way to make MySQL work without disabling SELinux? Thanks, Zeeshan A Zakaria
> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Moises Silva> Sent: 10 February 2006 01:42 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] attended call transfer > > this is a Normal behaviour, nevertheless i dont think is a correctbehaviour. Several weeks ago other user asked the same, i suggested him to open a feature request on bugs.digium.com, check for that> > regards >Hi Yes that was me, this is still a big issue for us. Unfortunately we only have 1.2.1 installed on our live / dev boxes at the moment and when I registered an account on the bug tracker and read the "rules" it said you must have tested the issue on the very latest CVS head. I have been up to my eye balls the last couple of weeks so haven't had time to do this. I didn't want to raise this as a feature request as in my opinion this has to be a defect as attended transfer is basically unusable for a commercial environment (unless there exists a business that doesn't have a problem cutting off its customers :P ) HTH Alex Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you.
Hi! I got this answer from the digium support:> >You may wish to use the attended transfer by either using hold or >flashhook instead of the # features.conf attended transfer option. >From >a phone connected via a Zap channel, you would need to hit flash. Then >enter the extension which you wish to transfer to. You can either >hangup once it begins ringing or wait until the remote end answers. >Once the remote ends answers you may announce the caller then hangup. >This method works better than the attended transfer option available in >the features.conf file. You must have your dial plan configured >properly to allow for transfers. Dial plan configuration also falls >under our Express Technical Support Service. > > >Regards, >Chris HozianThat means, that an attended transfer is possible as it would be liked in this mailing-list-thread. I tried to make call transfers with the flash button, but it doesnt work. "threewaycalling" and "transfer" is set to "yes" in my zapata.conf. But when I hit the flash-button - nothing happens. All incoming calls triggers a Dial() on all extensions with the Dial-Parameter "t" - so a call transfer should be possible. (Are here further configurations necesseary in my dialplan?) What am I doing wrong? Tom
John is absolutely correct - in the PBX world a transfer is a transfer, regardless of whether it is blind or attended. How many PBX phones out there have two different transfer buttons, one for blind and one for attended? Zilch. It's the user's behavior that determines whether or not the transfer is blind or attended. So having two separate types of transfers isn't ideal. I wouldn't call it a bug or a defect, just a design choice that might not be optimal now that Asterisk is showing up in traditional office environments. I don't know the whole history but it seems like the dichotomy of blind vs. attended transfers has been in * since the beginning. Questions for the community: is an "integrated" transfer feature valuable to you? If so, would you be willing to put out a bounty? (In other words, is it just a nice feature or is it so important that you'd be willing to pay a few bucks for it...) Last question, but possibly the most important: what have you done, if anything, to get around the split between blind and attended transfers? -MC -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John Novack Sent: Sunday, February 12, 2006 12:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] attended call transfer That certainly is the way it SHOULD work. Blind and attended transfer should be able to be initiated the same way. It certainly is the most efficient logical way. Attended transfer should revert to blind simply by the initiating party hanging up. Most "legacy" hybrid key/pbx systems work that way, and have for many a year Most users expect transfer to work that way. I would consider that a defect or bug, not a new feature request. John Novack Ira wrote:> At 12:57 AM 02/12/2006, you wrote: > >> Why don't you think it is correct behaviour? The purpose of attended >> transfer is that you consult with the party before transferring with >> hooking, otherwise it would be a blind transfer for which there is a >> blind transfer option. > > > So let's consider an operator, takes a call and decides to attended > transfer it to Bob because it's slow and she want's to ask something, > but the instant she picks that option another call comes in. If > hanging up converted it to blind transfer she could get on with her > work and answer the next call, as it is she needs to wait till > something happens and possibly lose the next call. OK, it's a stretch> but it does seem like hanging up the call is just wrong! > > Ira >_______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
> > Questions for the community: is an "integrated" transfer feature > valuable to you?Yes, merging blind and attended transfer would be valuable for me!> If so, would you be willing to put out a bounty?Maybe. Depends on how much it would be. Tom
On Sunday, February 12, 2006 9:36 PM John Novack wrote:> That certainly is the way it SHOULD work. Blind and attended transfer > should be able to be initiated the same way....> I would consider that a defect or bug, not a new feature request.I second that. Regarding the bounty: Once someone can at least make an educated guess as to how much this improvement would cost and if there is some sort of guarantee that this functionality will make it to Asterisk soon and not longer in SVN for several months, I am sure there are enough people kicking in a few bucks to make this happen. Kind regards, JP
Useful discussion on this. There are some other functions in this which need to be addressed. For example if doing an attended transfer and the recipient phone number goes to voicemail, you have to wait for the timeout to reconnect to the original caller - unless someone know differently. There should be a reconnect hot key. Again this is comparable to a conventional PABX where the attended transfer puts the caller on hold and pushing a button reconnects. Paul
