The file /var/log/asterisk/full will contain helpful log messages that
show how Asterisk is internally handling the call. It may be necessary to
increase the verbosity of the log to get more details however.
From the linux command line, you can follow these steps to get a copy of
the relevant
We have a plain vanilla installation of AsteriskNOW using Digium D40/50 phones.
All transfers are failing from any source to any extension with the message
that is not a valid extension. Does anyone have any ideas about where to
begin looking for the source of that error?
Phil Ledon
--
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call transfer problem.
Hi all,
I have a user who is having trouble transferring calls, using a
Grandstream
GXP2xxx.
Here's the use case that I've seen:
I call the user from phone A and he answers
Hi all,
I have a user who is having trouble transferring calls, using a
Grandstream GXP2xxx.
Here's the use case that I've seen:
I call the user from phone A and he answers on phone B.
Then, he hits the transfer button on his phone and dials an extension
that is reachable by him, but not by
the transfer?
--Don
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike Diehl
Sent: Monday, February 24, 2014 12:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users
, February 24, 2014 12:24 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Call transfer problem.
Hi all,
I have a user who is having trouble transferring calls, using a Grandstream
GXP2xxx.
Here's the use case that I've seen:
I call the user from phone A and he
Hi,
is possible that two sip extensions: user-1 and user-2 are connected and I
want to transfer the call from user-1 to a third user user-3.
I know it is possible through feature keys mapping in features.conf, but I
want to do this through AMI or Asterisk CLI Commands?
Please suggest if possible?
Hi faheem,
You can do this:
ACTION: Redirect
Channel: Channel ID
Context: Context
Exten: Exten
Priority: Priority
Regards,
Qasim
On Thu, May 16, 2013 at 3:13 PM, Muhammad Faheem faheem2...@gmail.comwrote:
Hi,
is possible that two sip extensions: user-1 and user-2 are connected and I
want
On 9/4/12 3:04 am, Takehiro Matsushima wrote:
// I don't know what's difference t and T.
T allows the caller to transfer. t allows the called user to transfer.
You very rarely want Tt - since I doubt you want an incoming caller to
be able to transfer their call all over the place. You
Thank you so much.
OK, I understood that to transfer the call t is usually used, is it right?
And I should write so in my last mail.
t and T are described with same sentences in official wiki...
Regards,
Takehiro Matsushima
2012/4/9 Chris Bagnall aster...@lists.minotaur.cc:
On 9/4/12 3:04
Thanks everyone. I was using the Tt flag but in the wrong place in the dial
application.
Cheers
On Mon, Apr 9, 2012 at 4:54 PM, Takehiro Matsushima
takehiro.dream...@gmail.com wrote:
Thank you so much.
OK, I understood that to transfer the call t is usually used, is it
right?
And I should
Hi.
Maybe you forgotten specify to allow the transferring a call.
Try with tT options in Dial() in extensions.conf.
// I don't know what's difference t and T.
--
Takehiro Matsushima
takehiro.dream...@gmail.com
2012/4/7 Rizwan Hisham rizwanhas...@gmail.com:
Hi All,
I am using asterisk
If call comes into PBX-A and based on the DNIS it comes into my box PBX-B
my box then says ring phone C. Person answers. They want to transfer the
call
to a phone going back out PBX-A. All this is fine of course.
my question is when phone C transfers the call is there a way PBX-B can
drop out
call transfer
call transfer from reception to other extensions.
Question: Details of Extensions
On Tue, 2010-02-16 at 17:25 +0530, cool dude wrote:
call transfer
call transfer from reception to other extensions.
Question: Details of Extensions
Reception - 2000
Sales - 2001
Accounts - 2002
any call comes it should be received by extenion 2000, n if person
wants to talk to
Tuesday, February 16, 2010, 12:55:12 PM, cool wrote:
call comes it should be received by extenion 2000, n if person wants to
talk to Sales, receptionist should put the caller on hold than connect
to Sales i.e exten 2001, while on hold the caller should hear music on
hold,now sale exten can
Hello, I am having a problem with getting call transfer to work.
This is what is happening:-
1) External call comes in on SIP from a DDI provider
2) The call is answered by extension 204
3) Then extension 204 presses the Xfer button and the call is
placed on hold
4)
Is there a way to transfer a call, while in the middle of the call, using
DTMF?
