A number of our clients has such issues. What we suggest for escalation is
to do a blind transfer to a second-level queue, so that the logging is
correct and even if second-line support cannot handle the call immediately,
you get the functionality and the logging.
Just my two euro cents,
l.
Hello,
I thought to post this here before my manager starts his own coding
project to give us a workaround. My situation I'm running into is as
follows:
1. A call comes into our Asterisk system, it's trunked from one office
to another via IAX.
2. Call enters a queue and is picked up by one of
On 091006 1249, Darrin Henshaw wrote:
1. A call comes into our Asterisk system, it's trunked from one office
to another via IAX.
2. Call enters a queue and is picked up by one of the agents.
3. That agent has to transfer the call, could be for a number of
reasons the client wanted someone in
Hi,
I am trying to understand why some of my call transfers fail.
My scenario is as follows:
Legacy PBX1 ---PRI (EuroISDN) Zaptel--- Asterisk PBX2
Step1: PBX1 extension 101 calls PBX2 extension 102
Step2: PBX2 extension 102 answers the call and transfers it to PBX1 extension
103
Step3:
On Mon, Mar 16, 2009 at 8:49 AM, Vieri rentor...@yahoo.com wrote:
Hi,
I am trying to understand why some of my call transfers fail.
My scenario is as follows:
Legacy PBX1 ---PRI (EuroISDN) Zaptel--- Asterisk PBX2
Step1: PBX1 extension 101 calls PBX2 extension 102
Step2: PBX2 extension
Mark Hamilton wrote:
a) How can I make it so #2 doesn't have to be exceptionally fast, and maybe
get a second of delay in there permitted?
;featuredigittimeout = 500 ; Max time (ms) between digits for
; feature activation (default is 500 ms)
--
Ben
call-limit=10Please helpThanks!
Original Message
Subject: Re: [asterisk-users] Transfers on AgentLogin()
From: "James Sneeringer" [EMAIL PROTECTED]
Date: Fri, September 05, 2008 10:57 pm
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
asterisk-users@list
]
type=friend
qualify=yes
nat=yes
host=dynamic
dtmfmode=auto
context=manila
canreinvite=no
callerid=Agent 1013
call-limit=10
Please help
Thanks!
Original Message
Subject: Re: [asterisk-users] Transfers on AgentLogin()
From: James Sneeringer [EMAIL PROTECTED]
Date
, 2008 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers on AgentLogin()
What do you get when you type show features?
On 9/6/08, Mark Hamilton [EMAIL PROTECTED] wrote:
Hi James,
Thank you very much for a detailed reply. (Matt, sorry about
] On Behalf Of Mark Hamilton
Sent: August 31, 2008 4:15 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Transfers on AgentLogin()
I've tried the regular, xfer button on xlite, dial 100 (to transfer to the
queue), and hit go back to line 1 and hit xfer again
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers on AgentLogin()
What did you try and how did it fail? Are you using the t option in queue?
___
-- Bandwidth and Colocation Provided by http://www.api
-Commercial Discussion
Subject: Re: [asterisk-users] Transfers on AgentLogin()
What did you try and how did it fail? Are you using the t option in queue?
On 8/30/08, Mark Hamilton [EMAIL PROTECTED] wrote:
So, no answers or is this thread going to remain unanswered too?
From: [EMAIL PROTECTED
6:15 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Transfers on AgentLogin()
Oh, by the way, the agent who will be doing the assisted transfer will be
using eyebeam.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark
So, no answers or is this thread going to remain unanswered too?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: August 28, 2008 6:15 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Transfers on AgentLogin
Hi,
I have the same question as:
http://lists.digium.com/pipermail/asterisk-dev/2003-November/002320.html
..which like all important things was never answered.
How do I do an assisted transfer on AgentLogin()? I don't use zaptel, it's
just pure SIP/VoIP.
Help please.
