[asterisk-users] Attended Transfer using AMI on PJSIP

[Dan Cropp]

I need to make attended transfer work via an AMI request.

Based on data from a Cisco trace from another system which successfully does an 
attended transfer, the Refer-To header requires the following format 


Using the PJSIP_HEADER support, I am abte to retrieve everything I need with 
the exception of the to-tag.

1. I retrieve the Via address from the destination channel.
2. I retrieve the Call-ID for the destination call using the PJSIP_HEADER 
support.
3. I retrieve the From header's tag field from the destination channel.

Is there a way to retrieve the tag from the To header?  When I use the 
PJSIP_HEADER support to read the To header, it does not include the tag field.


Is there a better way to retrieve this information for the Refer-To Replaces 
portion?

Have a great day!
Dan

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[asterisk-users] Attended transfer problem

[jg]

I have a setup where there are occasional problems with attended transfers. I have already 
checked the devices as well as the relevant DTMF modes (SIP INFO and rfc2833). I could not find 
any problems here.


The setup is a follows:

The front desk (F) accepts calls from customers (C). In some cases F needs to transfer C to a 
specific department (D). If D cannot handle the problem, D tries to transfer to a specialist S. 
The problem sometimes occurs when D tries an attended transfer for C to S. The description I 
have so far is that Asterisk sometimes does not seem to accept DTMF signals.


Is it conceivable that that this is a re-INVITE/directmedia problem?

Since I have only limited access to the system, I'd like to rule out some causes. I am also not 
sure if the transfer from F to D is handled via Asterisk using DTMF signalling, or whether F 
uses the SIP phone's capabilities for the transfer, so that Asterisk might already not know 
anything about D. Except for C, everybody is inside the same subnet.


jg

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[asterisk-users] attended transfer caller hears ringing after transfer done

[Mitch Claborn]

Asterisk 1.8.10.1~dfsg-1ubuntu1

A calls B.
B answers, intiaites an attendeded transfer to C.
C answers.
B hangs up.
A now hears ringing forever, until the call is terminated.

blind transfer does not have this problem.

What am I missing?

features.conf
[featuremap]
blindxfer = #1
atxfer = #2

extensions.conf
[macro-voicemail]
exten =s,1,Dial(${ARG1},10,t|m=default)
  same =n,GotoIf($[${DIALSTATUS} = BUSY]?busy:unavail)
  same =n(unavail),NoOp()
  same =n,VoiceMail(${ARG2}@default,u)
  same =n,Hangup()
  same =n(busy),NoOp()
  same =n,VoiceMail(${ARG2}@default,b)
  same =n,Hangup()

[LocalSets]

include =InternalSets



; xfer to voicemail using *extension
exten = _*2XX,1,NoOp();
  same =n,Verbose(xfer to ${EXTEN:1})
  same =n,Voicemail(${EXTEN:1})

include =external
include =emergency-services
include =services
include =QueueMemberFunctions

[InternalSets]
exten =295,1,Macro(voicemail,${MITEL1},295)
exten =mitel1,1,Macro(voicemail,${MITEL1},295)

exten =296,1,Macro(voicemail,${MLCM800},296)
exten =mlcm800,1,Macro(voicemail,${MLCM800},296)

exten =298,1,Macro(voicemail,${MLCX500},298)
exten =mlcx500,1,Macro(voicemail,${MLCX500},298)

exten =299,1,Macro(voicemail,${MLCX450},299)
exten =mlcx450,1,Macro(voicemail,${MLCX450},299)


CLI output
== Using SIP RTP CoS mark 5
  -- Executing [295@LocalSets:1] Macro(SIP/mlcm800-010a, 
voicemail,SIP/mitel1,295) in new stack
  -- Executing [s@macro-voicemail:1] Dial(SIP/mlcm800-010a, 
SIP/mitel1,10,t|m=default) in new stack

== Using SIP RTP CoS mark 5
  -- Called SIP/mitel1
  -- Started music on hold, class 'default', on SIP/mlcm800-010a
  -- SIP/mitel1-010b is ringing
  -- SIP/mitel1-010b answered SIP/mlcm800-010a
  -- Stopped music on hold on SIP/mlcm800-010a
  -- Started music on hold, class 'default', on SIP/mlcm800-010a
  -- SIP/mitel1-010b Playing 'pbx-transfer.slin' (language 'en')
  -- Executing [299@LocalSets:1] Macro(Local/299@LocalSets-fbe5;2, 
voicemail,SIP/mlcx450,299) in new stack
  -- Executing [s@macro-voicemail:1] Dial(Local/299@LocalSets-fbe5;2, 
SIP/mlcx450,10,t|m=default) in new stack

== Using SIP RTP CoS mark 5
  -- Called SIP/mlcx450
  -- Started music on hold, class 'default', on Local/299@LocalSets-fbe5;2
  -- SIP/mlcx450-010c is ringing
  -- SIP/mlcx450-010c answered Local/299@LocalSets-fbe5;2
  -- Stopped music on hold on Local/299@LocalSets-fbe5;2
  -- Stopped music on hold on SIP/mlcm800-010a
  -- Local/299@LocalSets-fbe5;1 Playing 'beep.slin' (language 'en')
== Spawn extension (macro-voicemail, s, 1) exited non-zero on 
'Transfered/SIP/mlcm800-010aZOMBIE' in macro 'voicemail'
== Spawn extension (LocalSets, 295, 1) exited non-zero on 
'Transfered/SIP/mlcm800-010aZOMBIE'
== Spawn extension (macro-voicemail, s, 1) exited non-zero on 
'Local/299@LocalSets-fbe5;2' in macro 'voicemail'
== Spawn extension (LocalSets, 299, 1) exited non-zero on 
'Local/299@LocalSets-fbe5;2'




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Re: [asterisk-users] attended transfer with CEL

[Marek Cervenka]

https://wiki.asterisk.org/wiki/display/AST/CEL+Function
on this wiki is THIS IS NO LONGER TRUE REWRITE

is there some way to write userfield,accountcode to the cel?

Dne 5.6.2012 13:21, Marek Cervenka napsal(a):

hello,

is there someone who successfully get info about attended transfer 
from CEL?

if yes, can you post some hints/algorithm/...?

scenario
A - customer
B - secretary
C - consultant1
D - consultant2

A - B
B axfer C
C axfer D

i need to know
time B with C (consultation)
time A with C
time C with D (consultation)
time A with D
time A with everyone (full time -  from start to the end of call)





--
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Marek Cervenka
===


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Re: [asterisk-users] attended transfer with CEL

[Marek Cervenka]

Dne 20.6.2012 18:40, Marek Cervenka napsal(a):

https://wiki.asterisk.org/wiki/display/AST/CEL+Function
on this wiki is THIS IS NO LONGER TRUE REWRITE

is there some way to write userfield,accountcode to the cel?



solved. it's   set(CHANNEL(userfield)=something)

another question
i'm using patch from https://issues.asterisk.org/jira/browse/ASTERISK-18037
it works great

but there is problem(bug?) in second axfer

A - call - B - axfer(AtoC) - C - axfer(AtoD) D

in cel is
eventtype, cid_num, exten
HOLD_START, A, B
HOLD_STOP, A, B
BUT second axfer is
HOLD_START, B, C
HOLD_STOP, B, C

this is strange because on hold is A. is it a bug?

very big problem is that, i cant get info about A - D call (after second 
axfer). there is no info about bridged channel A after axfer



Dne 5.6.2012 13:21, Marek Cervenka napsal(a):

hello,

is there someone who successfully get info about attended transfer 
from CEL?

if yes, can you post some hints/algorithm/...?

scenario
A - customer
B - secretary
C - consultant1
D - consultant2

A - B
B axfer C
C axfer D

i need to know
time B with C (consultation)
time A with C
time C with D (consultation)
time A with D
time A with everyone (full time -  from start to the end of call)








--
---
Marek Cervenka
Centrum Vypocetni Techniky
jabber  - cerve...@slu.cz
CVT - http://cvt.fpf.slu.cz
FPF SLU OPAVA   - http://www.fpf.slu.cz
RHCE,RHCVA 100-175-678
===


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[asterisk-users] attended transfer with CEL

[Marek Cervenka]

hello,

is there someone who successfully get info about attended transfer from CEL?
if yes, can you post some hints/algorithm/...?

scenario
A - customer
B - secretary
C - consultant1
D - consultant2

A - B
B axfer C
C axfer D

i need to know
time B with C (consultation)
time A with C
time C with D (consultation)
time A with D
time A with everyone (full time -  from start to the end of call)


--
---
Marek Cervenka
===


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[asterisk-users] Attended Transfer does not release channels

[Wolfgang Pichler]

Hi all,

i have the following setup

PSTN - routing server (asterisk 1.6.2.11) - IAX - callcenter asterisk
1.6.2.9 - SIP - agent


Does work quit fine - then agent does have the abibility to transfer a call
to a third party - the agent can initiate the transfer over a web interface
- it does generate a asterisk manager atxfer request...

So agent does initiate transfer - call flow is

agent - SIP - callcenter asterisk - NEW call over IAX - routing server
- PSTN

Then agent hangs up - so that the original caller and the new call will get
connected - and - it is working

But - the call will not get released on the callcenter asterisk machine

So the callflow after the transfer is

Original call PSTN - routing server - callcenter asterisk - routing
server - PSTN

But it should be

Original call PTN - routing server - PSTN

I have transfer = yes and mediaonly both tested on my connection routing
server to asterisk callcenter - does not help

the iax peer beetween the both does have trunk=yes

I do not get any error message (unable to transfer or something like this)

I have done a full network dump of such a call - and i can see that asterisk
callcenter does not make any attempt to directly bridge the calls - no TXREQ
or something like that.



So - why does it not try to directly bridge the both channels ?

