Adrian Marsh wrote:
Hi All,
In my old telco days (SS7), if I was wanting to hand back a call to
the network for transfer to a different PSTN number, there was a
specific SS7 action I could take, which send the call back to the
network, which in turn then routed the call appropriately. It
We have occasional problems with failed transfers. The PSTN call comes
into Cisco Call Manager, then to asterisk over a SIP trunk and then to
an asterisk controlled SIP phone. The SIP phone transfers back to a
CallManager controlled SCCP phone which sometimes fails.
Is there a wait to let
Hi All;
Can I do transfer for the call from zap/1 to zap/2
(both are fxs)? All what I need is to add the t
argument for the Dail function?
And how can I transfer to be in that senario: zap/1
dial a code to transfer for zap/2, once zap/2
answered, then he can talk with zap/1 (where the third
Hi there,
i'm beginner in Asterisk.. I have situation:
when a caller ,some SIP account on local network, dial outside number
trough some Zap channel, asterisk have to automaticly transfer call to
another local SIP channelIn other words, he have to change
caller..
For example:
I have SIP
The vast majority of what I've done with Asterisk has been with the
Grandstream GXP-2000's. These phones work great for us for everything
*except* speaker quality is quite poor and appears to be half-duplex.
So now that we've bought and are using 40 GXP-2000's we're doing some
testing on other
I have a SIP voice server which I want to place an Asterisk server in
front of to handle call routing.
At the moment I can call the apps on the server fine, but it cannot
transfer to another extension via asterisk.
Even attempting a call from the voice server to an asterisk extension
I've realize the follow situation:
the phone A call the phone B,
the phone B park, throgth the application map, the phone A,
the phone B call another phone C,
the phone B, througth the application-map, transfer the phone C
to the phone parked A.
I use the follow line in feature.conf
trans_CA
Hi All-
I have an asterisk server and GXP2000. If I want to send a call to
someone else (external), I can transfer the call where I can announce
it, and then send it over. But what I'd like is to start a 3-way
conference, and then drop out. But if I do a conference button on
the phone,
I'm having a tough time figuring out how to do something. If I have an
operator (which could potentially be in their own context) and an
internal-only context, is it possible to make it so the operator can
call the internal-only context but *NOT* transfer calls to it?
The idea is that the
Hi Mark -
I'm having a tough time figuring out how to do something. If I have an
operator (which could potentially be in their own context) and an
internal-only context, is it possible to make it so the operator can
call the internal-only context but *NOT* transfer calls to it?
Sort of.
Noah Miller wrote:
Sort of. You can create a special extension in the operator's context
with a Goto() statement. Something like this:
[operator]
exten = 100,1,Goto(internal,prompt,1)
Then in the internal context:
[internal]
exten = prompt,1,Background(who-do-you-want-to-call)
:99.9/99.9 FC:95.5390 LC:95.5390 R:95.9108
P:95.9108
M:97.0282 C:98.6951 )
X-pstn-settings: 3 (1.:1.) s fc lc gt3 gt2 gt1 r p m c
X-pstn-addresses: from [EMAIL PROTECTED] [db-null]
Subject: Re: [asterisk-users] Transfer Call to Cell Phone
X-BeenThere: asterisk-users
Ryan Goldberg wrote:
Alternatively, the first line could be:
exten = 101,1,Dial(SIP/${EXTEN}Zap/4/12185551212,30,tpm)
which would dial both the desk and the cell at the same time...
I've tried doing things like this. What got me was that SIP technology
allows for the phone not
Dear All,
I have a problem with call transfer. When I dial a number and then I want to
transfer current call to an extension, I'm getting disconnected. Transfering
incoming call works fine. I'm using macro for dialing.