Thanks,
Oh, by the way, the agent who will be doing the assisted transfer will be
using eyebeam.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Hamilton
Sent: August 28, 2008 5:54 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Transfers
Hi all,
when enabling blind and attended transfers in features.conf, these only seem
to work when I enable voicemail for a particular user. How can this be? Can I
have transferrring without voicemail?
Using Asterisk 1.4 by the way.
Thank you!
Bart
Bart Coninckx wrote:
Hi all,
when enabling blind and attended transfers in features.conf, these only seem
to work when I enable voicemail for a particular user. How can this be? Can I
have transferrring without voicemail?
Using Asterisk 1.4 by the way.
Thank you!
Bart
I think
Hi all,
when enabling blind and attended transfers in features.conf, these only seem
to work when I enable voicemail for a particular user. How can this be? Can
I
have transferrring without voicemail?
Using Asterisk 1.4 by the way.
Thank you!
Bart
I think some clarification
Hello all,
I have an asterisk PBX working perfectly, and the transfers between
extensions, works ok. The problem, when I receive a call from the line
connected to the TE12Xp, and I try to transfer it, the calls hangs up.
I have other analog lines and I can tranfer all the without problems.
I've
More info about the problem.
This occurs, when I try to transfer using the *2 funcionality into aterisk
Thanks
2008/6/16 voip crazy [EMAIL PROTECTED]:
Hello all,
I have an asterisk PBX working perfectly, and the transfers between
extensions, works ok. The problem, when I receive a call
On Mon, Jun 16, 2008 at 6:39 AM, voip crazy [EMAIL PROTECTED] wrote:
Hello all,
I have an asterisk PBX working perfectly, and the transfers between
extensions, works ok. The problem, when I receive a call from the line
connected to the TE12Xp, and I try to transfer it, the calls hangs up.
I
Hi everybody,
A question, how do I follow a call that is transferred? is the any event
or something in the CDR that would let me find all the call sequence?
Thanks
Rodrigo
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users
Dawson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, July 26, 2006 4:32 PM
Subject: Re: [asterisk-users] Transfers - No ringback or moh
I get round this bug by replacing:
exten = X,1,Dial(sip/blah)
with:
exten = X,1
I get round this bug by replacing:
exten = X,1,Dial(sip/blah)
with:
exten = X,1,Answer
exten = X,n,Dial(sip/blah)
It means the call is in an answered state before it starts ringing but
it doesn't seem to cause any major problems.
Mike
Martin Schrott - Thinking-Systems wrote:
Hi all,
I
Hi all,
I cannot exactly reproduce your problems, but I can tell you, what problem
we have on this topic:
a calles b.
b takes the call and can speak to a.
b sets up a attendend transfer (via the softkey configured in asterisk) to
c and hears ringing.
a hears music on hold.
b hears ringing
if c
Hi all,
Here is the situation:
A call comes in to an Alcatel PBX and it sends it to an E1 on * , this
* either sends the call to a VoIP extension or needs to forward it to an
extension back on the Alcatel, but WITHOUT using another slot of the
E1 (no tromboning or hairpinning).
I've
On Jul 25, 2006, at 12:53 PM, Manrique Feoli wrote:
Hi all,
Here is the situation:
A call comes in to an Alcatel PBX and it sends it to an E1 on * ,
this * either sends the call to a VoIP extension or needs to forward
it to an extension back on the Alcatel, but WITHOUT using another
slot
Hi Matt, thanks for your answer,
I guess it is still as you said a while back that you did it using 5ESS
Can you share how you did in 5ESS? (a sample of the extensions.conf )
and what kind of switch you were connected to?
I'm not sure if the Alcatel 4400 and the Nortel Meridian 11
I don't know
why, but when doing transfers between Polycom phones, once the transferring
party hits transfer a second time, to be removed from the call, User A no longer
hears music on hold, or a ring back.
Scenario.
1. User A dials User B.
2. User A and User B are
connected.