Yes - in a traditional PBX environment the transferring station has the ability to pull the call back by pressing a sequence of keys. In some PBX's, pressing the transfer key twice, like a double-click of a mouse, will pull the call back. In some analog environments, pressing the flash key twice will pull the call back. (Pressing the flash key only once will, in some PBX's, initiate a 3-way call.) Thanks for bringing this up, Paul. Any other suggestions about making transfer even better? -MC -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul Redstone Sent: Monday, February 13, 2006 5:29 AM To: Asterisk User Subject: RE: [Asterisk-Users] attended call transfer Useful discussion on this. There are some other functions in this which need to be addressed. For example if doing an attended transfer and the recipient phone number goes to voicemail, you have to wait for the timeout to reconnect to the original caller - unless someone know differently. There should be a reconnect hot key. Again this is comparable to a conventional PABX where the attended transfer puts the caller on hold and pushing a button reconnects. Paul _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Michael Collins > Sent: 13 February 2006 00:58 > To: jnovack@stromberg-carlson.org; Asterisk Users Mailing List - Non- > Commercial Discussion > Subject: RE: [Asterisk-Users] attended call transfer > > Questions for the community: is an "integrated" transfer feature > valuable to you? If so, would you be willing to put out a bounty?(In> other words, is it just a nice feature or is it so important thatyou'd> be willing to pay a few bucks for it...) Last question, but possibly > the most important: what have you done, if anything, to get around the > split between blind and attended transfers? > > -MC >To get around this issue we have had to only use attended transfer with Snom phones which are easy to train our users on. People with DECT / ??? phones have been told to only ever use blind (only about 10 out of 50 extensions so not the end of the world). We would definitely chip in some money for this to happen but again it would have be in the stable release soon as we cant deploy anything except stable. Cheers Alex Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you.
> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Eric "ManxPower" Wieling > Sent: 13 February 2006 18:18 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] attended call transfer > > As I understand it, these issues only apply to DTMF based transfers. > Why not use the transfer feature of your IP Phone or the Zap FXS port? > _______________________________________________Few reasons really: 1) Only one training method required for all users (we have multiple types of phone within any single deployment what if a Snom user picks up a call at a grandstream user's desk). 2) If we decide to start using a different phone for new deployments we wouldn't need to reprint training materials. 3) Some phones don't support attended transfer at all. 4) Causes problems when you want to limit the number of calls any device can make or receive. For example I have had to alter a script to allow attended transfers to work but this means that internal -> internal calls don't count towards the maximum limit. 5) Better CDR maybe, rather than having two calls and then joining them together Asterisk itself knows it's a transfer so can log the action much better. 6) The Snom 360 phones we use as reception phones do not behave well either going from attended to blind (I think) although they don't hang up the caller but rather than bring this up with Snom I believe this is a PBX issue. 7) Asterisk shouldn't claim to have attended transfer capability if it has fundamental flaws. I cant think of anything else right now but am sure there's more :-) But you are entirely correct, there is no option at the moment but to use the IP phones attended transfer. HTH Alex Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you.
JCC, The issue boils down to this: how much work does the human have to do to get the calls routed to the right place? In a traditional PBX environment, a receptionist does not have to choose beforehand whether he/she is going to do a blind or attended transfer. Like I said before, how many different types of transfer keys do you have on a "normal" PBX telephone? Just one. A transfer is a transfer, regardless of whether it is blind or attended. If the phone system forces the human to choose which type of transfer to do before he executes the operation then it is limiting his ability to do his work effectively. Ira's example about needing the flexibility to do a blind or attended transfer is not at all far fetched. Our receptionists answer over 1000 calls per day. If they had to choose between blind and attended transfers on every single call then that would increase their workload significantly. Thankfully, they don't have to because our PBX allows them to do blind or attended transfers by allowing them to release the transferred call simply by hanging up, regardless of whether or not the destination station has answered yet. That is the "proper" way of executing transfers in 10's of millions of offices around the world using legacy PBX's and key systems. With Asterisk pushing the envelope forward in so many other areas, does it really seem like a good idea not to have what many feel is a very basic feature? You've already seen 3 or 4 who would be willing to kick in a few $ to get it into the stable release, and I'd wager that dozens more would pay a modest sum to add this feature to their Asterisk installations. It's certainly less expensive than adding another receptionist. -MC -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of JCC Sent: Monday, February 13, 2006 5:03 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] attended call transfer Huh? I don't understand.. If the operator can't pick up the call you need more operators to compensate. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Ira Sent: Sunday, February 12, 2006 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] attended call transfer At 12:57 AM 02/12/2006, you wrote:>Why don't you think it is correct behaviour? The purpose of attended >transfer is that you consult with the party before transferring with >hooking, otherwise it would be a blind transfer for which there is a >blind transfer option.So let's consider an operator, takes a call and decides to attended transfer it to Bob because it's slow and she want's to ask something, but the instant she picks that option another call comes in. If hanging up converted it to blind transfer she could get on with her work and answer the next call, as it is she needs to wait till something happens and possibly lose the next call. OK, it's a stretch but it does seem like hanging up the call is just wrong! Ira -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.1.375 / Virus Database: 267.15.6/257 - Release Date: 02/10/2006 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users