extensions.conf:
[from-internal]
ignorepat = 9
exten =
Thanks work perfect,,
Otis Surratt Jr. / [EMAIL PROTECTED]
OCOSA Communications, LLC
PH: 918.585.9882 x 205 Fax: 918.585.5857
Visit Tulsa's Best Internet Data Center:
http://www.ocosa.com/hosting/colo/index.asp
works perfectThanks
--
Otis
John Faubion wrote:
We do have full features on our lines so both lines are free once the
transfer is complete. We also have toll calls on our lines so it would
not be a problem, so I do not have to worry about ATT dropping the
The issue
We do have full features on our lines so both lines are free once the
transfer is complete. We also have toll calls on our lines so it would
not be a problem, so I do not have to worry about ATT dropping the
The issue really isn't whether you have the ability to make toll calls on
your line. The
John Faubion wrote:
We do have full features on our lines so both lines are free once the
transfer is complete. We also have toll calls on our lines so it would
not be a problem, so I do not have to worry about ATT dropping the
The issue really isn't whether you have the ability to
Ryan Goldberg wrote:
OCOSA ListAcc wrote:
Can you give an example of creating an extension which points to a cell
phone. Secondly how can you have if no one answers an extension it dials
the cell number next. That maybe answered in the example. I have the
system setup so it just
by the way selling does not depend on the amount of lines you have and
we are very productive trust me
True, very true. There are lots of very productive sales people that don't
need a phone at all. From the paper boy to car dealers, lots of sales don't
require many phone lines. Of course at
John Faubion wrote:
by the way selling does not depend on the amount of lines you have and
we are very productive trust me
True, very true. There are lots of very productive sales people that don't
need a phone at all. From the paper boy to car dealers, lots of sales don't
Polycom Phones
1. New call
2. Press 9 access outside line
3. Dial Cell Number
4. Transfer the call that way.
Once you initiate a new call you will tie up the second line. Your asterisk
box will now be bridging the two lines. The lines will stay tied up until
the salesman drops the call.
One
Well seems like I am already doing first method minus the extension. We
do have full features on our lines so both lines are free once the
transfer is complete. We also have toll calls on our lines so it would
not be a problem, so I do not have to worry about ATT dropping the
call. I tried to
OCOSA ListAcc wrote:
Can you give an example of creating an extension which points to a cell
phone. Secondly how can you have if no one answers an extension it dials
the cell number next. That maybe answered in the example. I have the
system setup so it just dials out which ever line is
Hello All,
I apologize if this question has already been answered but how do you
transfer a call to a cell phone or another land line outside the PBX?
Setup
I have two pots lines into my current Asterisk Box. I have an outsides
sales guy who wants to work off his cell phone or transfer his
I have two pots lines into my current Asterisk Box. I have an outsides
sales guy who wants to work off his cell phone or transfer his calls
from his extension and the main sales extensions. How can I do this right?
Do it right? You really haven't provided enough information to make the
right
John
thanks for the input.
forget about my right way ok!
by the way selling does not depend on the amount of lines you have and
we are very productive trust me
I have seen a million dollar corp work off four lines so your statement
is quite vague...
Otis
John Faubion wrote:
to transfer
the call to 3+VM number).
- Original Message -
From: Drew Gibson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, June 15, 2007 9:01 PM
Subject: Re: [asterisk-users] Transfer caller direct
PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, June 15, 2007 9:01 PM
Subject: Re: [asterisk-users] Transfer caller direct to voicemail
Hi Wes,
thanks for the suggestion but I have gone a simpler route suggested by
Leonardo
Baehr
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Drew Gibson
Sent: Tuesday, June 12, 2007 11:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Transfer caller direct to voicemail
Hi
Hi,
Our operator frequently gets requests to transfer a call directly to
voicemail in order for the caller to leave a message without disturbing
the callee. Basicly, I'm looking for a blindxfer to vm.
My first thought was to prepend a digit (eg 7) to the extension but this
does not fit well
Hello Drew;
Assuming your extensions is 105 let's see the dialplan:
exten = 105,1,Dial(SIP/105,30,Tt)
exten = 105,n,Hangup
exten = *XXX,1,Answer
exten = *XXX,n,VoiceMail(${EXTEN:[EMAIL PROTECTED])
exten = *XXX,n,Hangup
I think this should work for what you want.
