Re: [asterisk-users] Atxfer Command

2008-10-23 Thread Mark Michelson
David Monteagudo Sanz wrote:
> Hi,
> 
> We are testing new Asterisk 1.6.0.1 because we would like to use the 
> Attended Transfer feature and we are trying to use the new action Atxfer 
> developed for AMI.
> 
> As far as we know, it is suposed to be in this release as it can be read 
> in Digium's changelog
> 
> /New command: Atxfer. See doc/manager_1_1.txt for more details or manager 
> show command Atxfer from the CLI/
> 
> But, when we try to see information in the CLI console, this command 
> doesn't exist, neither AtxferAction works, we got a message saying that 
> this command is unknown
> 
> Are we missing something?
> 
> Thanks in advance
> 
>  David

The AMI atxfer command is not in 1.6.0 and will be in no releases of 1.6.0. It 
is in 1.6.1, for which a beta is currently available.

Mark Michelson

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Re: [asterisk-users] atxfer attended transfer feature

2007-06-20 Thread Vieri

--- Don Pobanz <[EMAIL PROTECTED]> wrote:

> > I would like to know if atxfer is supported
> This was a little confusing for me also. A week or
> so ago, someone
> pointed out that you need to include featuremap in
> your extensions.conf

Thanks Don
I'll try that.
It surprises me that such an important feature is
vaguely documented in *.



   

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Re: [asterisk-users] atxfer attended transfer feature

2007-06-18 Thread Eric \"ManxPower\" Wieling
Vieri wrote:
> --- Don Pobanz <[EMAIL PROTECTED]> wrote:
> 
>>> I would like to know if atxfer is supported
>> This was a little confusing for me also. A week or
>> so ago, someone
>> pointed out that you need to include featuremap in
>> your extensions.conf
> 
> Thanks Don
> I'll try that.
> It surprises me that such an important feature is
> vaguely documented in *.

It is not all that important of a feature as most VoIP devices 
(including ATAs) have this feature built in.

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Re: [asterisk-users] atxfer attended transfer feature

2007-06-18 Thread Don Pobanz
> I would like to know if atxfer is supported somehow
> because there seems to be little documentation for
> this feature. 
...
> I have Asterisk 1.2/Freepbx and
> features.conf has a line regarding atxfer and I set it
> to *2 (Default). While # works fine (blind transfers),
> *2 doesn't.

This was a little confusing for me also. A week or so ago, someone
pointed out that you need to include featuremap in your extensions.conf
in order to enable attended transfer, even though it appears in
features.conf. I guess blind transfer works regardless of if you include
featuremap or not. 

Don Pobanz

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Re: [Asterisk-Users] ATXFER

2006-05-15 Thread Austin Denyer

*PLONK*

[EMAIL PROTECTED] wrote:
> Please change the email address
> of [EMAIL PROTECTED] to [EMAIL PROTECTED]
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Re: [Asterisk-Users] ATXFER

2006-05-14 Thread Josué Conti
Andrew, thank´s for the attention, but necessary to decide this my problem. You he could help me?
 
Regards
 
Josué 
2006/5/14, Andrew Kohlsmith <[EMAIL PROTECTED]>:
On Friday 12 May 2006 21:00, Josué Conti wrote:> Eric, but I to continue using version 1.0.9 of asterisk, would have some
> solution?Yes, find a consultant to try and backport this feature to the version ofAsterisk you desire.-A.___--Bandwidth and Colocation provided by 
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Re: [Asterisk-Users] ATXFER

2006-05-14 Thread Andrew Kohlsmith
On Friday 12 May 2006 21:00, Josué Conti wrote:
> Eric, but I to continue using version 1.0.9 of asterisk, would have some
> solution?

Yes, find a consultant to try and backport this feature to the version of 
Asterisk you desire.

-A.
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RE: [Asterisk-Users] ATXFER

2006-05-13 Thread James Harper
I think it's most likely that it's a mail loop caused by a brain dead 'change 
of address' script.

> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Alberto Sagredo
> Sent: Saturday, 13 May 2006 17:00
> To: asterisk-users@lists.digium.com
> Subject: Re: [Asterisk-Users] ATXFER
> 
> How many times do u need to repeat it?. You could change this info via
> web list manager.
> 
> I think you need to read how to do that before sending 20 emails with
> same subject.
> 
> [EMAIL PROTECTED] escribió:
> > Please change the email address
> > of [EMAIL PROTECTED] to [EMAIL PROTECTED]
> > Thanks
> >
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> >
> 
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Re: [Asterisk-Users] ATXFER

2006-05-13 Thread Alberto Sagredo
How many times do u need to repeat it?. You could change this info via 
web list manager.


I think you need to read how to do that before sending 20 emails with 
same subject.


[EMAIL PROTECTED] escribió:

Please change the email address
of [EMAIL PROTECTED] to [EMAIL PROTECTED]
Thanks 


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Re: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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RE: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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Re: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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RE: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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Re: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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RE: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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Re: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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RE: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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Re: [Asterisk-Users] ATXFER

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RE: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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Re: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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RE: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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Re: [Asterisk-Users] ATXFER

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RE: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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RE: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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Re: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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RE: [Asterisk-Users] ATXFER

2006-05-12 Thread billy
Can someone please kill this guy's account?
Isn't there a Moderator on this list?

bp

-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, May 12, 2006 10:05 PM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] ATXFER

Please change the email address
of [EMAIL PROTECTED] to [EMAIL PROTECTED]
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__ NOD32 1.1535 (20060512) Information __

This message was checked by NOD32 antivirus system.
http://www.eset.com


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Re: [Asterisk-Users] ATXFER

2006-05-12 Thread gary
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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

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Re: [Asterisk-Users] ATXFER

2006-05-12 Thread Josué Conti
Eric, but I to continue using version 1.0.9 of asterisk, would have some solution?
Regards
 
Josué 
2006/5/12, Eric ManxPower Wieling <[EMAIL PROTECTED]>:
If you want dynamic features you must upgrade to Asterisk 1.2.xJosué Conti wrote:> Eric, thank you very much. But It could help in this case me?
 Regards Josué 2006/5/12, Eric ManxPower Wieling <[EMAIL PROTECTED]> [EMAIL PROTECTED]>>:>> Josué Conti wrote:>  > Dinesh, very obliged for the attention. I am using version 1.0.9 of>  > asterisk>  > and it is really all good with this version, only this case of
> atxfer that>  > it does not function. The function DYNAMIC_FEATURE = to atxfer in> my [>  > globals ] of extensions.conf functions in version 1.0.9?>  > It could help in this case me?
>  > Best Regards>> 1.0.x does not support this.--Now accepting new clients in Birmingham, Atlanta, Huntsville,Chattanooga, and Montgomery.___
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Re: [Asterisk-Users] ATXFER

2006-05-12 Thread Eric \"ManxPower\" Wieling

If you want dynamic features you must upgrade to Asterisk 1.2.x

Josué Conti wrote:

Eric, thank you very much. But It could help in this case me?

 


Regards

 


Josué

 

2006/5/12, Eric ManxPower Wieling <[EMAIL PROTECTED] 
>:


Josué Conti wrote:
 > Dinesh, very obliged for the attention. I am using version 1.0.9 of
 > asterisk
 > and it is really all good with this version, only this case of
atxfer that
 > it does not function. The function DYNAMIC_FEATURE = to atxfer in
my [
 > globals ] of extensions.conf functions in version 1.0.9?
 > It could help in this case me?
 > Best Regards

1.0.x does not support this.




--
Now accepting new clients in Birmingham, Atlanta, Huntsville, 
Chattanooga, and Montgomery.

