asterisk users - Jan 2006 - Bug in attended transfer or as expected?

Hi all,

I have had quite a few customer complaints about attended transfer
cutting off callers.

The problem is when reception is busy she doesn't always wait for
someone to answer the call, however hanging up a ringing transfer on
attended also hangs up the caller.

I have checked the scripts I don't *think* this is a dial plan error but
if anyone has this working correctly on Asterisk 1.2.1 I will certainly
look again.

Personally I (and all the customers I have spoken to) consider this to
be incorrect behaviour and not how legacy PBX's function.

Retraining users to know the difference between blind and attended
transfer and their corresponding actions keys is one solution however I
think it's a very poor one and still prone to error, especially when the
error is cutting off a customer.

For a start it's a large undertaking (3 receptionists but on a weekend
any of the 15 sales staff may pickup the main incoming number if busy)
but the biggest problem is how do I explain to a customer that this
amazing cutting edge VOIP PBX cant do something that their 10 year old
phone system can :)

We do have 85% Snom 360's in this particular dealership so have switched
to using the phones attended transfer but this isn't my preferred
solution.

If anyone has any ideas to fix this that would be hugely appreciated, or
just a comment on whether this is expected behaviour.

Thanks again

Alex

---------------------------------------
Alex Barnes
Engineering Support
Ubiquity Software
---------------------------------------




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.

> The problem is when reception is busy she doesn't always wait for
> someone to answer the call, however hanging up a ringing transfer on
> attended also hangs up the caller.
If you have enabled "Disconnect Call" feature, then you can hangup
with "*0" for example, that will hangup only the current call, not the
call on hold.

Regards

--
"Su nombre es GNU/Linux, no solamente Linux, mas info en
http://www.gnu.org"

> -----Original Message-----
> From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-
> bounces@lists.digium.com] On Behalf Of Moises Silva
> Sent: 23 January 2006 15:35
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Bug in attended transfer or as expected?
> 
> > The problem is when reception is busy she doesn't always wait for
> > someone to answer the call, however hanging up a ringing transfer on
> > attended also hangs up the caller.
> 
> If you have enabled "Disconnect Call" feature, then you can
hangup
> with "*0" for example, that will hangup only the current call,
not the
> call on hold.
> 

Just so I understand is this the expected call flow?

- Receptionist picks up a call and wants to transfer
- Dials "*1" (attended transfer key)
- Transfer extension starts ringing
- New call comes in so receptionist decides to answer that one
- Receptionist dials "*0" to hang up the current call (expecting the
person on hold to be connected to the ringing extension)
- Original caller either gets answered or continues with the dial plan
for that extension (in my case that is forward back to the reception
queue after 30seconds)

If the receptionist decides to stop waiting for a ringing transfer, to
get the caller back she can dial "*2" which I think is good.

I still think that this should be much simpler and that hanging up an
attended transfer "mid transfer" should change it to being a blind
transfer.  Cutting off the caller instead is pretty terrible.


Thanks for the help

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 steve@daviesfam.org
> Sent: 24 January 2006 05:53
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Bug in attended transfer or as expected?
> 
> 
> 
> > > The problem is when reception is busy she doesn't always wait
for
> > > someone to answer the call, however hanging up a ringing transfer
on
> > > attended also hangs up the caller.
> 
> Its the phone that is responsible for hanging up both calls, not
Asterisk.
> 
> On the SNOM phones you can disable "disconnect on on-hook" to
stop the
> phone from doing that.
> 
> Steve
> 
Sorry think you misunderstand, I don't really want the phones to have to
do the attended transfer by merging two lines locally to the phone.

Asterisk 1.2.x now supports attended transfer natively (well kind of
supports it :P)

Thanks though

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.

Sorry for bumping my own thread but just hoping that someone out there
can help, I don't want to raise a bug on * if this is a strange issue
with my dial plan

Just to clarify this is attended transfer using asterisk and not a phone
feature (not joining two held calls etc)

Could someone with 1.2.x give the following call flow a try?

- Connect a call between two phones on *
- Called party initiates attended transfer, e.g *1 depending on how your
features.conf is setup.
- Dial internal extension
- Ringing from transfer extension
- Hangup original called party

*I / most people would expect the caller to be connected to the ringing
extension but instead on my * they get disconnected.

In other words in this scenario users expect the attended transfer to
switch to the same call flow of blind transfer.

I would look into the code but am a Java / PHP dev :(

Thank you for your help

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 Moises Silva
> Sent: 25 January 2006 16:27
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Bug in attended transfer or as expected?
> 
> Its not so hard to look into the source code and make small changes.
> Im not sure how hard it would be to implement what you want, but i
> have tested it, and yes, you are right, the call get disconnected, and
> i agree that souldnt be that way. You may want to open a feature
> request in bugs.digium.com
> 
> Regards
> 
Thanks Moises much appreciated.

I will do both I think since this is having a knock on effect now with
the new GROUP() method of limiting calls I have to not count outgoing
calls but thats another story :)

I will have to dig out my university course notes on C I guess :(


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.