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On 15/7/09 3:07 PM, Michael wrote:
Is there a way to transfer a call, while in the middle of the call, using
DTMF?
Yep, just pass the t or T options to the dial command and set it up in
/etc/asterisk/features.conf
--
Cheers,
Matt Riddell
Director
in the extension you want to transfer too.
Thank you,
Brad Finberg
- Original Message -
From: Michael as...@nettrust.co.nz
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Cc:
Date: Tuesday, July 14 2009 11:07 PM
Subject: [asterisk-users] call transfer
On Thu, Jan 15, 2009 at 4:09 AM, Rilawich Ango maillist...@gmail.com wrote:
Hi,
I wonder how I can relate the CDR records for the case of call
transfer. I can't find their relationship in CDR. Any can advice?
ango
You may want to read this thread.
Hi,
I wonder how I can relate the CDR records for the case of call
transfer. I can't find their relationship in CDR. Any can advice?
ango
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To
You could simply have it Dial() to wherever it needs to go at the end of
the script.
2009/1/6 Rajkumar S rajkum...@gmail.com
Hi,
I have a typical call center with queues and agents added via
AddQueueMember. One of my requirement is to implement a forgot
password function. If a caller does
Hi,
I have a typical call center with queues and agents added via
AddQueueMember. One of my requirement is to implement a forgot
password function. If a caller does not remember the password, he
calls up an unauthenticated line and the agent manually authenticates
him. Then the caller should have
Ciao Noah,
What flags do you have in your Dial() statement? If you want both
parties to be able to transfer with the features.conf transfer, you
need to have 'Tt' in your dial statement, like this:
Dial(IAX2/user:[EMAIL PROTECTED]/exten,20,Tt)
Bingo. That was the problem.
Thanks a lot,
--
Hi Andrea -
I have two asterisk servers, an IAX trunk between and some SIP users
registered
to each server.
The scenario is this: user A, registered to PBX 1, calls user B, registered to
PBX 2. Then A wants to transfer the call using the features.conf method (in my
case, **), but is
Hello everybody,
I have two asterisk servers, an IAX trunk between and some SIP users registered
to each server.
The scenario is this: user A, registered to PBX 1, calls user B, registered to
PBX 2. Then A wants to transfer the call using the features.conf method (in my
case, **), but is unable
Hello everybody,
I have two asterisk servers, an IAX trunk between and some SIP users registered
to each server.
The scenario is this: user A, registered to PBX 1, calls user B, registered to
PBX 2. Then A wants to transfer the call using the features.conf method (in my
case, **), but is unable
Hello all
I'ts my first message here although I follow the list for about a month now.
I'd like to ask a question because googling was not so helpful. Here it is:
Is there any way to transfer the Incoming CallerID (the one who called my
office) when I transfer the call to an internal
On Sat, Jun 7, 2008 at 8:24 AM, Theodore Patsiouras
[EMAIL PROTECTED] wrote:
If my secretary or anyone else picks up the call when the line is transferred
in my ext then I have the internal caller ID. Can I have somehow the
External callerID?
Look at the channel variables that contain the
Hi,
I use asterisk 1.2.23
I have the following issue with transfer:
I call from from sipA to sipB
when sipB press transfer (not blanktransfer) sipA hear the music until sipB
put down the phone, in this time sipC is ringing but sipA don't hear
anything
can you tell me where to lookup the
On 9/13/07, Rilawich Ango [EMAIL PROTECTED] wrote:
Hi all,
In default, we can use # to transfer the call. I want to know how I
can know the user presse # to transfer the call in dial plan.
ango
Set TRANSFER_CONTEXT or GOTO_ON_BLINDXFER variable (depending on *
version) before Dial(). I
Hi all,
In default, we can use # to transfer the call. I want to know how I
can know the user presse # to transfer the call in dial plan.
ango
___
Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/
--Bandwidth and
check to see if you have dtmf=rfc2833 and canreinvite=no in sip.conf general
settings.