Regards;
Leonardo Kamache
-users-
[EMAIL PROTECTED] On Behalf Of Drew Gibson
Sent: Tuesday, June 12, 2007 11:15 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Transfer caller direct to voicemail
Hi,
Our operator frequently gets requests to transfer a call directly
Discussion
Subject: [asterisk-users] transfer call sip to zap
how to transfer a call from sip channel to zap channel
thanks
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On Fri, 25 May 2007, Cosmin Prund wrote:
It just works. Simply transfer your call to the desired extension and
let Asterisk take care of the details.
Indeed. A key appeal of Asterisk does lie precisely in that it
abstracts, to a considerable degree, the chore of dealing with the
how to transfer a call from sip channel to zap channel
thanks
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Any ideas on this?
Closest thing that comes to mind is FOP : http://www.asternic.org/
hth
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Phil Menico wrote:
I used autodial to allow a user to make a call by clicking on a web
directory and placing a call file into the Asterisk outgoing
directory. That works perfectly for me.
What if I want to click on the web directory and transfer my existing
call? Is there a comparable
, 2007 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Transfer via CTI
I used autodial to allow a user to make a call by clicking on a web
directory and placing a call file into the Asterisk outgoing
directory. That works perfectly for me.
What if I want
I used autodial to allow a user to make a call by clicking on a web
directory and placing a call file into the Asterisk outgoing
directory. That works perfectly for me.
What if I want to click on the web directory and transfer my existing
call? Is there a comparable interface?
Thank you.
Phil
Tim Panton ha scritto:
I'd be tempted to simplify things even more by removing the codec
negotiation
and have all the boxes be _forced_ to use alaw.
Tim
The same, can't hear nothing (also upgraded to 1.4.2)
I still have quite a bad feeling about opening a bug like mediaonly
doesn't works
I am not up on the detail of the media-only transfer, but I did
notice that in the second example
the weas one you get a VNAK from 192.168.52.94:4569 in response to
the first
post transfer VOICE packet, then everything starts to go wrong !
In the first example you also see VNAKs and the
Kevin P. Fleming ha scritto:
OK, then you'll need to get a verbose/debug console trace, and
preferably a packet capture of the IAX2 traffic on 'Server', and post a
bug on bugs.digium.com with those files attached.
___
While setting up the servers to
Simone Cittadini wrote:
I post once again here, sorry for the verbosity, if then in your opinion
there's still something wrong with * internals and not with my
understanding of the configs I'll open the bug.
I would encourage you to open the bug anyway; I am currently at a trade
show in
Hi,
im trying to use bling transfer on asterisk 1.4.0. it doesnt work i have
configured # key for transfer when i press it nothing happens. also my
dynamic features do not get executed. everything works fine in asterisk
1.2.13 which is on another workstation. can somebody plz help? here is my
I've setup this simple configuration to test the new mediaonly iax
feature in 1.4 :
Input (client) - Server (routing) - Termination
transfer=notransfer=mediaonly transfer=no
all the machines are in the same 192.168.0.x net
the routing Server in the middle has iaxusers realtime
Simone Cittadini wrote:
I've setup this simple configuration to test the new mediaonly iax
feature in 1.4 :
What version of Asterisk exactly?
Input (client) - Server (routing) - Termination
transfer=notransfer=mediaonly transfer=no
This doesn't make any sense; the 'Server' is
Kevin P. Fleming ha scritto:
I've setup this simple configuration to test the new mediaonly iax
feature in 1.4 :
What version of Asterisk exactly?
1.4.1
Input (client) - Server (routing) - Termination
transfer=notransfer=mediaonly transfer=no
This doesn't make
Simone Cittadini wrote:
So the config is :
realtime mysql users on the server to auth the customers (Input) and one
user entry in iax.conf on the Termination to auth the Server
transfer=mediaonly is set in [general]
OK, then you'll need to get a verbose/debug console trace, and
preferably a
Possibly the called party is not sending their DTMF properly? maybe
experiment with inband/rfc2833/etc in the CALLED party's peer definition
Denis V. Gudtsov wrote:
Hello!
I'm using asterisk 1.2.13, in extension.ael is set Dial(SIP/${EXTEN},12,tT)
but only calling party can do forward. How
Hello!