I am using a local channel in the middle on asterisk callcenter - with /n
option - could this be the problem ?

best regards,
Wolfgang
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Re: [asterisk-users] Attended Transfer does not release channels

[Olivier]

2010/9/17 Wolfgang Pichler wpich...@yosd.at

 Hi all,

 i have the following setup

 PSTN - routing server (asterisk 1.6.2.11) - IAX - callcenter asterisk
 1.6.2.9 - SIP - agent


 Does work quit fine - then agent does have the abibility to transfer a call
 to a third party - the agent can initiate the transfer over a web interface
 - it does generate a asterisk manager atxfer request...

 So agent does initiate transfer - call flow is

 agent - SIP - callcenter asterisk - NEW call over IAX - routing server
 - PSTN

 Then agent hangs up - so that the original caller and the new call will get
 connected - and - it is working

 But - the call will not get released on the callcenter asterisk machine

 So the callflow after the transfer is

 Original call PSTN - routing server - callcenter asterisk - routing
 server - PSTN

 But it should be

 Original call PTN - routing server - PSTN

 I have transfer = yes and mediaonly both tested on my connection routing
 server to asterisk callcenter - does not help

 the iax peer beetween the both does have trunk=yes

 I do not get any error message (unable to transfer or something like this)

 I have done a full network dump of such a call - and i can see that
 asterisk callcenter does not make any attempt to directly bridge the calls -
 no TXREQ or something like that.



 So - why does it not try to directly bridge the both channels ?


see http://issues.asterisk.org/view.php?id=17999 and related bugs


 I am using a local channel in the middle on asterisk callcenter - with /n
 option - could this be the problem ?

 best regards,
 Wolfgang

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Re: [asterisk-users] Attended Transfer does not release channels

[Wolfgang Pichler]

2010/9/17 Olivier oza_4...@yahoo.fr



 2010/9/17 Wolfgang Pichler wpich...@yosd.at

 Hi all,

 i have the following setup

 PSTN - routing server (asterisk 1.6.2.11) - IAX - callcenter asterisk
 1.6.2.9 - SIP - agent


 Does work quit fine - then agent does have the abibility to transfer a
 call to a third party - the agent can initiate the transfer over a web
 interface - it does generate a asterisk manager atxfer request...

 So agent does initiate transfer - call flow is

 agent - SIP - callcenter asterisk - NEW call over IAX - routing server
 - PSTN

 Then agent hangs up - so that the original caller and the new call will
 get connected - and - it is working

 But - the call will not get released on the callcenter asterisk machine

 So the callflow after the transfer is

 Original call PSTN - routing server - callcenter asterisk - routing
 server - PSTN

 But it should be

 Original call PTN - routing server - PSTN

 I have transfer = yes and mediaonly both tested on my connection routing
 server to asterisk callcenter - does not help

 the iax peer beetween the both does have trunk=yes

 I do not get any error message (unable to transfer or something like this)

 I have done a full network dump of such a call - and i can see that
 asterisk callcenter does not make any attempt to directly bridge the calls -
 no TXREQ or something like that.



 So - why does it not try to directly bridge the both channels ?


 see http://issues.asterisk.org/view.php?id=17999 and related bugs

I have taken a look at these bugs - but they don't seem to be related to my
problem - then transfer is working in my scenario - the problem is that the
call legs are not getting optimized out as it should be the case...

A calls B - B makes attended transfer to C - B talks to C - B hangs  up -
asterisk should optimize out the call leg A - B and B - C to only A-C if
it is possible



 I am using a local channel in the middle on asterisk callcenter - with /n
 option - could this be the problem ?

 best regards,
 Wolfgang

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[asterisk-users] Attended Transfer question

[Warren Selby]

I've been asked to implement the following transfer workflow in an asterisk
system, and I'm not seeing an easy way to do the bolded steps below (steps 4
and 5 for those with a text-only email client):

1 - Put the call on hold
2 - Call the extension for the staff member needed
3 - Give them a rundown of the  caller and situation
*4 - Bring the caller on with the staff member the call will be transferred
to
5 - The person transferring the call will recap the situation with the
caller and staff member the call will be transferred to*
6 - Transfer the call and drop off without the call being dropped

Now, the way things work now:

1 - Press the transfer button on the phone, putting the original call on
hold.
2 - Dial the staff member needed.
3 - Explain situation.
4 - Press transfer button on phone again, transferring the caller to the
staff member, removing yourself from the call entirely.

Any suggestions?

-- 
Thanks,
--Warren Selby
http://www.selbytech.com
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Re: [asterisk-users] Attended Transfer question

[Danny Nicholas]

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren Selby
Subject: [asterisk-users] Attended Transfer question

 

I've been asked to implement the following transfer workflow in an asterisk
system, and I'm not seeing an easy way to do the bolded steps below (steps 4
and 5 for those with a text-only email client):

You could create a dynamic meetme room for the 3 legs and drop out when
done.  Or do it with X static meetme rooms.  You could set up a context to
create the MM room and call the supervisor, connecting them to the room.

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Re: [asterisk-users] Attended Transfer question

[Philipp von Klitzing]

Hi!

 I've been asked to implement the following transfer workflow in an 
 asterisk system, and I'm not seeing an easy way to do the bolded steps
 below (steps 4 and 5 for those with a text-only email client):
 
 You could create a dynamic meetme room for the 3 legs and drop out when
 done. Or do it with X static meetme rooms. You could set up a context to
 create the MM room and call the supervisor, connecting them to the room.

Why not just do it with your phone?

* Call the staff member using your 2nd line appearance
* Create a phone-based 3-party conference
* Dissolve the conference
* Transfer the caller to the staff member

If that is not what you want: Look at MeetMe() and ChanSpy().

Philipp


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[asterisk-users] Attended Transfer using AMI

[Grant Murray]

I am looking for a way to have an agent execute an attended transfer
using the asterisk manager interface (AMI).

I have been trying to use the dual Redirect from svn trunk. The problem
with this function is that the ExtraChannel does not get redirected
properly afaict.

Now, I am looking for other solutions for the list, I will probably try
playing DTMFs on the agent channel to simulate the manual transfer next
unless anyone has some better ideas.

Thanks Grant 



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Re: [asterisk-users] Attended transfer and callerID updates forSiemens Openstage phones

[Loic Didelot]

Hi Kevin,
is this feature implemented in the Business Version of Asterisk?

Best reards,
Loïc.



On Thu, 2010-03-25 at 07:20 -0500, Kevin P. Fleming wrote:
 Loic Didelot wrote:
 
  I am testing the Openstage phones from Siemens but I can not find a
  solution on how to update the caller-id after a successful attended
  transfer. Of course, I mean an attended transfer by using the phones
  functionality, not something defined in asterisks features.conf.
 
 This is called 'Connected Party ID', and it isn't supported in any
 released version of Asterisk... but it is supported in SVN trunk and
 will be part of Asterisk 1.8.
 
 -- 
 Kevin P. Fleming
 Digium, Inc. | Director of Software Technologies
 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
 skype: kpfleming | jabber: kflem...@digium.com
 Check us out at www.digium.com  www.asterisk.org
 



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[asterisk-users] Attended transfer and callerID updates forSiemens Openstage phones

[Loic Didelot]

Hello,
I am testing the Openstage phones from Siemens but I can not find a
solution on how to update the caller-id after a successful attended
transfer. Of course, I mean an attended transfer by using the phones
functionality, not something defined in asterisks features.conf.

Any idea on how to achieve this, or any technical document from Siemens
on on how this is support to work would help.

Thanks in advance,
Loïc.

-- 
Loïc DIDELOT
MIXvoip S.a.
Tel: +352 20  20
Fax: +352 20  90
ldide...@mixvoip.com
http://www.mixvoip.com


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Re: [asterisk-users] Attended transfer and callerID updates forSiemens Openstage phones

[Kevin P. Fleming]

Loic Didelot wrote:

 I am testing the Openstage phones from Siemens but I can not find a
 solution on how to update the caller-id after a successful attended
 transfer. Of course, I mean an attended transfer by using the phones
 functionality, not something defined in asterisks features.conf.

This is called 'Connected Party ID', and it isn't supported in any
released version of Asterisk... but it is supported in SVN trunk and
will be part of Asterisk 1.8.

-- 
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
skype: kpfleming | jabber: kflem...@digium.com
Check us out at www.digium.com  www.asterisk.org

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Re: [asterisk-users] Attended transfer and callerID updates forSiemens Openstage phones

[Philipp von Klitzing]

Hi!

 I am testing the Openstage phones from Siemens but I can not find a
 solution on how to update the caller-id after a successful attended
 transfer. 

When I tested the OpenStage 60 recently I did not get that to work 
either, but this was with a medium aged Asterisk 1.4.17. Not sure where 
Asterisk 1.6.x stands with this, though.

If you are into German then here's a decently frequented forum:
http://www.ip-phone-forum.de/forumdisplay.php?f=620

Philipp


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[asterisk-users] Attended transfer broken in 1.6.0.25

[Theo Band]

I have the following problem with the 1.6.0.25 version of Asterisk:

1. A calls B
2. B picks up and talks to A
3. B does attended transfer to C
4. C picks up, but B still hears ringing
5. A and B are connected again (AT timeout exceeded on console)

This is exactly the same problem as mentioned in bug 16816
https://issues.asterisk.org/view.php?id=16816
This bug is solved but filed against 1.6.2 branch, so not the 1.6.0
branch. Searching further I found a couple of other related bugs, but
all are closed/duplicate or for another branch.

How can I tell whether this problem gets fixed for the 1.6.0 branch? And
if it gets fixed, how long in general will it take before it will be
merged into the 1.6.0 branch? My users rely on the attended transfer
feature, so I like to tell them when it will be solved.

