Re[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
I think that's mostly right, but it should also be a native xfer function working the same way regarding of the user agent, some sort of common ground we can trust for installation with mixed devices. By the way: anyone got experience in attended trasfer with snom ? :) Alessio Focardi PF Oh, you mean the completely natural feeling put them on hold, dial PF new party, tell them you have a transfer, hit transfer? I want some of PF whatever kool-aid the person who thought that one up had. I still feel PF like I'm losing a call every time I do an attended transfer. In my opinion there should be only one transfer function, let suppose it's called by #. - You get a call - You want to transfer it - You hit # - You are presented a tone - You dial the extension you want to transfer to Now the hard part - If you hang up prior of the other party has answered you get an unattended transfer if, for any reason the other party dont answer (busy, no answer, wrong extension etc) call should be bounced back to you - If you stay on the phone and the other party answers you talk to him, introduce the call then hitting # again will switch back and forth between the call you are tranfering and the transfer party if you hang up call is trasfered to the other party if the other party hangs up you get back to the original call Eventually another function key can be enabled (let's say *): if you do an attendend xfer transfer the * key will put in a conference the original call, you and the other party you are transfering. If any of the 3 hangs up while conferencing the conference should stay up with the 2 remaining. What do you think about this flow ? -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[4]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
Hello Adam, In my opinion there should be only one transfer function, let suppose it's called by #. AG Wrong, which other phone system have you used where every time you try AG and use some IVR that says Enter your xyz number followed by the # key AG and you end up being interrupted by asterisk to transfer the call ?? Well as you can see it was an example, actually you have to decide this mapping in features.conf, so what's the point ? Let say is *# or any other sequence :) Eventually another function key can be enabled (let's say *): if you do an attendend xfer transfer the * key will put in a conference the original call, you and the other party you are transfering. If any of the 3 hangs up while conferencing the conference should stay up with the 2 remaining. AG Nope, because if there are three parties: AG A - You AG B - Outside caller 1 AG C - Outside transfer party AG When you hangup, you don't want the other two legs to stay up, AG potentially forever depending on your hangup detection etc... I know what I want! :) Why not, I'm announcing a call, then going conference, then leaving because I already did my part, why the other 2 calls have to be disconnected ... because hangup detection works bad ? What do you think about this flow ? AG Any SIP phone (decent one) should have much more intuitive/instructive AG transfer process. All I'm asking is a native function that can be used regardless of the UA, if you got such functions integrated in the phone, better yet, is up to you to choose then. -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
On Thu, 2005-07-21 at 09:59 +0200, Alessio Focardi wrote: PF Oh, you mean the completely natural feeling put them on hold, dial PF new party, tell them you have a transfer, hit transfer? I want some of PF whatever kool-aid the person who thought that one up had. I still feel PF like I'm losing a call every time I do an attended transfer. In my opinion there should be only one transfer function, let suppose it's called by #. Wrong, which other phone system have you used where every time you try and use some IVR that says Enter your xyz number followed by the # key and you end up being interrupted by asterisk to transfer the call ?? Eventually another function key can be enabled (let's say *): if you do an attendend xfer transfer the * key will put in a conference the original call, you and the other party you are transfering. If any of the 3 hangs up while conferencing the conference should stay up with the 2 remaining. Nope, because if there are three parties: A - You B - Outside caller 1 C - Outside transfer party When you hangup, you don't want the other two legs to stay up, potentially forever depending on your hangup detection etc... What do you think about this flow ? Not only have you suggested pretty much what we have, except you've made it worse by taking away the # and * keys... If you are on a zap channel, just hook flash (or press flash/recall/whatever) and transfer the call, complete with conference option. Any SIP phone (decent one) should have much more intuitive/instructive transfer process. Regards, Adam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ATXFER discussion, what's your opinion ?
Hi, I'm experimenting attended calls tranfers and I'm a little bit confused. In usual pbx's normaly there is no difference between an attended call transfer and a blind one: you just hit transfer then dial the extension you want the call to be transfered. If you stay on the phone you can talk to the other party, then, when you hangup, he will get the call. If you hang immediately after the transfer sequence the call is just transfered, and if the other party is busy or does not answer the transfered call is bounced back to you again. That's how pbx's users are expecting call transfer to work, is there a way to reproduce this behavior in asterisk ? For what I can see it's not possible and you will have to select two codes, one for blind and one for attended tranfers What do you think about it ? -- Best regards, Alessio mailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATXFER discussion, what's your opinion ?