I'm using asterisk 1.2.13, in extension.ael is set Dial(SIP/${EXTEN},12,tT)
but only calling party can do forward. How to configure '*' to take this
possibility to called party?
ps
both calling/called use sip
--
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As far as I can tell, the only way to do this using Polycom soundpoint
phones and NOT asterisk's built-in blindxfer function, is to hit their
Transfer button first, and then the Blind softkey that appears on the
screen. Then continue as normal; dial the number and hit Send I
believe. If you
I'm sure this was asked before, but I can't seem to make this work...
If a customer dials one of our DIDs, and the operator transfers that
call to another employee, the Caller ID doesn't seem to do what I would
expect it to. I would expect it to show the original caller's ID.
Example:
John calls
Not sure about others, but on Polycoms a blind transfer sends
original callerid, screened sends operators callerid
On Feb 19, 2007, at 8:55 AM, Rob Schall wrote:
I'm sure this was asked before, but I can't seem to make this work...
If a customer dials one of our DIDs, and the operator
Not sure about an Asterisk only call transfer but in an Asterisk/SER
environment the SER server will use the REFER method to perform the
transfer. In this case ehe caller ID needs to be the contents of the
Refer-By header of the SIP message. Not the contents of EXTEN
-Steve
Jerry Jones
I probably have a screened transfer setup. Is that just a setting
somewhere I can easily change? I'm trying to avoid making users press
extra keys, like #1 or anything like that.
Rob
Jerry Jones wrote:
Not sure about others, but on Polycoms a blind transfer sends original
callerid, screened
Not an asterisk setting. It is how the endpoints perform the transfer.
On Feb 19, 2007, at 9:21 AM, Rob Schall wrote:
I probably have a screened transfer setup. Is that just a setting
somewhere I can easily change? I'm trying to avoid making users press
extra keys, like #1 or anything like
Hi all,
Me again ... for a new question! Again
Here the scenario:
A call B ( A -- B)
B transfer to C ( A -- C)
In this case, how can I have the B caller id number and the A caller id
number?
Thanks a lot for your help
Thomas
I am running some Polycom phones and have Auto Answer setup(*51
initiates that when you call an extension)
With an attended transfer you can take a call, hit transfer,
*51extension, announce the call and if the person wants it, complete
the transfer, the call is now on speaker at the end.
On 01/28/07 18:52 Florian Overkamp said the following:
Nokia seems to have done something like this in their E-series (E60 etc)
with Avaya and Cisco. Anyone have a lowdown on the technical stuff there ?
i think that's a FMC (fixed mobile convergence) client which both avaya and
cisco wrote
Hi all,
We are looking at VoIP over Wifi and I was wondering if anybody had any
ideas around automatically transfering calls after an RTP timeout? The
idea is this: a user is on a call with their IP phone and the connection
drops (e.g. user walks out of range of their Wifi AP). Using RTP
Hi,
Ray Jackson wrote:
transfer to that number. That way the call can stay up rather than the
user having to redial. Is there a way of transferring back to the *
dialplan on RTP timeout to perform some additional steps (instead of
just hanging up?)
Nokia seems to have done something like
Hello, I've tried to transfer a IAX call to a number configured on a
traditional
PBX, but it doesn't work. I have a traditional PBX connected with a zap channel
to Asterisk in the following way:
IAX/SIP client -- Asterisk (FXO) -- (FXS) traditional PBX --- OFFICE
Phones
Asterisk is connected
I don't think that the first priority (exten = _44XX,1,Answer) is ok,
have you tried without it?
Try not answering and post what happens.
On 1/17/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello, I've tried to transfer a IAX call to a number configured on a
traditional
PBX, but it doesn't
Hi guys,
This should be has an easy answer for you, my users are complaining
that when they press # and then ear gorgeous Allison Transfer the
timeout is very small, they must enter immediatly the extension to
transfer the call.
Is it possible to change this?
;transferdigittimeout = 3 ;
Discussionasterisk-users@lists.digium.com
Subject: [asterisk-users] #Transfer - Timeout is configurable?