Recently I started to use the yum repository distribution which is based
on the 1.6.0 branch. I see about once per month an update and I waited
for the 1.6.0.25 update last week. It didn't fix this problem...
I like using yum because of the ease of updating. I prefer not to
compile the code myself as I did in the past for the 1.4 branch. That's
why I try to understand something about the patch release. Should I
perhaps file a bug as well but now against this specific branch?

Theo

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Re: [asterisk-users] Attended transfer broken in 1.6.0.25

[Leif Madsen]

Theo Band wrote:
 I have the following problem with the 1.6.0.25 version of Asterisk:
 
 1. A calls B
 2. B picks up and talks to A
 3. B does attended transfer to C
 4. C picks up, but B still hears ringing
 5. A and B are connected again (AT timeout exceeded on console)
 
 This is exactly the same problem as mentioned in bug 16816
 https://issues.asterisk.org/view.php?id=16816
 This bug is solved but filed against 1.6.2 branch, so not the 1.6.0
 branch. Searching further I found a couple of other related bugs, but
 all are closed/duplicate or for another branch.

Have you looked at the notes that shows the merging of the fix to various 
branches, such as this one?

https://issues.asterisk.org/view.php?id=16816#118692

That note shows it merged to 1.4. It was then merged to trunk, 1.6.2, 1.6.1, 
and 
1.6.0 branches.

 How can I tell whether this problem gets fixed for the 1.6.0 branch? And
 if it gets fixed, how long in general will it take before it will be
 merged into the 1.6.0 branch? My users rely on the attended transfer
 feature, so I like to tell them when it will be solved.

Whenever an issue is closed, it is merged into all applicable branches at the 
same time. You can see the revision number and the branches it was merged into 
via the svnbot notes in the bug, which are usually the last notes of an issue.

 Recently I started to use the yum repository distribution which is based
 on the 1.6.0 branch. I see about once per month an update and I waited
 for the 1.6.0.25 update last week. It didn't fix this problem...
 I like using yum because of the ease of updating. I prefer not to
 compile the code myself as I did in the past for the 1.4 branch. That's
 why I try to understand something about the patch release. Should I
 perhaps file a bug as well but now against this specific branch?

There is no need to file a patch. You can check the ChangeLogs from the archive 
files (even if you don't install from source), or you can look at the 'svn log' 
on a branch or tag to see what fixes have been applied.

Leif Madsen.

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Re: [asterisk-users] Attended transfer broken in 1.6.0.25

[Theo Band]

Leif Madsen wrote:
 Have you looked at the notes that shows the merging of the fix to various 
 branches, such as this one?

 https://issues.asterisk.org/view.php?id=16816#118692

 That note shows it merged to 1.4. It was then merged to trunk, 1.6.2, 1.6.1, 
 and 
 1.6.0 branches.

   
No, I didn't realize the svn notes were also present here.
 How can I tell whether this problem gets fixed for the 1.6.0 branch? And
 if it gets fixed, how long in general will it take before it will be
 merged into the 1.6.0 branch? My users rely on the attended transfer
 feature, so I like to tell them when it will be solved.
 

 Whenever an issue is closed, it is merged into all applicable branches at the 
 same time. You can see the revision number and the branches it was merged 
 into 
 via the svnbot notes in the bug, which are usually the last notes of an issue.

   
 Recently I started to use the yum repository distribution which is based
 on the 1.6.0 branch. I see about once per month an update and I waited
 for the 1.6.0.25 update last week. It didn't fix this problem...
 I like using yum because of the ease of updating. I prefer not to
 compile the code myself as I did in the past for the 1.4 branch. That's
 why I try to understand something about the patch release. Should I
 perhaps file a bug as well but now against this specific branch?
 

 There is no need to file a patch. You can check the ChangeLogs from the 
 archive 
 files (even if you don't install from source), or you can look at the 'svn 
 log' 
 on a branch or tag to see what fixes have been applied.
   
Thanks for your explanations. I just have to wait for the next yum
release in a month from now and it will then probably be fixed.

Theo

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[asterisk-users] Attended transfer: transferring a call as soon as the destination starts ringing

[A. B.]

Hi all!

Ext A, B and C are SIP phones.

Ext A receives a call from Ext B. Ext A wants to transfer the call to Ext
C.  Ext A puts the first call on hold, dials Ext C, then simply hangs up as
soon as the call to Ext C starts *ringing*. In other words, Ext B wants to
be sure Ext C is ringing (i.e. it is not busy or unavailable) but doesn't
want to talk to him.

Unfortunately, as soon as Ext A hears Ext C is ringing and hangs up or hits
Transfer, the call is closed and a *new* call from Ext B to Ext C starts.
This way, Ext C sees an unanswered call from Ext A, which is an unexpected
behaviour.

I played with directmedia and directrtpsetup, but no success so far. Any
ideas, please?

Thanks in advance.

Alex
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Re: [asterisk-users] Attended Transfer with REFER

[Örn Arnarson]

Thanks a lot guys. Exactly what I needed.

Best regards,
Örn

On Tue, Jan 26, 2010 at 8:48 PM, Olle E. Johansson o...@edvina.net wrote:


 26 jan 2010 kl. 16.48 skrev Örn Arnarson:

  Hi guys,
 
  I am wondering (and have been unable to find out thus far) whether
 Asterisk sets some special channel variables or something when a call is
 transfered with the REFER method.
  Basically, I'm trying to figure out if it is possible to somehow get a
 transferred call back to the transferrer (as it is done with the built-in
 atxfer) after X seconds (or an unsuccessful attempt).
 
  Using a timeout in the Dial command is not suitable unless I am able to
 tell somehow that the call in question is being forwarded (which is of
 course not the case, as the Dial command is called befer the REFER is sent).
 
  Can anyone think of a way to get the call back to the transferrer after
 this timeout?
 
 THe transferred call is sent to a context set with the channel variable
 TRANSFER_CONTEXT before you call DIAL().

 In there, run DUMPCHAN to see which variables you have and then dial with a
 timeout. After the timeout, dial back.

 /O
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[asterisk-users] Attended Transfer with REFER

[Örn Arnarson]

Hi guys,

I am wondering (and have been unable to find out thus far) whether Asterisk
sets some special channel variables or something when a call is transfered
with the REFER method.
Basically, I'm trying to figure out if it is possible to somehow get a
transferred call back to the transferrer (as it is done with the built-in
atxfer) after X seconds (or an unsuccessful attempt).

Using a timeout in the Dial command is not suitable unless I am able to tell
somehow that the call in question is being forwarded (which is of course not
the case, as the Dial command is called befer the REFER is sent).

Can anyone think of a way to get the call back to the transferrer after this
timeout?

Best regards,
Örn Arnarson
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Re: [asterisk-users] Attended Transfer with REFER

[Magnus Benngård]

checkout ${BLINDTRANSFER}

On Tue, 26 Jan 2010 15:48:51 +, Örn Arnarson  wrote: Hi guys, 
 I am wondering (and have been unable to find out thus far) whether
Asterisk sets some special channel variables or something when a call is
transfered with the REFER method. Basically, I'm trying to figure out if it
is possible to somehow get a transferred call back to the transferrer (as
it is done with the built-in atxfer) after X seconds (or an unsuccessful
attempt). 
 Using a timeout in the Dial command is not suitable unless I am able to
tell somehow that the call in question is being forwarded (which is of
course not the case, as the Dial command is called befer the REFER is
sent). 
 Can anyone think of a way to get the call back to the transferrer after
this timeout? 
 Best regards, Örn Arnarson  

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Re: [asterisk-users] Attended Transfer with REFER

[Olle E. Johansson]

26 jan 2010 kl. 16.48 skrev Örn Arnarson:

 Hi guys,
 
 I am wondering (and have been unable to find out thus far) whether Asterisk 
 sets some special channel variables or something when a call is transfered 
 with the REFER method.
 Basically, I'm trying to figure out if it is possible to somehow get a 
 transferred call back to the transferrer (as it is done with the built-in 
 atxfer) after X seconds (or an unsuccessful attempt).
 
 Using a timeout in the Dial command is not suitable unless I am able to tell 
 somehow that the call in question is being forwarded (which is of course not 
 the case, as the Dial command is called befer the REFER is sent).
 
 Can anyone think of a way to get the call back to the transferrer after this 
 timeout?
 
THe transferred call is sent to a context set with the channel variable 
TRANSFER_CONTEXT before you call DIAL().

In there, run DUMPCHAN to see which variables you have and then dial with a 
timeout. After the timeout, dial back. 

/O
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[asterisk-users] Attended transfer and 'pbx-invalid' - 1.4.26

[Gabriel Ortiz Lour]

Hi,

  I've created a tiny dialplan to test the return of a call on transfers,
like this: (I had to use the DEVSTATE backport here)

[phones]
exten = _12XX,1,Dial(SIP/${EXTEN},6,tT)
exten = _12XX,n,GotoIf($[ x${BLINDTRANSFER} = x ]?noBT)
exten = _12XX,n,Set(DIALRET=${CUT(BLINDTRANSFER,-,1)});
exten = _12XX,n,Goto(dRet)
exten = _12XX,n(noBT),GotoIf($[ x${TRANSFERERNAME} = x ]?sai)
exten = _12XX,n,Set(DIALRET=${CUT(TRANSFERERNAME,-,1)});
exten = _12XX,n,GotoIf($[ ${DEVSTATE(${DIALRET})} = INUSE ]?sai);
exten = _12XX,n(dRet),Set(CALLERID(all)=RET_${EXTEN} ${CALLERID(num)})
exten = _12XX,n,Dial(${DIALRET},,mTt)
exten = _12XX,n(sai),Hangup()

It all works like a charm, except that when I do an atxfer and dial another
SIP and it rings, but dont answer, asterisk plays the 'pbx-invalid' sound,
that is a bit confusing, because the phone is there and actually rang . Here
is the CLI output

*CLI
-- Executing [1...@irrestrito-user:1] Dial(SIP/1202-08330f80,
SIP/1201|6|tT) in new stack
-- Called 1201
-- SIP/1201-08335530 is ringing
-- SIP/1201-08335530 answered SIP/1202-08330f80
-- Started music on hold, class 'default', on SIP/1202-08330f80
-- SIP/1201-08335530 Playing 'pbx-transfer' (language 'en')
-- Executing [1...@irrestrito-user:1]
Dial(Local/1...@irrestrito-user-70b2,2, SIP/1203|6|tT) in new stack
-- Called 1203
-- SIP/1203-08325260 is ringing
-- Local/1...@irrestrito-user-70b2,1 is ringing


 Ring and no answer...