On 7/4/07, satish patel [EMAIL PROTECTED] wrote:
Dear all
I have install asterisk 1.2.x and it is working fine my
setup is like
[*]---[Mediant2k][Avaya]
Now i want to
Dear all
I have install asterisk 1.2.x and it is working fine my setup is
like
[*]---[Mediant2k][Avaya]
Now i want to transfer call in internal extension i have read more document on
www.voip-info.com but it is now so much clear so if u have any sample
dear all
I am new in asterisk and i have now setup asterik for 40 phone
now i want to configure call transfer between phone so how it is possible and
what configuration part in asterisk will perfomed for this task give me
suggestion for my solution
Regards
Satish Patel
On Monday 02 July 2007 01:45:44 pm satish patel wrote:
dear all
I am new in asterisk and i have now setup asterik for 40
phone now i want to configure call transfer between phone so how it is
possible and what configuration part in asterisk will perfomed for this
task give
satish patel wrote:
dear all
I am new in asterisk and i have now setup asterik for
40 phone now i want to configure call transfer between phone so how it
is possible and what configuration part in asterisk will perfomed for
this task give me suggestion for my solution
Dear all
i have read that document but dont understand about function i
have include featuremap in extension.conf
[mysip]
include = featuremap
and reload extention.conf i got this error
*CLI extensions reload
Jul 2 19:23:04 WARNING[16320]: pbx.c:6444
satish patel wrote:
Dear ALL
I want to transfer call from one phone 2 another
phone so this is asterisk feature or SIP Phone feature or endpoint
feature how can i transfer phone call from to another phone
Rgd
Satish patel
Check out this page:
Dear ALL
I want to transfer call from one phone 2 another phone so
this is asterisk feature or SIP Phone feature or endpoint feature how can i
transfer phone call from to another phone
Rgd
Satish patel
-
Looking for a deal? Find
Dear ALL
I have asterisk with sip and it is integrated with avaya
through mediant
[*]-[mediant 2000]-E1--[Avaya]
Now i want to call transfer feature in asterisk means transfer call from one
phone 2 another phone how could it possible with asterisk
Regrads
Hi,
I want to transfer the call to a conferencing
room while dialing.
I tried to do that using manager API(Redirect),
but it did't work.
Regards,
Jason.
Don't pick lemons.
See all the new 2007 cars at
Hello All.
I am having some trouble with call transfers when asterisk is the 2nd
party called and I hope to benefit from your experience.
I want to use asterisk for call park/pickup and have configured openser
to relay calls made to ruri 700-720 to asterisk running on
localhost:5069
Hi Guys,
I'm implementing my Asterisk step by step, so far the communications between
softphones, hardphones with Gateways, voice mail, are working fine. Rightnow
I would like to enable Call tranfer (like Traditional PBX) in Blind Transfer
and AttendXFER, I'm reading features.conf in accordance
Your dial string must have either the t or T option set.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of omar
parihuana
Sent: Friday, December 01, 2006 9:10 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users
, December 01, 2006 9:10 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] CALL TRANSFER
Hi Guys,
I'm implementing my Asterisk step by step, so far the communications
between softphones, hardphones with Gateways, voice mail, are working fine.
Rightnow I would
] On Behalf Of omar
parihuana
Sent: Friday, December 01, 2006 5:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] CALL TRANSFER
Thanks!!!
I forget Tt option! (too basis!!)
On 12/1/06, Damon Estep [EMAIL PROTECTED] wrote:
Your dial string must
Can anyone help with the following problem please?
1) On a receptionist's phone (Snom 360 latest firmware), a call is answered.
2) While on this call a second call comes to the phone but she does not answer it.
3) The receptionist makes an attended transfer placing the first caller on hold
My guess is I stumped everyone ;)
Anyway, I rolled back asterisk to 1.2.9.1 (same for libpri and zaptel
back one release) and transfers were working again. Now I'm still quite
new to asterisks, I know enough to hold my own, but not enough to know
the full inter workings of it. But here is my
Hey everyone,
Currently we are running IP601's with 1.6.6 SIP firmware on Asterisk
1.2.10. It has been reported to me when doing an attended transfer the
audio drops out. I ran a few different tests and here is what I noticed.
1. Blind transfers work with no problem.
2. Attended transfers
Does anybody knows how to transfer calls from Sipura SPA 1001 configured
as asterisk internal ?
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Hi !
