Hi,
We are having a problem that is preventing users from using *2 to manage an
attended transfer.
After dialling *2, asterisk places the call on hold, but you can only dial one
digit before it times out, and the cli says:-
[2011-06-29 18:33:55] WARNING[29877]: res_features.c:938
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
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
--- 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
I would like to know if atxfer is supported somehow
because there seems to be little documentation for
this feature. I know most people expect a good SIP/IAX
phone to do the job but I think it's nice to be able
to do attended trasnfers with a simple ATA-connected
analog phone. I have Asterisk
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
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
Hi,
I cannot get attended working on my Asterisk 1.2.9.1 during an inbound
call via an ISDN card to a Snom SIP phone.
The called party is not able to transfer even if :
1 - atxfer is enabled (set to *7) in in features.conf
2 - the dial option is set to value 't'
3 - I see * and then 7 on
*PLONK*
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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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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,
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:
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Thanks
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:
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of [EMAIL
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
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]
mailto:[EMAIL PROTECTED]:
Josué Conti wrote:
Dinesh, very
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?
Please change the email address
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Can someone please kill this guy's account?
Isn't there a Moderator on this list?
bp
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Hi all.
Iam with the following problem:
I have one perfectly queue functioning, however when the agent receives a call and effects a transference the blind people, blindxfer, asterisk functions informs to transfer and the transference is ok. However, if the agent tries to effect an attended
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
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
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
Hi there i just
can't find an answer on the featuremap config i want all phones to use the same
method for transfering a call on all phones but i just can't get the atxfer or
other functions to work on my grandsteam and sipura spa 2000
it's confusing for
users with different phones to
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
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
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
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
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
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
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
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
[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
[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
Hi all.
At the end, i get atxfer with sip dowloading head cvs version of
asterisk and this is ok, but now i have errors with h323.
following the instructions i could compile h323 channel and load it, but
when i call from sip to h323 or viceversa, i obtain this.
debug
-
May 4
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]
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,
- 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
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
Hi.
I've been trying to develop this module since some time now.
CVS already has a dial version with atxfer. When trying this, using the
modifiers tT and having configures features.conf accordingly, i havent
been able to use such a feature in the called party.
I also tried using t and T
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