-- Nobody picked up in 6000 ms
-- Executing [1...@irrestrito-user:2]
GotoIf(Local/1...@irrestrito-user-70b2,2, 1?noBT) in new stack
-- Goto (irrestrito-user,1203,5)
-- Executing [1...@irrestrito-user:5]
GotoIf(Local/1...@irrestrito-user-70b2,2, 0?sai) in new stack
-- Executing [1...@irrestrito-user:6]
Set(Local/1...@irrestrito-user-70b2,2, DIALRET=SIP/1201) in new stack
-- Executing [1...@irrestrito-user:7]
GotoIf(Local/1...@irrestrito-user-70b2,2, 1?sai) in new stack
-- Goto (irrestrito-user,1203,10)
-- Executing [1...@irrestrito-user:10]
Hangup(Local/1...@irrestrito-user-70b2,2, ) in new stack
  == Spawn extension (irrestrito-user, 1203, 10) exited non-zero on
'Local/1...@irrestrito-user-70b2,2'
-- Stopped music on hold on SIP/1202-08330f80
? -- SIP/1201-08335530 Playing 'pbx-invalid' (language 'en')


am I doing something wrong?

Thanks,
Gabriel
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Re: [asterisk-users] Attended transfer and dialplan

[Steve Murphy]

On Fri, May 29, 2009 at 11:35 PM, Olivier oza-4...@myamail.com wrote:



 2009/5/29 Danny Nicholas da...@debsinc.com

  I’m pretty sure that attended transfer is a “features” function, not a
 dialplan one.


 Yes, you're right but do you think there's such a big difference between
 both that it shouldn't be easy or even possible to add support of attended
 transfer in dialplan ?

 What I have in my mind is this :

 Today, Dial application M or U options allows macro execution when caller
 and callee are connected.
 What if this same macro could be also launched during some later events
 (like attended transfer) ?
 With features.conf, you could then specify :
 - how lo launch an attended transfer (which key to type as today),
 - if a given feature (attended transfer, parking, ...) should be
 supported by Dial macro option (for compatibilty, default could be set to
 none)

 and with extension.conf, you could specify :
 - which specific treatment (sending UserEvents, launching an external
 program, ...) to apply

 In this puzzle, if Asterisk could support a few more standard variables
 like ATTENDED_TRANSFERER ATTENDED_TRANSFER_TARGET, you would everything to
 define and run attended transfers specific logic :

 exten = 123,1,Dial(SIP/123,M(mymacro^arg1^arg2)) ; mymacro is launched
 upon connection and specified (in features.conf) events

 [macro-mymacro]
 GotoIf(x${ATTENDED_TRANSFERER}, 

 What about that ?


Olivier--

This is actually not a bad idea.Why single out just the Attended xfer? Why
would you treat
attended xfers differently than blind xfers? Just curious.

Also, calling just one macro for all features seems a bit restricted. Why
not allow the features.conf
to specify which macro/gosub to call, for each feature?  Dial is already
overloaded with options,
anything that could be offloaded would probably be desirable. Plus, calls
that were not initiated by
a dialplan Dial() invocation might not be able to provide that option.

Another question: what do you need this functionality to *do*? It could be
that there is an already
existing functionality that you could exploit to get the same results?

murf





 On my system I do *2 and asterisk says transfer, then I punch in the new
 extension.




  --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
 *Sent:* Friday, May 29, 2009 10:29 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Attended transfer and dialplan



 Hi,

 How can you add specific statements into Asterisk dialplan (extension.ael,
 ...) for attented transfers ?

 I can see Asterisk sending Transfer or Masquerade events through AMI (in
 1.6.1) but I could use an external program to catch those events but I would
 prefer to use dialplan instead.

 Any idea ?

 Regards

 --
Steve Murphy
ParseTree Corp
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[asterisk-users] Attended transfer and dialplan

[Olivier]

Hi,

How can you add specific statements into Asterisk dialplan (extension.ael,
...) for attented transfers ?

I can see Asterisk sending Transfer or Masquerade events through AMI (in
1.6.1) but I could use an external program to catch those events but I would
prefer to use dialplan instead.

Any idea ?

Regards
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Re: [asterisk-users] Attended transfer and dialplan

[Danny Nicholas]

I'm pretty sure that attended transfer is a features function, not a
dialplan one.

 

On my system I do *2 and asterisk says transfer, then I punch in the new
extension.

 

 

  _  

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Friday, May 29, 2009 10:29 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Attended transfer and dialplan

 

Hi,

How can you add specific statements into Asterisk dialplan (extension.ael,
...) for attented transfers ?

I can see Asterisk sending Transfer or Masquerade events through AMI (in
1.6.1) but I could use an external program to catch those events but I would
prefer to use dialplan instead.

Any idea ?

Regards

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Re: [asterisk-users] Attended transfer and dialplan

[Olivier]

2009/5/29 Danny Nicholas da...@debsinc.com

  I’m pretty sure that attended transfer is a “features” function, not a
 dialplan one.


Yes, you're right but do you think there's such a big difference between
both that it shouldn't be easy or even possible to add support of attended
transfer in dialplan ?

What I have in my mind is this :

Today, Dial application M or U options allows macro execution when caller
and callee are connected.
What if this same macro could be also launched during some later events
(like attended transfer) ?
With features.conf, you could then specify :
- how lo launch an attended transfer (which key to type as today),
- if a given feature (attended transfer, parking, ...) should be supported
by Dial macro option (for compatibilty, default could be set to none)

and with extension.conf, you could specify :
- which specific treatment (sending UserEvents, launching an external
program, ...) to apply

In this puzzle, if Asterisk could support a few more standard variables like
ATTENDED_TRANSFERER ATTENDED_TRANSFER_TARGET, you would everything to define
and run attended transfers specific logic :

exten = 123,1,Dial(SIP/123,M(mymacro^arg1^arg2)) ; mymacro is launched
upon connection and specified (in features.conf) events

[macro-mymacro]
GotoIf(x${ATTENDED_TRANSFERER}, 

What about that ?





 On my system I do *2 and asterisk says transfer, then I punch in the new
 extension.




  --

 *From:* asterisk-users-boun...@lists.digium.com [mailto:
 asterisk-users-boun...@lists.digium.com] *On Behalf Of *Olivier
 *Sent:* Friday, May 29, 2009 10:29 AM
 *To:* Asterisk Users Mailing List - Non-Commercial Discussion
 *Subject:* [asterisk-users] Attended transfer and dialplan



 Hi,

 How can you add specific statements into Asterisk dialplan (extension.ael,
 ...) for attented transfers ?

 I can see Asterisk sending Transfer or Masquerade events through AMI (in
 1.6.1) but I could use an external program to catch those events but I would
 prefer to use dialplan instead.

 Any idea ?

 Regards

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[asterisk-users] Attended transfer problems

[James Lamanna]

Hi,
A couple of our customers are having issues with doing attended transfers.
What happens is Caller A receives a call, they transfer to Caller B,
tell Caller B who is calling, etc.. and then
hit the Transfer key again to transfer the call.
Caller A's side hangs up as expected, but the call is never completed
to Caller B, and the call is dropped.
Blind transfers at these couple companies work fine.
The transfers are occuring on Linksys 942/962 phones, and I have about
20 other customers using these same phones that
have no problems.

My only guess so far is that in both of these cases, the SOHO
firewalls that these companies are using (the * Box is on a public
internet IP),
is rewriting the VIA header to [firewall address]:[firewall port].
Could this be causing the issue?
I'm going to try replacing the firewall with ones I know don't do
packet rewriting, so maybe that is the answer,
but any additional insight would be of a big help!

Thanks.

-- James

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Re: [asterisk-users] Attended transfer to Queue

[Steve Davies]

On 12/3/07, Steve Davies [EMAIL PROTECTED] wrote:

 *BUMP* Does anyone have a workaround for the above? When a call is
 attended-transferred to a queue, MOH is stopped. Perhaps someone can
 tell me if it is fixed in 1.4.x ?

Sorry, replying to my own thread - Even if there is no fix, if someone
can suggest a useful way that this might be handled, I am happy to
have a go at a fix - I just cannot think how it could be done.

Steve

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Re: [asterisk-users] Attended transfer to Queue

[Steve Davies]

On 11/27/07, Steve Davies [EMAIL PROTECTED] wrote:
 Hi,

 I will confess immediately that this is only tested on 1.2.24, and I
 would be interested to know if it happens on 1.4, but I cannot find a
 bug-tracker entry which represents this issue.

 Consider a PSTN call which comes into asterisk, and is bridged to a
 SIP phone. The phone operator then places the call on hold (hold music
 plays) and a second call is made from this handset to a Queue...
 Operator can now hear hold music from the queue.

 The operator then completes the attended transfer, bridging the
 initial PSTN call to the Queue.

 The system sees a transfer being completed, and stops MOH on both of
 the channels. This means that the caller is correctly transferred to a
 queue, but the MOH has been stopped, and they hear silence.

 While this behaviour is expected, it is not ideal and if anyone
 can point me at a workaround, or an existing bug-tracker entry, I
 would be most grateful.

*BUMP* Does anyone have a workaround for the above? When a call is
attended-transferred to a queue, MOH is stopped. Perhaps someone can
tell me if it is fixed in 1.4.x ?