I am trying to transfer calls between internal SIP softclients, but it does
not work. Every time I press a key on the softclient, the CLI shows the
following output:
Attempting native bridge of SIP/456-9ee0 and SIP/173-f586
This is my extensions.conf:
[macro-voicemail]
exten =
Hi,
Anyone has experience in using Call Transfer Disconnect (CT-5) over a
PRI with Asterisk ?
Call Transfer Disconnect allows you to transfer a call to a third
party and disconnect yourself from the communication and also freeing
your PRI channels.
Here is a document that explains how
Hi!
Is anyone managed to transfer an alredy bridged call, to a cell phone?
Some days ago, someone told me to look for the solution in features.conf,
but I still haven't found it. I tryied to use de default blindxfer, but
it only
accept internal extensions.
Thanks in advance,
Giuseppe
Hi!
I tried this in features.conf
testfeature = *9,callee,Dial,CAPI/ISDN4/my_phone_number/b,60,T
and it works... but... I would be able to transfer a call to any phone
number,
so I tried to use this line:
testfeature = _*9.,callee,Dial,CAPI/ISDN4/${EXTEN:2}/b,60,T
but... Asterisk crash! (it
Hi!
Is it possible to transfer a call to an external phone instead of
transferring the call to internal phone?
(I'm sorry for my bad english, I hope you understand)
When, during a call, I digit #123, the call is transferred to internal
extension 123,
but if I digit #external_phone_number, it
Hi!
Is it possible to transfer a call to an external
phone instead of
transferring the call to internal phone?
(I'm sorry for my bad english, I hope you
understand)
When, during a call, I digit #123, the call is
transferred to internal
extension 123,
but if I digit
Yes, as long as the context that the phone transfering has an exten
declared for that number.
On 4/3/06, Giuseppe [EMAIL PROTECTED] wrote:
Hi!
Is it possible to transfer a call to an external phone instead of
transferring the call to internal phone?
(I'm sorry for my bad english, I hope you
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of C F
Sent: Monday, April 03, 2006 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call transfer to external phone number
Yes, as long as the context
] On Behalf Of C F
Sent: Monday, April 03, 2006 3:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] call transfer to external phone number
Yes, as long as the context that the phone transfering has an exten
declared for that number.
Does
Discussion
Subject: Re: [Asterisk-Users] call transfer to external phone number
Yes, as long as the context that the phone transfering has an exten
declared for that number.
Does Asterisk make any distinction between an internal number and an
external number? I'm inclined to think it might
Hi,
I'm trying to call an extension and then transfer the call
to another extension, but something strange happens.
This is the extension:
exten = _9.,1,Dial(CAPI/ISDN4/${EXTEN:1}/b,tT)
When I dial any number starting with 9, I always
get CALL FAILED, but the called party still receive
the call
Hi,
I'm trying to call an extension and then transfer the call
to another extension, but something strange happens.
This is the extension:
exten = _9.,1,Dial(CAPI/ISDN4/${EXTEN:1}/b,tT)
When I dial any number starting with 9, I always
get CALL FAILED, but the called party still receive
the call
Hi All, in regards to my previous queries about call transfers not working from
inside, several days of searching turned up this posting:
I got this to work by editing the line
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM})
to say
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM},,Tt)
in
I have a SIP user, 2944093 that dialled 3254102. I'm trying to transfer the
call from 3254102 to 3254104. When I try and transfer the call, I get the
following on the Asterisk console.
Mar 3 15:14:18 NOTICE[23124]: chan_sip.c:6731 get_refer_info: Supervised
transfer requested, but unable to
I have a SIP user, 2944093 that dialled 3254102. I'm trying to transfer the
call from 3254102 to 3254104. When I try and transfer the call, I get the
following on the Asterisk console.
Mar 3 15:14:18 NOTICE[23124]: chan_sip.c:6731 get_refer_info: Supervised
transfer requested, but unable to
Title: Call Transfer
Can anyone point me in the right direction. My users (all using Sipura SPA-841 phones) need the ability to transfer a call to another number. How can I setup a dial plan to do this?
David A. Morrow
Technical Systems Lead
Autodata Solutions Company
[EMAIL PROTECTED]
I would use asterisk's built in blind or attended transfer features.
This way the system is based around dtmf and the users aren't tied to a
specific model of phone to accomodate future upgrades.