Date: Fri, 20 Oct 2006 15:54:40 +0100
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@lists.digium.com
Sent: Tuesday, October 10, 2006 8:45 PM
Subject: [asterisk-users] transfer from VM to Cell Phone
Hi Folks,
I'm not sure if this is possible, but I'd like to give users the
option of transfering to an employee's cell phone when they get to
their greeting. This is a feature
Hi Folks,
I'm not sure if this is possible, but I'd like to give users the
option of transfering to an employee's cell phone when they get to
their greeting. This is a feature that is common on Nortel KSUs.
Is there an easy way to do this on a per employee basis? I can see
how it can be done
it.
- Original Message -
From: Mr. Jones [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, October 10, 2006 8:45 PM
Subject: [asterisk-users] transfer from VM to Cell Phone
Hi Folks,
I'm not sure if this is possible, but I'd
]To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.comSent: Tuesday, October 10, 2006 8:45 PMSubject: [asterisk-users] transfer from VM to Cell Phone Hi Folks,
I'm not sure if this is possible, but I'd like to give users the option of transfering to an employee's cell
Hello asterisk-users,
I'm currently investigating a problem related to the Transfer app and
DTMF tones via SipInfo.
My setup depends on:
Asterisk 1.2.10
Zaptel 1.2.8
libpri 1.2.3
Elmeg IP 290 (snom190)
Wildcard TE400 (E1)
The following dialplan is given:
exten = 555, 1, Transfer(554);
exten
Eric ManxPower Wieling wrote:
I don't know if this is even possible. I might be totally wrong but
once this call is on the cell network, how are you gonna communicate
with asterisk?? From what I understand, while the voice (RTP) traffic
still travels through asterisk, You have no access to
Technically DTMF should be a signaling thing, but I believe Asterisk
must stay in the media stream if you want to use t/T/w/W. This may have
changed in 1.4. canreinvite=no in sip.conf would keep Asterisk in the
media stream.
Steve Glaus wrote:
Eric ManxPower Wieling wrote:
I don't know
Hi,
My setup is the
following: Voip provider---(SIP DID)---Asterisk box(SIP
through a termination provider)---multiple cell
phones.
The cell phones each
have their extension (201,202,203,204) and I'd like to be able to have them
transfer a call to somebody
Mike wrote:
Hi,
My setup is the following: Voip provider---(SIP
DID)---Asterisk box(SIP through a termination
provider)---multiple cell phones.
The cell phones each have their extension (201,202,203,204) and I'd
like to be able to have them transfer a call
procesed by Asterisk when the call hits the cell?
-Original Message-
From: Steve Glaus [mailto:[EMAIL PROTECTED]
Sent: Wednesday, October 04, 2006 1:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfer feature - howto?
Mike wrote:
Hi
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfer feature - howto?
Mike wrote:
Hi,
My setup is the following: Voip provider---(SIP
DID)---Asterisk box(SIP through a termination
provider)---multiple cell
: RE: [asterisk-users] Transfer feature - howto?
Ah. I'd like to know what others think, but if you're right than it's a
lost cause.
I thought Asterisk kept some sort of control over the call.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Do you know for a fact that inband DTMF is being
procesed by Asterisk when the call hits the cell?
Well it seems I'm wrong but how do you setup asterisk to process inband
dtmf? I dial the cell phone with the 't' in the dial string.
When I hit anything from my cell phone, asterisk doesn't
I don't know if this is even possible. I might be totally wrong but once
this call is on the cell network, how are you gonna communicate with
asterisk?? From what I understand, while the voice (RTP) traffic still
travels through asterisk, You have no access to any kind of signalling.
Please
Looks Great
Thanks a lot!!!
On Wed, 16 Aug 2006, C F wrote:
I think this will help you:
${TRANSFER_CONTEXT} Context for transferred calls
Just set it before the Dial statement.
Don't forget to read this first:
Hi!!!
Somebody knows how to avoid that one FXS channel can transfer between
FXO's channels but can do it between FXS's channel?