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Re: [Asterisk-Users] ATXFER

2006-05-12 Thread Josué Conti
Eric, thank you very much. But It could help in this case me?
 
Regards
 
Josué 
2006/5/12, Eric ManxPower Wieling <[EMAIL PROTECTED]>:
Josué Conti wrote:> Dinesh, very obliged for the attention. I am using version 1.0.9 of> asterisk
> and it is really all good with this version, only this case of atxfer that> it does not function. The function DYNAMIC_FEATURE = to atxfer in my [> globals ] of extensions.conf functions in version 
1.0.9?> It could help in this case me?> Best Regards1.0.x does not support this.--Now accepting new clients in Birmingham, Atlanta, Huntsville,Chattanooga, and Montgomery.___
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Re: [Asterisk-Users] ATXFER

2006-05-11 Thread Eric \"ManxPower\" Wieling

Josué Conti wrote:
Dinesh, very obliged for the attention. I am using version 1.0.9 of 
asterisk

and it is really all good with this version, only this case of atxfer that
it does not function. The function DYNAMIC_FEATURE = to atxfer in my [
globals ] of extensions.conf functions in version 1.0.9?
It could help in this case me?
Best Regards


1.0.x does not support this.

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Chattanooga, and Montgomery.

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Re: [Asterisk-Users] ATXFER

2006-05-11 Thread Josué Conti
Dinesh, very obliged for the attention. I am using version 1.0.9 of asterisk and it is really all good with this version, only this case of atxfer that it does not function. The function DYNAMIC_FEATURE = to atxfer in my [ globals ] of 
extensions.conf functions in version 1.0.9?
It could help in this case me?
Best Regards
Josué
2006/5/11, Dinesh Nair <[EMAIL PROTECTED]>:
On 05/11/06 19:46 Josué Conti said the following:> functions informs "to transfer" and the transference is ok. However, if
> the agent tries to effect an attended transference the ATXFER, knocks> down the call. All the agents of this queue are with canreinvite=no ini'm guessing that the feature code for ATXFER in features.conf
 begins witha '*'. this '*' is trapped instead by chan_agent, which is a hardcodedvalue within chan_agent to hang up the call. that's the same symptomsyou're seeing.i've submitted a patch for this, which has been committed to trunk, so the
next release of asterisk should have an endcall parameter in agents.confwhich allows you to turn off this "feature". however, if you're using1.2.x, and you need it now, you can apply the 1.2 related patch from
http://bugs.digium.com/view.php?id=6897apply agent-endcall.patch for 1.2.x. as noted above, this patch has alreadybeen committed to trunk.--
Regards,   /\_/\   "All dogs go to heaven."[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+| for a in past present future; do||   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a $b."  || done; done  |+=+
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Re: [Asterisk-Users] ATXFER

2006-05-11 Thread Dinesh Nair


On 05/11/06 19:46 Josué Conti said the following:
functions informs "to transfer" and the transference is ok. However, if 
the agent tries to effect an attended transference the ATXFER, knocks 
down the call. All the agents of this queue are with canreinvite=no in 


i'm guessing that the feature code for ATXFER in features.conf begins with 
a '*'. this '*' is trapped instead by chan_agent, which is a hardcoded 
value within chan_agent to hang up the call. that's the same symptoms 
you're seeing.


i've submitted a patch for this, which has been committed to trunk, so the 
next release of asterisk should have an endcall parameter in agents.conf 
which allows you to turn off this "feature". however, if you're using 
1.2.x, and you need it now, you can apply the 1.2 related patch from 
http://bugs.digium.com/view.php?id=6897


apply agent-endcall.patch for 1.2.x. as noted above, this patch has already 
been committed to trunk.


--
Regards,   /\_/\   "All dogs go to heaven."
[EMAIL PROTECTED](0 0)http://www.alphaque.com/
+==oOO--(_)--OOo==+
| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
|   echo "The opinions here in no way reflect the opinions of my $a $b."  |
| done; done  |
+=+
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Re: [Asterisk-Users] ATXFER discussion, what's your opinion ?