3. User B
Douglas Garstang wrote:
1. User A dials User B.
2. User A and User B are connected.
3. User B hits the transfer soft key. User A gets music on hold.
4. User B dials user C. User C's phone rings, and user A continues to
hear music on hold.
5. When User B presses the transfer soft key again to
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, July 24, 2006 9:34 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers - No ringback or moh
Douglas Garstang wrote:
1. User A dials User B.
2. User
Douglas Garstang wrote:
-Original Message-
And that's normal, for user A to just hear dead air?
I have a Polycom IP501 sitting on my desk (Test phone):
I call it with my Avaya phone
pick up the ringing extension
press transfer button (I hear hold music on the
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, July 24, 2006 11:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers - No ringback or moh
Douglas Garstang wrote:
-Original Message
Douglas Garstang wrote:
I don't know... all I know is that when user C starts to ring, and user B has
dropped from the call, the music on hold stops for user A, until user C
answers. I would have expected User A to hear ringing at this point.
Then they need to do a blind transfer.
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, July 24, 2006 12:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers - No ringback or moh
Douglas Garstang wrote:
I don't know... all I know
Douglas Garstang wrote:
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, July 24, 2006 12:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers - No ringback or moh
Doug,
The transfer soft button can
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, July 24, 2006 11:48 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers - No ringback or moh
Douglas Garstang wrote:
-Original Message
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, July 24, 2006 12:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers - No ringback or moh
Douglas Garstang wrote:
-Original Message-
From
Douglas Garstang wrote:
passes at all?
I don't think this is the same scenario. When you transfer to an Asterisk
extension, ie voicemail, your not going to get a period of ring back as
Asterisk will answer the call immediately.
In this example, I'm dialing to another extension on
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, July 24, 2006 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers - No ringback or moh
Douglas Garstang wrote:
passes at all?
I don't
Douglas Garstang wrote:
People (Users that is) seem to think, pressing less is better.
Doug, as it turns out, the transfer button on the polycom and the transfer soft button, both behave in exactly the same way.
What firmware?
I'm running Bootrom 3.1.3, sip.ld 1.5.2
Doug
--
Ben
Douglas Garstang wrote:
-Original Message-
When the extension on your desk is ringing, after you have pressed transfer key a second time(soft or hard key), does the original caller still hear music on hold, or ringback or nothing?
Following your example, pressing transfer
-Original Message-
From: Doug Lytle [mailto:[EMAIL PROTECTED]
Sent: Monday, July 24, 2006 2:15 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Transfers - No ringback or moh
Douglas Garstang wrote:
-Original Message
On 7/24/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Douglas Garstang wrote:
-Original Message-
When the extension on your desk is ringing, after you have
pressed transfer key a second time(soft or hard key), does
the original caller still hear music on hold, or ringback or
I am trying to get parked calls/transfers working on our polycom 501s + asterisk.The transer button on the polycom phone does not seem to transfer/park the call properly. I have to use the # - 700 to park the call.Furthermore the # - 700 only works on incomming calls. If I dial out then
Greetings fellow list members,
I am using ABE and I am attempting to impliment
transfers using "#". I am using both "T" and "t" as options in my Dial()
command. I am attempting to hit "#" then enter another extension from my
dialplan. I have tried this on both ends of the conversation and
Any idea what version of Asterisk ABE is based
on?
PaulH
- Original Message -
From:
Franklin Webb
To: asterisk-users@lists.digium.com
Sent: Friday, December 30, 2005 8:43
AM
Subject: [Asterisk-Users] transfers using
# in asterisk
Greetings fellow list
On Thu, 2005-12-29 at 16:43 -0500, Franklin Webb wrote:
Greetings fellow list members,
I am using ABE and I am attempting to impliment transfers using #.
I am using both T and t as options in my Dial() command. I am
attempting to hit # then enter another extension from my dialplan.