In order to do this I would recommend editing features.conf so blindxfer
= ** instead of *. A
I'm not sure how this is suppose to work. But I want to be able to call
people from a SIP phone and transfer them into a conference room. If I
call another extension that is a SIP phone I can hit # and then enter
the conference room number. If I call from the PSTN to the SIP extension
phone I
I got this to work by editing the line
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM})
to say
exten = s,14,Dial(${OUT_${ARG1}}/${OUTNUM},,Tt)
in extensions.conf
Do you know of anyway to set it up through AMP, so it works with all calls?
Michael Sampson
Information Systems Manager
Customer Contact
http://www.voip-info.org/wiki/view/Asterisk+authenticate+using+voicemail+passwords
Cheers
2005/12/1, Joe Pukepail [EMAIL PROTECTED]:
Look into the findme feature, there is a patch on the bug tracker to add
this feature. I believe that someone shows how to do it in the dial plan.
I plan on
When I arrived a call, I would the call transfer in to another telephone number, but Asterisk show error: Executing GotoIfTime("Zap/4-1", "08:30-12:30|mon-fri|*|*?4") in new stack -- Executing GotoIfTime("Zap/4-1", "15:30-18:30|mon-fri|*|*?4") in new stack -- Executing Goto("Zap/4-1", "6") in new
Hi,
I'm trying to have an extension ring my SIP phone then try my cell
phone. I can transfer the call fine to the cell but I want it to ask
for a pin , voicemail pin, before transferring the call.
This is so if my cell's voicemail answers , the call doesn't transfer
to it.
Any ideas?
Look into the findme feature, there is a patch on the bug tracker to add this feature. I believe that someone shows how to do it in the dial plan. I plan on implementing this, but haven't gotten around to it yet.
On 11/30/05, Benjamin Lenard [EMAIL PROTECTED] wrote:
Hi,I'm trying to have an
Hi,
Does anyone have a sample config for phones (like the Zyxel P2000wv2) that
cannot handle more than one line. I have tried using # followed by the
extension and nothing happens??? I have parking setup but for some reason we
cannot retrieve the parked call. I call the user who the call is
AFAIK there were some known issues preventing call transfer from H323
terminals, at least with Innovaphone ones.
Yours
l.
On Fri, 18 Nov 2005 17:45:39 +0100, Santosh Rao
[EMAIL PROTECTED] wrote:
Hello list,
We have asterisk v1.2.0 CVS head and ooh323 in place. calls
can be
Hello list,
We have asterisk v1.2.0 CVS head and ooh323 in place. calls can be
made and recieved to and from extensions.
How to implement call transfer and call pickup. when using asterisk 1.0.x
dtmf=inband registers and sends dtmf but with asterisk 1.2 and ooh323 it does
not.. is
I'm using IAX2 with VP-320I hardphones for remote users. Everything seems to
be working fine except for call transfer. Is this an issue with the IAX2
itself or the phone? If I flash the same phone with SIP, the problem
disappears.
Regards,
Shaun Singh, Manager
Travelwave
1655 Dufferin Street,
Hi,
Thanks for the clarification. I had seen that the two options
existed, but the docs for the dial() command didn't state the
difference.
On Sun, Oct 30, 2005 at 08:23:32PM -0500, David Bandel wrote:
On 10/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,
Recently got
Hi All,
Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem. I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
this in extensions.conf to have it handle all outgoing calls beginning
with 1:
On 10/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,
Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem. I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
this in
David Bandel wrote:
On 10/30/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hi All,
Recently got call-transfer somewhat working on my asterisk-1.0.9
install, and came across an interesting problem. I have an account on a
VOIP Provider (voipbuster using iax to be exact) and use a line like
hello list,
in my asterisk i have blind transfer and attendent transfer.
when call Z which is a public call through Capi(BRI) is received by user A he can see the Caller ID of Z and
if user A blind transfer the call to user B, user B can see the caller ID of user Z but
when user A attendent
Hello,
I have my [EMAIL PROTECTED] working beautifully for basic call function. So now I
am testing extended functions for my office users and am hitting a wall.