2005-07-20 Thread Patrick Friedel

[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?
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Re: [Asterisk-Users] ATXFER discussion, what's your opinion ?

2005-07-20 Thread Emanuele Pucciarelli
[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
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Re: [Asterisk-Users] ATXFER discussion, what's your opinion ?

2005-07-20 Thread Michael Puchol

Alessio Focardi wrote:

Hi,

I'm experimenting attended calls tranfers and I'm a little bit
confused.




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

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Re: [Asterisk-Users] atxfer

2005-03-28 Thread asterisk
Julian,
Could you then tell  me, from which version on atxfer is available?
for instance does the 1.07 version from 19-3 support it ?
I just don't like the idea of running bleeding edge code on my pbx :-)
Kind  regards,
Joop Marijne
On Fri, 25 Mar 2005 11:54:21 +0100, asterisk <[EMAIL PROTECTED]> wrote:
 

I have installed asterisk 1.05 on debian sarge (binary package)
with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100)
I am trying to get supervised/ attended tranfer working, blind transfer
by pressing the # key works fine
atxfer => *
   

Attended transfers are only supported in CVS, not 1.0.X
Julian J. M.
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Re: [Asterisk-Users] atxfer

2005-03-25 Thread Julian J. M.
On Fri, 25 Mar 2005 11:54:21 +0100, asterisk <[EMAIL PROTECTED]> wrote:
> I have installed asterisk 1.05 on debian sarge (binary package)
> with an I4l modem and 4 x-lite softphone and 2 SIP hardphones (Yuxin 100)
> I am trying to get supervised/ attended tranfer working, blind transfer
> by pressing the # key works fine
> atxfer => *

Attended transfers are only supported in CVS, not 1.0.X

Julian J. M.
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RE: [Asterisk-Users] atxfer

2005-03-25 Thread Damon Estep
Look at the options for the dial command on the wiki, you have to use t
or T or calls are not eligible to be transferred.


> -Original Message-
> From: [EMAIL PROTECTED] 
> [mailto:[EMAIL PROTECTED] On Behalf Of asterisk
> Sent: Friday, March 25, 2005 3:54 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: [Asterisk-Users] atxfer
> 
> Hi list,
> 
> This wll be my first post, so I want to thank all the 
> developers for the great product they have created.
> 
> Now, the question,
> 
> I have installed asterisk 1.05 on debian sarge (binary 
> package) with an I4l modem and 4 x-lite softphone and 2 SIP 
> hardphones (Yuxin 100)
> 
> This all works fine, exept for som echo on the ISDN channel, 
> but I'll replace the I4L card with an AVM-C4 card next week
> 
> I am trying to get supervised/ attended tranfer working, 
> blind transfer by pressing the # key works fine
> 
> this is my features.conf
> 
> [general]
> parkext => 700  ; What ext. to dial to park
> parkpos => 701-720  ; What extensions to park calls on
> context => parkedcalls  ; Which context parked calls are in
> parkingtime => 600  ; Number of seconds a call 
> can be parked for
> ; (default is 45 seconds)
> transferdigittimeout => 3   ; Number of seconds to wait between 
> digits when
> transfering a call
> courtesytone = beep ; Sound file to play to the 
> parked caller
> ; when someone dials a parked call
> adsipark = yes  ; if you want ADSI parking 
> announcements
> pickupexten = *8; Configure the pickup extension.  
> Default is *8
> 
> [featuremap]
> blindxfer => #
> atxfer => * 
> 
> I have tried
> atxfer=>#
> and atxfer=>*2
> 
> but nothing works, also, when I disable both blindxfer and 
> atxfer, blind transfer still works.
> I have turned sip debug on, and I can see that asterisk does 
> receieve the * keys
> 
> parking calls does work btw.
> 
> Kind regards,
> 
> Joop
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