Thanks
Steve

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[asterisk-users] Attended transfer to Queue

[Steve Davies]

Hi,

I will confess immediately that this is only tested on 1.2.24, and I
would be interested to know if it happens on 1.4, but I cannot find a
bug-tracker entry which represents this issue.

Consider a PSTN call which comes into asterisk, and is bridged to a
SIP phone. The phone operator then places the call on hold (hold music
plays) and a second call is made from this handset to a Queue...
Operator can now hear hold music from the queue.

The operator then completes the attended transfer, bridging the
initial PSTN call to the Queue.

The system sees a transfer being completed, and stops MOH on both of
the channels. This means that the caller is correctly transferred to a
queue, but the MOH has been stopped, and they hear silence.

While this behaviour is expected, it is not ideal and if anyone
can point me at a workaround, or an existing bug-tracker entry, I
would be most grateful.

Thanks,
Steve

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[asterisk-users] Attended Transfer of a queue call fails

[Alexander Topolanek]

Hi,

I'm using Grandstreams as the agents phone of a queue. Attended
transfers in a normal situation (direct call to the extension) work
fine, but when the agent has a queue call and tries to transfer it to
another sip extension the called party is hung up. 

Transfer is to pick another line on the Grandstream, dial the other
extension, wait for a pickup, and then press the transfer button + the
parked line on the Grandstream. I can reproduce that in several
asterisk, and the inbound call comes from an mISDN channel or a sip
channel, and with similar results using a SNOM 190 terminal

Asterisk is a 1.2.14-BRIstuffed-0.3.0-PRE-1x built on Gentoo

any ideas?

best regards
-- 
Alexander 

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[asterisk-users] Attended Transfer with snom phones

[Michael Boers]

I have setup an asterisk based phone system using snom-320 (SIP based)
phones.

I would like to change what seems to be the default procedure for an
attended call transfer.  Right now, the phone user places the call on hold,
calls the extension using a extension button on the phone, speaks with the
call recipient , and presses transfer to transfer the held call.

The users would like to press transfer, call the extension using a
extension button on the phone, speak to the recipient, then hangup to
complete the call.

Can you give me a suggestion as to how to do this.

Thank you,

Michael Boers
Michael Scott Technology
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Re: [asterisk-users] Attended Transfer on queue_log

[equis software]

Yes, I have de same problem...I dont know if there is an error...

Regards

On 12/15/06, Miguel Paolino [EMAIL PROTECTED] wrote:


I'm using asterisk blind/attended transfer feature on  a queue (also
tried with sip phones feature), and both type of transfers work fine. The
problem is that attended trasfers doesn't get logged to queue_log, but blind
transfers are logged just fine. Anyone knows if this is the correct
behavior?

--
Regards,

Miguel Paolino
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[asterisk-users] Attended Transfer on queue_log

[Miguel Paolino]

I'm using asterisk blind/attended transfer feature on  a queue (also  tried
with sip phones feature), and both type of transfers work fine. The problem
is that attended trasfers doesn't get logged to queue_log, but blind
transfers are logged just fine. Anyone knows if this is the correct
behavior?

--
Regards,

Miguel Paolino
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Re: [asterisk-users] Attended Transfer

[Anselm Martin Hoffmeister]

Am Dienstag, den 05.12.2006, 20:07 -0600 schrieb Eric ManxPower
Wieling:
 Attended transfer is supported by every decent SIP device out there.  It 
 is a basic phone feature.  There are a few SIP devices out there that do 
 NOT support attended transfer but I would not call them decent.  The 
 GS BT101 and the FREE version of X-Ten's phone are both devices that do 
 not support attended transfer.

GRANDSTREAM-NOT TRUE.

I have a Budgetone 101 and a 102 here, both doing transfers fine. It
took me a while to figure that though, as I had no manual in the
beginning, and the PDF I found was way older than the firmware on my
devices.

However, transfer on BG100 series (most recent firmware anyway) works by
pressing FLASH during a connection, getting you a new dialtone. Then
dial the extensions and press SEND (I have no automagic end-of-number
detection, so it's exactly the same you would do on a regular call). You
will be connected to the second person. You can switch between those two
connections with FLASH. Pressing TRANSFER will connect those two
parties, disconnecting you.

UN-attented transfer works also. press TRANSFER the number SEND and
your caller will be redirected to the other extension.

That said, I'm not too much a fan of those Budgetones - they look cheap,
and have an absolute minimum of functions. On the other hand, those
_are_ cheap, and quite robust when it comes to dropping from the desk
etc. And their functions work with Asterisk, including MWI and message
button.

BR
Anselm

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Re: [asterisk-users] Attended Transfer

[Henry.L.Coleman]

Thats quite correct however if you have a multi-line phone like
Grandstream GXP 2000 or Aastra 480iyou can put the call on hold manually.
As for using an ATA, You can program the ATA using the vertical service
codes. In this case you can use a code to tell the (SIP) ATA that you want
to do a transfer. You must make certain that the code you choose doesn't
conflit with Asterisk's feature codes.

Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada


 Henry, according with voip-info.org, attended transfer is
 While on conversation with another party, you dial the atxfer key
 sequence. Asterisk says Transfer then gives you a dial tone, while
 putting the other party on hold. You dial the transferee number and
 talk with the transferee to introduce the call, then you can hang up
 and the other party will be connected with the transferee. In case the
 transferee does not want to answer the call, he/she simply hangs up
 and you will be back to your original conversation.
 The callee is put on hold automatically

 Eric, attended transfer is only possible with an ATA??

 On 12/5/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
 Henry.L.Coleman wrote:
  Attended transfer is really four functions
  1. Put the caller on Hold while you dial another number
  2. Speak to the dialed number (announce the call)
  3. Patch the call on hold to the other party using transfer button.
  4. Disconnect (otherwise this would be a 3 party conference)
 
  How these functions work depend on what type of device the operator is
  using. SIP phones have this functionality ie a hold button, a transfer
  button and multi-line appearances. If you are using an ATA with an
  ordinary
  phone and standard dial-pad then you may be able to put a call on hold
 by
   using the * and transfer by #. But obviously one is limited to
 the
  vacant digits on the dial pad (DTMF).

 With an ATA you would use FLASH (aka RECALL)
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 --
 Arlen Nascimento
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Re: [asterisk-users] Attended Transfer

[Fran Oliveira]

I had problems with featuredigittimeout . It was too short and betwen digit
and digit was happened a timeout.
modify to featuredigittimeout = 1000


2006/12/5, Arlen Nascimento [EMAIL PROTECTED]:


Dear List,

I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable
attended transfer feature. but i just can't do it work. I've already
set atxfer = * (and many other combinations) and all extensions on
extensions.conf have the t and T option. But when I'm going to test,
it doesn't work. Is there any other file that i have to configure in
order to make it work? I've already looked at google so many times and
nothing

Does anybody have an idea??

Regards
--
Arlen Nascimento
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[asterisk-users] Attended Transfer

[Arlen Nascimento]

Dear List,

I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable
attended transfer feature. but i just can't do it work. I've already
set atxfer = * (and many other combinations) and all extensions on
extensions.conf have the t and T option. But when I'm going to test,
it doesn't work. Is there any other file that i have to configure in
order to make it work? I've already looked at google so many times and
nothing

Does anybody have an idea??

Regards
--
Arlen Nascimento
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Re: [asterisk-users] Attended Transfer

[Henry.L.Coleman]

Attended transfer is really four functions
1. Put the caller on Hold while you dial another number
2. Speak to the dialed number (announce the call)
3. Patch the call on hold to the other party using transfer button.
4. Disconnect (otherwise this would be a 3 party conference)

How these functions work depend on what type of device the operator is
using. SIP phones have this functionality ie a hold button, a transfer
button and multi-line appearances. If you are using an ATA with an
ordinary
phone and standard dial-pad then you may be able to put a call on hold by 
 using the * and transfer by #. But obviously one is limited to the
vacant digits on the dial pad (DTMF).
Note: If your analog (POTS) phone has a hold button this will not work
as the hold button simply applies a resistive load to hold the loop in
an off-hook status.
Hope this helps...

Henry L.Coleman CEO
*VoIP-PBX* 1-866-415-5355
Toronto Ontario
Canada


 Dear List,

 I've been working with Asterisk CVS-v1-0-09 and i'm trying to enable
 attended transfer feature. but i just can't do it work. I've already
 set atxfer = * (and many other combinations) and all extensions on
 extensions.conf have the t and T option. But when I'm going to test,
 it doesn't work. Is there any other file that i have to configure in
 order to make it work? I've already looked at google so many times and
 nothing

 Does anybody have an idea??

 Regards
 --
 Arlen Nascimento
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Re: [asterisk-users] Attended Transfer

[Eric \ManxPower\ Wieling]

Henry.L.Coleman wrote:

Attended transfer is really four functions
1. Put the caller on Hold while you dial another number
2. Speak to the dialed number (announce the call)
3. Patch the call on hold to the other party using transfer button.
4. Disconnect (otherwise this would be a 3 party conference)

How these functions work depend on what type of device the operator is
using. SIP phones have this functionality ie a hold button, a transfer
button and multi-line appearances. If you are using an ATA with an
ordinary
phone and standard dial-pad then you may be able to put a call on hold by 
 using the * and transfer by #. But obviously one is limited to the

vacant digits on the dial pad (DTMF).


With an ATA you would use FLASH (aka RECALL)
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Re: [asterisk-users] Attended Transfer

[Arlen Nascimento]

Henry, according with voip-info.org, attended transfer is
While on conversation with another party, you dial the atxfer key
sequence. Asterisk says Transfer then gives you a dial tone, while
putting the other party on hold. You dial the transferee number and
talk with the transferee to introduce the call, then you can hang up
and the other party will be connected with the transferee. In case the
transferee does not want to answer the call, he/she simply hangs up
and you will be back to your original conversation.
The callee is put on hold automatically

Eric, attended transfer is only possible with an ATA??