I simply need to be able to put a call on hold and forward it to any
another internal extension. I have an Eezee AT-320 IAX2 phone
try # and then dial the extension.On 10/20/05, Rhonda Herron [EMAIL PROTECTED] wrote:
Hello,I have my [EMAIL PROTECTED] working beautifully for basic call function. So now Iam testing extended functions for my office users and am hitting a wall.I simply need to be able to put a call on hold and
I have the phone specific directions to transfer calls, but I tried your
suggestion. No go. I have 3 of the Eezee phones and call transfer
doesn't work on any of them, so I really don't think it is hardware
related. I think the problem may be with my feature.conf which had no
reference to
maybe its not setting the DTMF tones properly. What do you have
setup for the phone and extensions. Usually its rfc2833 but could
be inband.On 10/20/05, Rhonda Herron [EMAIL PROTECTED] wrote:
I have the phone specific directions to transfer calls, but I tried yoursuggestion. No go. I have 3 of the
It is set to rfc2833.
Tom Vile wrote:
maybe its not setting the DTMF tones properly. What do you have setup
for the phone and extensions. Usually its rfc2833 but could be inband.
On 10/20/05, *Rhonda Herron* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I have the phone
I'm not sure the txfer functionality is in the 1.0.X branch. I'm pretty sure you will need HEAD or the 1.2 betas.On 10/20/05, Rhonda Herron
[EMAIL PROTECTED] wrote:It is set to rfc2833.Tom Vile wrote:
maybe its not setting the DTMF tones properly.What do you have setup for the phone and
Blind transfer should work fine #. Can you dial into Voicemail and enter your password succesfully?On 10/20/05, BJ Weschke
[EMAIL PROTECTED] wrote:I'm not sure the txfer functionality is in the 1.0.X
branch. I'm pretty sure you will need HEAD or the 1.2 betas.On 10/20/05, Rhonda Herron
[EMAIL
Yes, I can dial *97 for VM and check messages. When I select # during a
call it does nothing though. I tried inband for DTMF but that didnt
work. Am going to run debug mode ( first I have to figure out how :) )
and I will let you know what I find out.
Thanks so far,
R
Tom Vile wrote:
Blind
He is what happens from the time the extension is selected from the time
the digital receptionist answers until I hangup. I watched the logs as
I was pushing all sorts of transfer button possibilities and nothing. It
just stayed at 'ooh, voice format changed to 4' Which, while humorous
tells
Hi,
I try to set up attended transfer in my Asterisk Box . My
features.conf look like this:
[general]
parkext = 100
parkpos = 1-5
context = parkedcalls
parkingtime = 100
transferdigittimeout = 3l
courtesytone = beep
xfersound = beep
xferfailsound = invalid
featuredigittimeout = 500
;adsipark =
Are you using 1.0.x? DTMF Attended Transfer is not supported in 1.0.x.
Unless you have a brain dead phone, you should be able to use SIP
attended transfer in 1.0.x. (that would be the transfer key on the phone)
Andrew Nowrot wrote:
Hi,
I try to set up attended transfer in my Asterisk Box
Hi,
Thank for the Email
I'm using 1.0.9 so probably I'm will not have this feature. In which
version of Asterisk the DTMF Attended Transfer is supported, in 1.2
Beta?
Best wishes
Andrew
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Andrew Nowrot wrote:
Hi,
Thank for the Email
I'm using 1.0.9 so probably I'm will not have this feature. In which
version of Asterisk the DTMF Attended Transfer is supported, in 1.2
Beta?
CVS-HEAD and 1.2Beta1 and later.
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Hello,
how i can tranfer call to another user? Im using X-Lite, i have configured in
features.conf:
[featuremap]
blindxfer = #1
disconnect = *0
automon = *1
atxfer = *2
But when im dial *2 in conversation nothig happens.
What can br problem?
Im using asterisk CVS-HEAD from 02/09/05.
Hi,
I try to set up planet VIP-050 with asterisk. Everything works fine
instead of the call transfer. When I press # console says something
like this:
Oct 5 11:11:20 DEBUG[25104]: chan_sip.c: sip_rtp_read: Oooh,
format changed to 1024
Oct 5 11:11:20 WARNING[25104]: codec_ilbc.c:144
I'm having the same problem you had Frank, so I'm pleased you came up
with a fix. No luck for me yet!
Incoming outgoing calls work fine using X-Lite, I just cannot transfer.
It's the first time I've ventured in to features.conf so I'm likely
doing something silly. I'd be grateful if you could
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