In other words:
An FXS channel make a call to a FXO channel, press flash-hook, it takes
another fxo channel and dial another telephone. FXS hangup and both FXO
I think this will help you:
${TRANSFER_CONTEXT} Context for transferred calls
Just set it before the Dial statement.
Don't forget to read this first:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+features.conf
On 8/16/06, Ing. Germán González B. [EMAIL PROTECTED] wrote:
Hi list,
how can I realize explicit call transfer? I want to transfer a call
which I answered to another phone and it the other one answers I want to
hang up so that my resources are freed.
Is that possible with Zaptel or which channel can I use else?
TIA, Christophorus
On Thursday, August 10, 2006 10:43 AM Christophorus Laube wrote:
Hi list,
how can I realize explicit call transfer? I want to transfer a call
which I answered to another phone and it the other one answers I want
to hang up so that my resources are freed. Is that possible with
Zaptel or
Hello,
I am running TrixBox.
if already in a call session from ZAPTEL to SIP, the user want to
transfer the call to a different extension.
The user have to dial *extension ?
Any configuration is needed to be done in trixbox?
Thanks
Victor
___
- Original Message -
From: Victor Moreno
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Fri, 28
Jul 2006 06:57:48 -0300
Subject: [asterisk-users] Transfer call in SIP
Hello,
I am running TrixBox.
if already in a call session from ZAPTEL to SIP, the user want
Transferring a call from 80014154 to 2944051.
Asterisk is sending an ACCEPT message to the party transfer the call,
immediately followed by a DECLINED message. There appears to be NOTHING logged
in between. Anyone got any ideas?
Jul 14 08:06:23 VERBOSE[16688] logger.c: Transfer to 2944051 in
Hi!Where can I find more informations about the Transfer() application in a All-SIP environment?
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Goto voip-info.org and search it.On 7/13/06, Benjamin Stocker [EMAIL PROTECTED]
wrote:Hi!Where can I find more informations about the Transfer() application in a All-SIP environment?
___--Bandwidth and Colocation provided by Easynews.com --
Thanks for all the help so far on this, but I was wondering if there was
a way of simulating an attended transfer from the AMI or dialplan ?
Julian.
Moises Silva wrote:
Piece of cake Julian:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect
Regards
On
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Transfer call via AMI or dialplan
Thanks for all the help so far on this, but I was wondering if there was
a way of simulating an attended transfer from the AMI or dialplan ?
Julian.
Moises Silva wrote
At the moment when one of our users wants to transfer a call, they press
the transfer button on the phone, enter the extension and away they go.
Is there any way to do this via the AMI or dialplan ? I want them to
push a button on the screen rather than using the phone itself.
Julian
Piece of cake Julian:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect
Regards
On 6/19/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
At the moment when one of our users wants to transfer a call, they press
the transfer button on the phone, enter the
: Monday, June 19, 2006 9:31 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Transfer call via AMI or dialplan
At the moment when one of our users wants to transfer a call, they press
the transfer button on the phone, enter the extension and away they
go.
Is there any way to do
Thanks for the tip. No idea why I missed this.
Off the top of your head, does this support attended xfer, or is it a
blind xfer facility ?
Julian.
Moises Silva wrote:
Piece of cake Julian:
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Redirect
Regards
On
This would be more like a blind transfer :)
On 6/19/06, Julian Lyndon-Smith [EMAIL PROTECTED] wrote:
Thanks for the tip. No idea why I missed this.
Off the top of your head, does this support attended xfer, or is it a
blind xfer facility ?
Julian.
Moises Silva wrote:
Piece of cake Julian:
If you want something ready to go I have implemented this in Snap.
Here is a screen shot of the transfer feature:
http://www.snapanumber.com/portals/0/transfer.png
and website:
http://www.snapanumber.com/
As well, you may also want to look at ADM (Asterisk Desktop Manager):
On Sun, 2006-06-04 at 17:46 +0800, Ronald Wiplinger wrote:
*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# ##
Attended Transfer *2
One
*CLI show features
Builtin Feature Default Current
--- --- ---
Pickup*8 *8
Blind Transfer# ##
Attended Transfer *2
One Touch Monitor *1
Disconnect Call
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