On 12/5/06, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:

Henry.L.Coleman wrote:
 Attended transfer is really four functions
 1. Put the caller on Hold while you dial another number
 2. Speak to the dialed number (announce the call)
 3. Patch the call on hold to the other party using transfer button.
 4. Disconnect (otherwise this would be a 3 party conference)

 How these functions work depend on what type of device the operator is
 using. SIP phones have this functionality ie a hold button, a transfer
 button and multi-line appearances. If you are using an ATA with an
 ordinary
 phone and standard dial-pad then you may be able to put a call on hold by
  using the * and transfer by #. But obviously one is limited to the
 vacant digits on the dial pad (DTMF).

With an ATA you would use FLASH (aka RECALL)
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--
Arlen Nascimento
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Re: [asterisk-users] Attended Transfer

[Eric \ManxPower\ Wieling]

Arlen Nascimento wrote:

Henry, according with voip-info.org, attended transfer is
While on conversation with another party, you dial the atxfer key
sequence. Asterisk says Transfer then gives you a dial tone, while
putting the other party on hold. You dial the transferee number and
talk with the transferee to introduce the call, then you can hang up
and the other party will be connected with the transferee. In case the
transferee does not want to answer the call, he/she simply hangs up
and you will be back to your original conversation.
The callee is put on hold automatically

Eric, attended transfer is only possible with an ATA??


Attended transfer is supported by every decent SIP device out there.  It 
is a basic phone feature.  There are a few SIP devices out there that do 
NOT support attended transfer but I would not call them decent.  The 
GS BT101 and the FREE version of X-Ten's phone are both devices that do 
not support attended transfer.


There are a couple of reasons to want to do DTMF Transfers (configured 
in Asterisk via /etc/asterisk/features.conf.  One reason might be that 
you are stuck, for some reason, with a phone that does not support 
attended transfer.  Another reason would be if you have several 
different types of phones and ATAs around and do not want to make users 
learn different ways to do a transfer, depending on the phone the person 
is using at the moment.  Another reason, and one I think is the most 
common, is that you simply don't know any better.

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Re: [asterisk-users] Attended transfer hanging PRI channel

[Doug Lytle]

Michael Welter wrote:
The attendant attempts an attended call transfer (all phones are 
IP501).  The attendant pushes hold, transfer, dials the extension 
and announces the call.  When the attendant pushes transfer the 
second time, the original call is lost.


The attendant is doing it incorrectly.  Pressing the hold button isn't 
necessary.  Pressing transfer will automatically put the call on hold.


Doug


-- Ben Franklin quote: Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety.


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Re: [asterisk-users] Attended transfer hanging PRI channel

[Michael Welter]

It happens both when the hold button is pushed and when not pushed.

Blind transfers seem to work properly.


Doug Lytle wrote:

Michael Welter wrote:
The attendant attempts an attended call transfer (all phones are 
IP501).  The attendant pushes hold, transfer, dials the extension 
and announces the call.  When the attendant pushes transfer the 
second time, the original call is lost.


The attendant is doing it incorrectly.  Pressing the hold button isn't 
necessary.  Pressing transfer will automatically put the call on hold.


Doug


-- Ben Franklin quote: Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety.


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--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
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Re: [asterisk-users] Attended transfer hanging PRI channel

[Doug Lytle]

Michael Welter wrote:

It happens both when the hold button is pushed and when not pushed.

Blind transfers seem to work properly.

We are currently using firmware 1.5.2, but you could try a different 
version?


Doug

-- Ben Franklin quote: Those who would give up Essential Liberty to 
purchase a little Temporary Safety, deserve neither Liberty nor Safety.


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[asterisk-users] Attended transfer hanging PRI channel

[Michael Welter]

The attendant attempts an attended call transfer (all phones are IP501). 
 The attendant pushes hold, transfer, dials the extension and 
announces the call.  When the attendant pushes transfer the second 
time, the original call is lost.


The reason this is a big problem is that the PRI channel for the call 
remains busy.  Subsequent inbound calls on that channel are rejected.


Asterisk 1.2.12.1, Polycom SIP 1.6.6.

Has anyone seen this?  Thanks.


--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
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[asterisk-users] attended transfer unreliable (2nd try)

[Stefan Friedrich]

Is there really nobody who has any idea about this?help would be really apreciated, as otherwise we're forced to buy a conventional pbxDate: 29.09.2006 15:33Subject: attended transfer unreliable
To: asterisk-users@lists.digium.comHi,running asterisk 1.2.9 with freepbx 2.1.1, I have a strange problem:sometimes, call transfer works as expectet, and sometimes not. So far, I couldn't figure out any pattern in this behaviour,
features.conf
:featuredigittimeout = 1500atxfer = *3-works:# user enters *Sep 29 14:52:14 DEBUG[21578] channel.c: Got DTMF on channel (SIP/230-a983)Sep 29 14:52:14 DEBUG[21578] 
channel.c: Bridge stops bridging channels SIP/210-859a and SIP/230-a983Sep 29 14:52:14 DEBUG[21578] res_features.c: Feature interpret: chan=SIP/210-859a, peer=SIP/230-a983, sense=2, features=18Sep 29 14:52:14 DEBUG[21578] res_features.c: Set time limit to 1500
Sep 29 14:52:14 VERBOSE[21578] logger.c: -- Attempting native bridge of SIP/210-859a and SIP/230-a983# user enters 3Sep 29 14:52:15 DEBUG[21578] channel.c: Got DTMF on channel (SIP/230-a983)Sep 29 14:52:15 DEBUG[21578] 
channel.c: Bridge stops bridging channels SIP/210-859a and SIP/230-a983Sep 29 14:52:15 DEBUG[21578] res_features.c: Feature interpret: chan=SIP/210-859a, peer=SIP/230-a983, sense=2, features=18# here is the transfer:
Sep 29 14:52:15 DEBUG[21578] res_features.c: Executing Attended Transfer SIP/210-859a, SIP/230-a983 (sense=2) XXXSep 29 14:52:15 VERBOSE[21578] logger.c: -- Started music on hold, class 'default', on SIP/210-859a
-doesn't work:# user enters *Sep 29 09:17:54 DEBUG[20534] channel.c: Got DTMF on channel (SIP/230-9e2a)Sep 29 09:17:54 DEBUG[20534] channel.c: Bridge stops bridging channels SIP/210-c701 and SIP/230-9e2a
Sep 29 09:17:54 DEBUG[20534] res_features.c: Feature interpret: chan=SIP/210-c701, peer=SIP/230-9e2a, sense=2, features=18Sep 29 09:17:54 DEBUG[20534] res_features.c: Set time limit to 1500Sep 29 09:17:54 VERBOSE[20534] 
logger.c: -- Attempting native bridge of SIP/210-c701 and SIP/230-9e2a#user enters 3Sep 29 09:17:55 DEBUG[20534] channel.c: Got DTMF on channel (SIP/230-9e2a)Sep 29 09:17:55 DEBUG[20534] channel.c: Bridge stops bridging channels SIP/210-c701 and SIP/230-9e2a
Sep 29 09:17:55 DEBUG[20534] res_features.c: Feature interpret: chan=SIP/210-c701, peer=SIP/230-9e2a, sense=2, features=18# no transferSep 29 09:17:55 DEBUG[17507] chan_sip.c: Stopping retransmission on '

[EMAIL PROTECTED]' of Request 102: Match FoundSep 29 09:17:55 DEBUG[17507] chan_sip.c: Stopping retransmission on '
[EMAIL PROTECTED]
' of Request 103: Match FoundSep 29 09:17:55 VERBOSE[20534] logger.c: -- Attempting native bridge of SIP/210-c701 and SIP/230-9e2a--when we have a timeout, it looks different:
Sep 29 12:00:34 DEBUG[21122] channel.c: Got DTMF on channel (SIP/240-6746)Sep 29 12:00:34 DEBUG[21122] channel.c: Bridge stops bridging channels Zap/2-1 and SIP/240-6746Sep 29 12:00:34 DEBUG[21122] res_features.c: Feature interpret: chan=Zap/2-1, peer=SIP/240-6746, sense=2, features=18
Sep 29 12:00:34 DEBUG[21122] res_features.c: Set time limit to 1500Sep 29 12:00:36 DEBUG[21122] channel.c: Got DTMF on channel (SIP/240-6746)Sep 29 12:00:36 DEBUG[21122] channel.c: Bridge stops bridging channels Zap/2-1 and SIP/240-6746
Sep 29 12:00:36 DEBUG[21122] res_features.c: Timed out for feature!hope your can help meStefan


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Re: [asterisk-users] attended transfer unreliable (2nd try)

[Doug Lytle]

Stefan Friedrich wrote:

Is there really nobody who has any idea about this?
help would be really apreciated, as otherwise we're forced to buy a 
conventional pbx


Have you tried upgrading to 1.2.12.1 or 1.2 branch from SVN?  There have 
been a few fixes in the branch that may help.


You can get instructions towards the center of the page at:

http://www.asterisk.org/download

Doug


--

Ben Franklin quote:

Those who would give up Essential Liberty to purchase a little Temporary Safety, 
deserve neither Liberty nor Safety.


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Re: [asterisk-users] attended transfer unreliable (2nd try)

[Florian Hars]

Doug Lytle wrote:

Have you tried upgrading to 1.2.12.1 or 1.2 branch from SVN?


Transfer (rather, dynamic features in general) is broken in 1.2.12.1:
http://bugs.digium.com/view.php?id=7982
So you should try the version from the SVN branch.