I
When transferring a call that came in on the Sipura and picked up by a
Polycom 501 (sip 1.52), then transferred to another polycom using the
transfer button on the polycom (havn't tried with the blind transfer
from the polycom phone), then as soon as the transfer is complete
(after pressing
Hi,
Were using our legacy PBX as a channel bank with
asterisk sitting between the pbx and our telco provider spliced by a TE410P.
If it were a straight analog FXS card then wed use a
hook flash to break into asterisk for transfers etc, does anybody know what the
equivalent is for the
I am currently running stable, CVS-v1-0-05/25/05-12:07:15, with Polycom
SIP phones, running 1.4.1.
Too many of our transfers using the Transfer end up with zombie channels
after a REFER. As such, I implemented # transfers, and all is well.
Sort of.
I have a reproducible issue. Take a call from
David,
Check out bug number 4375. Does this relate? 4375 is plaquing us like mad
and if I can find more people that get this too then it might move up in
priority.
-Matthew
David Gomillion wrote:
I am currently running stable, CVS-v1-0-05/25/05-12:07:15, with
Polycom SIP phones, running
Hi
We have a fairly simple Asterisk setup: E1 card, 8 FXO lines connected
to old PBX, and some SIP phones, used by a callcenter with queues.
Almost all calls are incoming (through E1 line), answered by some
callcenter operator (using SIP phones, call assigned by queue app),
and in some cases, are
Hi
We have a fairly simple Asterisk setup: E1 card, 8 FXO lines connected
to old PBX, and some SIP phones, used by a callcenter with queues.
Almost all calls are incoming (through E1 line), answered by some
callcenter operator (using SIP phones, call assigned by queue app),
and in some cases, are
Hi,
I'm using zaptel with three way calling and call transfers with a hookflash.
If I try transfering a call to an extension that is engaged I get an
engaged tone. This is fine, but how do I get back to the caller?
If I hookflash again I seem to put on a three-way call and the caller
can hear
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jim Kou
Sent: 21 January 2005 02:02
To: Asterisk List; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users] # Transfers.
No, it's doesn't work.
Asterisk List on 2005/1/21 01:48 wrote
@lists.digium.com
Subject: Re: [Asterisk-Users] # Transfers.
I justed edited the Wiki Asterisk config file features.conf for this
attended transfer features. Please check Wiki again for details.
Best regards,
--JJL44
On Thu, 20 Jan 2005 09:04:40 -0800 (PST), Bruce Komito [EMAIL PROTECTED]
wrote
Am Mittwoch, 19. Januar 2005 19:18 schrieb Asterisk List:
The current CVS HEAD version already has ## transfer built-in. See
the included configs/features.conf.sample file. You can define your
own transfer key sequence. There is also an attended transfer
feature.
What is an attended
Attended transfer, also called supervised transfer, works like this:
While on conversation with another party, you dial ** the transfer
key sequence. Asterisk says Transfer then gives you a dial tone,
while put the other party on hold music. You dial the transferee
number and talk with the
Sorry if I missed the beginning of this thread, but I've never heard of
the ** transfer key sequence, nor have I found a way to make it work.
Would you mind, please explaining this further or pointing me to somewhere
where it's documented? (I checked Wiki and Google but no joy.)
Thanks
Bruce
features.conf
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Bruce Komito
Sent: Thursday, January 20, 2005 11:05 AM
To: Asterisk List
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] # Transfers.
Sorry if I
I justed edited the Wiki Asterisk config file features.conf for this
attended transfer features. Please check Wiki again for details.
Best regards,
--JJL44
On Thu, 20 Jan 2005 09:04:40 -0800 (PST), Bruce Komito [EMAIL PROTECTED]
wrote:
Sorry if I missed the beginning of this thread, but
Does this work with app_queue/chan_agent?
Cheers,
Ben
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
List
Sent: 20 January 2005 17:28
To: Bruce Komito
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] # Transfers.
I justed
]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk
List
Sent: 20 January 2005 17:28
To: Bruce Komito
Cc: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] # Transfers.