Alessio Focardi wrote: Hi, I'm experimenting attended calls tranfers and I'm a little bit confused. SNIP I honestly think that transfers is one thing that Asterisk should improve a LOT to be able to stand up to even the most cheapo taiwanese no-name PBXs, which support attended transfers out of the box. I've had two possible clients refuse an Asterisk installation because attended transfers were unreliable. I honestly didn't know how to explain that a feature available in PBXs for decades was not available or didn't always work. I don't know how the current HEAD is going, but so far, attended transfers weren't available in stable. Regards, Mike ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
Hello Michael, Wednesday, July 20, 2005, 11:54:40 AM, you wrote: MP Alessio Focardi wrote: Hi, I'm experimenting attended calls tranfers and I'm a little bit confused. MP SNIP MP I honestly think that transfers is one thing that Asterisk should MP improve a LOT to be able to stand up to even the most cheapo taiwanese MP no-name PBXs, which support attended transfers out of the box. That's exactly my opinion: isn't ironic that the only function joe sixpack will use in a pbx is the worst implemented ? Maybe we can try to write down a sort of flow chart of a new transfer function and then set up a bounty, anyone else would like to join me ? -- Best regards, Alessiomailto:[EMAIL PROTECTED] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
On Wed, 2005-07-20 at 12:26 +0200, Alessio Focardi wrote: Wednesday, July 20, 2005, 11:54:40 AM, you wrote: MP Alessio Focardi wrote: I'm experimenting attended calls tranfers and I'm a little bit confused. MP I honestly think that transfers is one thing that Asterisk should MP improve a LOT to be able to stand up to even the most cheapo taiwanese MP no-name PBXs, which support attended transfers out of the box. That's exactly my opinion: isn't ironic that the only function joe sixpack will use in a pbx is the worst implemented ? Maybe because most asterisk PBX's are implemented using business class softphones rather than analogue phones? Most business class SIP phones (and even grandstream phones) allow attended/non-attended transfers with asterisk-stable and/or asterisk-head... Personally I love the polycom IP600 phones, because they 'guide' the user on-screen to help them do the most simple task (like answer the phone :) PS, and yes, these are helpful features, because otherwise there really would be more support calls like the one I got today (read on for a laugh). User calls and reports that their cordless phone extension is not working (connected via TDM card). I suspect that it is the usual problem, and I need to unload/reload the wcfxs driver and restart asterisk because I haven't replaced their card with the latest revision and every few weeks it does this. Anyway, so I ask what is happening, and he says he presses the pickup button and hears 3 loud beeps, and then the phone hangs up. I ask him if the base station is plugged in, and I then hear something along the lines of Oh... ummm, yeah... thanks, cya... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 9345 4395[EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: Re[2]: [Asterisk-Users] ATXFER discussion, what's your opinion ?
That's exactly my opinion: isn't ironic that the only function joe sixpack will use in a pbx is the worst implemented ? Maybe because most asterisk PBX's are implemented using business class softphones rather than analogue phones? Most business class SIP phones (and even grandstream phones) allow attended/non-attended transfers with asterisk-stable and/or asterisk-head... I think that's mostly right, but it should also be a native xfer function working the same way regarding of the user agent, some sort of common ground we can trust for installation with mixed devices. By the way: anyone got experience in attended trasfer with snom ? :) Alessio Focardi ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATXFER discussion, what's your opinion ?
[EMAIL PROTECTED] wrote: By the way: anyone got experience in attended trasfer with snom ? :) Works OK here, using a lightly patched CVS from a couple of months ago and the instructions that they provided (HOLD, dial extension, speak to said extension, then TRANSFER). Of course this isn't going to work nicely when you have more than one call on hold, so I'm looking forward to testing the patch that Frank Sautter wrote about on these lists some days ago! Bye, -- Emanuele ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] ATXFER discussion, what's your opinion ?
[EMAIL PROTECTED] wrote: That's exactly my opinion: isn't ironic that the only function joe sixpack will use in a pbx is the worst implemented ? Maybe because most asterisk PBX's are implemented using business class softphones rather than analogue phones? Most business class SIP phones (and even grandstream phones) allow attended/non-attended transfers with asterisk-stable and/or asterisk-head... I think that's mostly right, but it should also be a native xfer function working the same way regarding of the user agent, some sort of common ground we can trust for installation with mixed devices. By the way: anyone got experience in attended trasfer with snom ? :) Alessio Focardi Oh, you mean the completely natural feeling put them on hold, dial new party, tell them you have a transfer, hit transfer? I want some of whatever kool-aid the person who thought that one up had. I still feel like I'm losing a call every time I do an attended transfer. Is there a _technical_ (e.g. SIP) limitation on this? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