Yours, Florian.
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[asterisk-users] attended transfer unreliable

[Stefan Friedrich]

Hi,running asterisk 1.2.9 with freepbx 2.1.1, I have a strange problem:sometimes, call transfer works as expectet, and sometimes not. So far, I couldn't figure out any pattern in this behaviour,features.conf
:featuredigittimeout = 1500atxfer = *3-works:# user enters *Sep 29 14:52:14 DEBUG[21578] channel.c: Got DTMF on channel (SIP/230-a983)Sep 29 14:52:14 DEBUG[21578] 
channel.c: Bridge stops bridging channels SIP/210-859a and SIP/230-a983Sep 29 14:52:14 DEBUG[21578] res_features.c: Feature interpret: chan=SIP/210-859a, peer=SIP/230-a983, sense=2, features=18Sep 29 14:52:14 DEBUG[21578] res_features.c: Set time limit to 1500
Sep 29 14:52:14 VERBOSE[21578] logger.c: -- Attempting native bridge of SIP/210-859a and SIP/230-a983# user enters 3Sep 29 14:52:15 DEBUG[21578] channel.c: Got DTMF on channel (SIP/230-a983)Sep 29 14:52:15 DEBUG[21578] 
channel.c: Bridge stops bridging channels SIP/210-859a and SIP/230-a983Sep 29 14:52:15 DEBUG[21578] res_features.c: Feature interpret: chan=SIP/210-859a, peer=SIP/230-a983, sense=2, features=18# here is the transfer:
Sep 29 14:52:15 DEBUG[21578] res_features.c: Executing Attended Transfer SIP/210-859a, SIP/230-a983 (sense=2) XXXSep 29 14:52:15 VERBOSE[21578] logger.c: -- Started music on hold, class 'default', on SIP/210-859a
-doesn't work:# user enters *Sep 29 09:17:54 DEBUG[20534] channel.c: Got DTMF on channel (SIP/230-9e2a)Sep 29 09:17:54 DEBUG[20534] channel.c: Bridge stops bridging channels SIP/210-c701 and SIP/230-9e2a
Sep 29 09:17:54 DEBUG[20534] res_features.c: Feature interpret: chan=SIP/210-c701, peer=SIP/230-9e2a, sense=2, features=18Sep 29 09:17:54 DEBUG[20534] res_features.c: Set time limit to 1500Sep 29 09:17:54 VERBOSE[20534] 
logger.c: -- Attempting native bridge of SIP/210-c701 and SIP/230-9e2a#user enters 3Sep 29 09:17:55 DEBUG[20534] channel.c: Got DTMF on channel (SIP/230-9e2a)Sep 29 09:17:55 DEBUG[20534] channel.c: Bridge stops bridging channels SIP/210-c701 and SIP/230-9e2a
Sep 29 09:17:55 DEBUG[20534] res_features.c: Feature interpret: chan=SIP/210-c701, peer=SIP/230-9e2a, sense=2, features=18# no transferSep 29 09:17:55 DEBUG[17507] chan_sip.c: Stopping retransmission on '
[EMAIL PROTECTED]' of Request 102: Match FoundSep 29 09:17:55 DEBUG[17507] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]
' of Request 103: Match FoundSep 29 09:17:55 VERBOSE[20534] logger.c: -- Attempting native bridge of SIP/210-c701 and SIP/230-9e2a--when we have a timeout, it looks different:
Sep 29 12:00:34 DEBUG[21122] channel.c: Got DTMF on channel (SIP/240-6746)Sep 29 12:00:34 DEBUG[21122] channel.c: Bridge stops bridging channels Zap/2-1 and SIP/240-6746Sep 29 12:00:34 DEBUG[21122] res_features.c: Feature interpret: chan=Zap/2-1, peer=SIP/240-6746, sense=2, features=18
Sep 29 12:00:34 DEBUG[21122] res_features.c: Set time limit to 1500Sep 29 12:00:36 DEBUG[21122] channel.c: Got DTMF on channel (SIP/240-6746)Sep 29 12:00:36 DEBUG[21122] channel.c: Bridge stops bridging channels Zap/2-1 and SIP/240-6746
Sep 29 12:00:36 DEBUG[21122] res_features.c: Timed out for feature!hope your can help meStefan
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[asterisk-users] Attended transfer and parking calls

[Elpidio Ramos]

Can anyone help me with information on how to implement or use the Attended transfer and parking calls?I have tried the extension 700 getting a number for the parked call but I was never been able to retrieve the call (don't know how) by dialing the indicated extension number.Also, we need to learn how to transfer a call by talking first to the extension before actually transfering the call (I assume this is called attended transfer?).Any help will be highly appreciated.Elpidio___
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Re: [asterisk-users] Attended transfer and parking calls

[Andrew Joakimsen]

What sort of handset/ata are you using? On the IP phones it should all
be in the menus. On the ATAs we use blind transfer would be flash
and dial the numbe to transfer too, then hang up. Attened transfer is
the same thing, except you wait for the other party to answer then
hang up, the call is connected.

Call park would be the same thing. flash 700  [seven zero one]
hangup. Then dial 701 to get the call back, you need include =
parkedcalls in the correct place of your extensions.conf

On 9/15/06, Elpidio Ramos [EMAIL PROTECTED] wrote:


Can anyone help me with information on how to implement or use the Attended
transfer and parking calls?

I have tried the extension 700 getting a number for the parked call but I
was never been able to retrieve the call (don't know how) by dialing the
indicated extension number.

Also, we need to learn how to transfer a call by talking first to the
extension before actually transfering the call (I assume this is called
attended transfer?).

Any help will be highly appreciated.


Elpidio
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Re: [asterisk-users] Attended transfer and parking calls

[Elpidio Ramos]

Good information.  We use mostly analog phones and some of our extensions are soft phones.One question I still have:When I use flash, I hear the "TRANSFER" prompt and if I dial an extension the call goes direct without a chance for me to talk to the guy on the other side.Can it be that I need to include something else in my dial plan?  Flash doesn't seem to work with our analog phones.Thanks a lot.Andrew Joakimsen [EMAIL PROTECTED] wrote:  What sort of handset/ata are you using? On the IP phones it should allbe in the menus. On the ATAs we use blind transfer would be and dial the numbe to transfer too, then hang up. Attened transfer isthe same thing, except you wait
 for the other party to answer thenhang up, the call is connected.Call park would be the same thing. 700 [seven zero one]hangup. Then dial 701 to get the call back, you need include =parkedcalls in the correct place of your extensions.confOn 9/15/06, Elpidio Ramos <[EMAIL PROTECTED]>wrote: Can anyone help me with information on how to implement or use the Attended transfer and parking calls? I have tried the extension 700 getting a number for the parked call but I was never been able to retrieve the call (don't know how) by dialing the indicated extension number. Also, we need to learn how to transfer a call by talking first to the extension before actually transfering the call (I assume this is called attended transfer?). Any help will be highly appreciated. Elpidio
 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users  Elpidio Ramos PresidentRM International ServicesSA CV Web: http://www.ramosoft.com Mex: +52 (55) 5116-9804 Office +52 (55) 5116-9805 Fax+52 (55)1755-6601 CellUSA: +1 (801) 494-1415 Office   +1 (240) 250-8264 Fax   +1 (801)
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[asterisk-users] Attended Transfer Asterisk 1.2.11

[Alberto Sagredo]

Im updating from 1.2.9.1 to 1.2.11 and im having a issue with attendad 
transfer via SPA 941 that i did not have with 1.2.9.1. I get this 
message on Cli log.


Sep 14 16:09:52 NOTICE[5780]: chan_sip.c:6897 get_refer_info: Supervised 
transfer requested, but unable to find callid 
'[EMAIL PROTECTED]'.  Both legs must reside on Asterisk box 
to transfer at this time.


I have canreinvite=yes on all extensions, and tried with canreinvite=no, 
but same happens.


When i press tranfer, i could talk with destination extension, but when 
i transfer call, it hangups both sides.


Regards


--
Alberto Sagredo
I+D Area (Asterisk // Cisco-Linksys)
Peoplecall


Email : [EMAIL PROTECTED]
Blog: http://www.voipnovatos.es

Tel./Ph. : +34 91 120 5080
Tel. Dir./Dir. Ph.: 700 757 139 / 91 120 50 39
Fax./Fax.: +34 91 661 9460
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[asterisk-users] Attended Transfer call return with asterisk + sipura spa2002

[Thomas Artner]

Hi!

I have connected my analog phones to an asterisk box with sipura spa2002
devices.
I can do an attended transfer by taking the call which should be
transferred, pressing the flash button, dialing the number to which the
call should be transferred and now i can hang up or talk to the person
who receives the transferred call.

Thats working perfectly.

But if the other person (the person who gets the transferred call) isnt
on his place and doesnt take the call, the call gets disconnected after
about a minute.

Does anyone have an idea how i could make that the call gets transferred
back (to the person who initially did the transfer) automatically?

thx,
Tom

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Re: [Asterisk-Users] attended transfer issue

[Mike Dawson]

I believe this bug is independent of handset and though I've only come 
across it with SIP I think it affects all channels too.  The transfer 
method is that defined in features.conf.


Mike

Tong wrote:

Are you using a cisco 7960 with POS 8.2?

- Original Message - From: Mike Dawson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Wednesday, July 19, 2006 2:31 AM
Subject: Re: [Asterisk-Users] attended transfer issue



Thomas Artner wrote:


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking 
the call

.) the caller get lost at this point !!


I've just come across this issue too.  As the call gets hung up if the 
transfer is attempted before answer I tried changing this:


exten = _90ZX,1,Dial(zap/g1/${EXTEN},,TW)

to this:

exten = _90ZX,1,Answer
exten = _90ZX,n,Dial(zap/g1/${EXTEN},,TW)

So the call is 'answered' in one sense before it starts ringing.  I've 
only tested it on a zap channel so far but it seems to fix it.  Unless 
this is how Answer() is supposed to be used, I'm not sure then it's a 
bit of a dirty hack and I don't know what else it might break.