I justed edited the Wiki Asterisk config file features.conf for this
attended transfer features. Please check Wiki
No, it's doesn't work.
Asterisk List on 2005/1/21 01:48 wrote:
I have no idea if atxfer works with app_queue/chan_agent. Can anyone try it?
Best regards,
--JJL44
On Thu, 20 Jan 2005 17:38:25 -, Ben Merrills [EMAIL PROTECTED] wrote:
Does this work with app_queue/chan_agent?
Cheers,
Ben
--
The current CVS HEAD version already has ## transfer built-in. See
the included configs/features.conf.sample file. You can define your
own transfer key sequence. There is also an attended transfer
feature.
features.conf file:
[featuremap]
blindxfer = ##
atxfer = **
This worked very well for
So I have read and read and read... google is my friend and the wiki is
by brother...
However, I am still unclear on what the preferred method of using the
pound sign is.
If the Pound sign is set aside for Transfer.. Then when I make an
outbound call to my bank I can not Enter my PIN followed by
-users@lists.digium.com
Sent: Wednesday, January 19, 2005 1:32 PM
Subject: [Asterisk-Users] # Transfers.
So I have read and read and read... google is my friend and the wiki is
by brother...
However, I am still unclear on what the preferred method of using the
pound sign is.
If the Pound sign
Ok, I've seen discussion before on doing transfers
(attended and unattended), there seems to be much confusion over
it.
As things sit, I've been trying (unsuccessfully) to
do transfers with a zap channel (analog phone attached to TDM400). I have
no idea what I'm missing. My current
- Original Message -
From:
Paul
Fielding
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, January 17, 2005 6:12
PM
Subject: [Asterisk-Users] transfers with
zap channel
Ok, I've seen discussion before on doing
transfers (attended
Subject: Re: [Asterisk-Users] transfers
with zap channel
How long between getting parked is the orginal
call dropping?
Depending on your dialplan, yes dialing 700x will
almost immediately send the call to call parking. (IMHO, poor extension
planning can also cause
Have you looked at features.conf?
Lyle
- Original Message -
From:
Paul
Fielding
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, January 17, 2005 8:53
PM
Subject: Re: [Asterisk-Users] transfers
with zap channel
The outside line
List -
Non-Commercial Discussion
Sent: Monday, January 17, 2005 8:20
PM
Subject: Re: [Asterisk-Users] transfers
with zap channel
Have you looked at features.conf?
Lyle
- Original Message -
From:
Paul
Fielding
To: Asterisk Users Mailing List
Can someone please help with this. After an outside caller has been parked,
they inherit our abilitites to transfer. I have played with all the
different combinations of T and t, but nothing seems to work. I found a way
to get my Sipura to work with a flash transfer. So right now I am stuck. Is
I've got an interesting scenario where transfers while getting an invite
seem to break.
Here is the scenario: You have a receptionist who has a 6 line phone (in
this case, a polycom ip600 - also tested with a Cisco 7960) the
receptionist has all six lines available for use (in the case of the
On Thu, 2003-07-10 at 10:26, Kim C. Callis wrote:
I noticed that there is a soft button for transfer when you initiate a
call. I pressed it, and it actually put the call on hold, although I
was able to call another extension. Is that soft button functional?
And if so, how do you make use of
We support both blind and supervised transfers on the Cisco.
Mark
On Thu, 10 Jul 2003, Kim C. Callis wrote:
I noticed that there is a soft button for transfer when you initiate a
call. I pressed it, and it actually put the call on hold, although I was
able to call another extension. Is that
On Thursday 10 July 2003 03:24 pm, Mark Spencer wrote:
We support both blind and supervised transfers on the Cisco.
Mark
On Thu, 10 Jul 2003, Kim C. Callis wrote:
I noticed that there is a soft button for transfer when you initiate a
call. I pressed it, and it actually put the call on
79 matches
Mail list logo