I'm not back in the office until next week so can't test my brainwave 
out fully.


Mike
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Re: [Asterisk-Users] attended transfer issue

[Tong]

Are you using a cisco 7960 with POS 8.2?

- Original Message - 
From: Mike Dawson [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion 
asterisk-users@lists.digium.com

Sent: Wednesday, July 19, 2006 2:31 AM
Subject: Re: [Asterisk-Users] attended transfer issue



Thomas Artner wrote:


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the 
call

.) the caller get lost at this point !!


I've just come across this issue too.  As the call gets hung up if the 
transfer is attempted before answer I tried changing this:


exten = _90ZX,1,Dial(zap/g1/${EXTEN},,TW)

to this:

exten = _90ZX,1,Answer
exten = _90ZX,n,Dial(zap/g1/${EXTEN},,TW)

So the call is 'answered' in one sense before it starts ringing.  I've 
only tested it on a zap channel so far but it seems to fix it.  Unless 
this is how Answer() is supposed to be used, I'm not sure then it's a bit 
of a dirty hack and I don't know what else it might break.


I'm not back in the office until next week so can't test my brainwave out 
fully.


Mike
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No virus found in this incoming message.
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Re: [Asterisk-Users] attended transfer issue

[Mike Dawson]

Thomas Artner wrote:


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the call
.) the caller get lost at this point !!


I've just come across this issue too.  As the call gets hung up if the 
transfer is attempted before answer I tried changing this:


exten = _90ZX,1,Dial(zap/g1/${EXTEN},,TW)

to this:

exten = _90ZX,1,Answer
exten = _90ZX,n,Dial(zap/g1/${EXTEN},,TW)

So the call is 'answered' in one sense before it starts ringing.  I've 
only tested it on a zap channel so far but it seems to fix it.  Unless 
this is how Answer() is supposed to be used, I'm not sure then it's a 
bit of a dirty hack and I don't know what else it might break.


I'm not back in the office until next week so can't test my brainwave 
out fully.


Mike
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[Asterisk-Users] Attended transfer and queue

[aston martin]

It seems that Asterisk does not free up agent after attended transfer. The agent stays in 'busy' state for as long as the conversation between the caller and person, to which call was transfered, is active.Does anyone know any workarounds for this problem? __Do You Yahoo!?Tired of spam?  Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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Re: [Asterisk-Users] Attended transfer and queue

[Matt]

I've never seen this happen, and we run almost 300 calls through
Asterisk with around 20 agents every day.   How are you performing
your attended transfer?   Step-by-step.

On 6/12/06, aston martin [EMAIL PROTECTED] wrote:


It seems that Asterisk does not free up agent after attended transfer. The
agent stays in 'busy' state for as long as the conversation between the
caller and person, to which call was transfered, is active.

Does anyone know any workarounds for this problem?


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Re: [Asterisk-Users] Attended transfer and queue

[aston martin]

1. Call comes in the queue (command Queue(...) gets executed)  2. Call reaches extension at which agent is registered (I'm using AgentCallbackLogin), where Dial(SIP/.,,tT) command gets executed  3. Agent answers the call and enters *2 (that's my default for attended transfers as set in features.conf) + a number to which the call should be transfered  4. Agent hangs up  5. Agent stays in 'busy' state  Matt [EMAIL PROTECTED] wrote:  I've never seen this happen, and we run almost 300 calls throughAsterisk with around 20 agents every day. How are you performingyour attended transfer? Step-by-step.On 6/12/06, aston martin <[EMAIL PROTECTED]>wrote: It seems that Asterisk does not free up agent after attended transfer. The agent
 stays in 'busy' state for as long as the conversation between the caller and person, to which call was transfered, is active. Does anyone know any workarounds for this problem? __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options
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Re: [Asterisk-Users] Attended transfer and queue

[BJ Weschke]

On 6/12/06, aston martin [EMAIL PROTECTED] wrote:


1. Call comes in the queue (command Queue(...) gets executed)
2. Call reaches extension at which agent is registered (I'm using
AgentCallbackLogin), where Dial(SIP/.,,tT) command gets executed
3. Agent answers the call and enters *2 (that's my default for attended
transfers as set in features.conf) + a number to which the call should be
transfered
4. Agent hangs up
5. Agent stays in 'busy' state





Is the agent's device you're dialing on the same server that
app_queue is operating on? I've seen this happen when it is not on the
same server, and there's a reason behind why it doesn't work at the
present time in that configuration.

--
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Re: [Asterisk-Users] Attended transfer and queue

[Matt]

AHHH!  We use the Xfer button on our Aastra 9133is to do transfers
for some reason (see another post I just made) when I hit * queue
calls disconnect.

On 6/12/06, aston martin [EMAIL PROTECTED] wrote:


1. Call comes in the queue (command Queue(...) gets executed)
2. Call reaches extension at which agent is registered (I'm using
AgentCallbackLogin), where Dial(SIP/.,,tT) command gets executed
3. Agent answers the call and enters *2 (that's my default for attended
transfers as set in features.conf) + a number to which the call should be
transfered
4. Agent hangs up
5. Agent stays in 'busy' state



Matt [EMAIL PROTECTED] wrote:

I've never seen this happen, and we run almost 300 calls through
Asterisk with around 20 agents every day. How are you performing
your attended transfer? Step-by-step.

On 6/12/06, aston martin wrote:

 It seems that Asterisk does not free up agent after attended transfer. The
 agent stays in 'busy' state for as long as the conversation between the
 caller and person, to which call was transfered, is active.

 Does anyone know any workarounds for this problem?


 __
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Re: [Asterisk-Users] Attended transfer and queue

[aston martin]

I think it is. Agent's device is actually a softphone, registeredas one of the clients defined in sip.conf.Would it help if I posted the configs?BJ Weschke [EMAIL PROTECTED] wrote:  On 6/12/06, aston martin <[EMAIL PROTECTED]>wrote: 1. Call comes in the queue (command Queue(...) gets executed) 2. Call reaches extension at which agent is registered (I'm using AgentCallbackLogin), where Dial(SIP/.,,tT) command gets executed 3. Agent answers the call and enters *2 (that's my default for attended transfers as set in features.conf) + a number to which the call should be transfered 4. Agent hangs up 5. Agent stays in 'busy' stateIs the agent's device you're dialing on the same server
 thatapp_queue is operating on? I've seen this happen when it is not on thesame server, and there's a reason behind why it doesn't work at thepresent time in that configuration.-- Bird's The Word Technologies, Inc.http://www.btwtech.com/___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam?  Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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Re: [Asterisk-Users] Attended transfer and queue

[BJ Weschke]

On 6/12/06, Matt [EMAIL PROTECTED] wrote:

AHHH!  We use the Xfer button on our Aastra 9133is to do transfers
for some reason (see another post I just made) when I hit * queue
calls disconnect.

On 6/12/06, aston martin [EMAIL PROTECTED] wrote:

 1. Call comes in the queue (command Queue(...) gets executed)
 2. Call reaches extension at which agent is registered (I'm using
 AgentCallbackLogin), where Dial(SIP/.,,tT) command gets executed
 3. Agent answers the call and enters *2 (that's my default for attended
 transfers as set in features.conf) + a number to which the call should be
 transfered
 4. Agent hangs up
 5. Agent stays in 'busy' state




With 1.2.X, the * was hardcoded into chan_agent to drop the call.
With /trunk and 1.4, this is now a configurable option.

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Re: [Asterisk-Users] Attended transfer and queue

[Dinesh Nair]

On 06/12/06 20:21 Matt said the following:

AHHH!  We use the Xfer button on our Aastra 9133is to do transfers
for some reason (see another post I just made) when I hit * queue
calls disconnect.


take a look at http://bugs.digium.com/view.php?id=6897 which solves this 
problem. also, since this has been committed to 1.2 and trunk, i would 
think that 1.2.9.1 would also have this patch applied.


--
Regards,   /\_/\   All dogs go to heaven.
[EMAIL PROTECTED](0 0)   http://www.openmalaysiablog.com/
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| for a in past present future; do|
|   for b in clients employers associates relatives neighbours pets; do   |
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Re: [Asterisk-Users] Attended transfer and queue

[Matt]

On 6/12/06, BJ Weschke [EMAIL PROTECTED] wrote:

On 6/12/06, Matt [EMAIL PROTECTED] wrote:
 AHHH!  We use the Xfer button on our Aastra 9133is to do transfers
 for some reason (see another post I just made) when I hit * queue
 calls disconnect.

 On 6/12/06, aston martin [EMAIL PROTECTED] wrote:
 
  1. Call comes in the queue (command Queue(...) gets executed)
  2. Call reaches extension at which agent is registered (I'm using
  AgentCallbackLogin), where Dial(SIP/.,,tT) command gets executed
  3. Agent answers the call and enters *2 (that's my default for attended
  transfers as set in features.conf) + a number to which the call should be
  transfered
  4. Agent hangs up
  5. Agent stays in 'busy' state
 
 

 With 1.2.X, the * was hardcoded into chan_agent to drop the call.
With /trunk and 1.4, this is now a configurable option.


Uhhh who's bright idea was that?
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Re: [Asterisk-Users] Attended transfer and queue

[Matt]

On 6/12/06, Dinesh Nair [EMAIL PROTECTED] wrote:



On 06/12/06 20:21 Matt said the following:
 AHHH!  We use the Xfer button on our Aastra 9133is to do transfers
 for some reason (see another post I just made) when I hit * queue
 calls disconnect.

take a look at http://bugs.digium.com/view.php?id=6897 which solves this
problem. also, since this has been committed to 1.2 and trunk, i would
think that 1.2.9.1 would also have this patch applied.


Ahhh I always love a good upgrade :P
I'd just as soon change the transfer to #2, however my aastra phones
don't seem to let me use #270 in my softkeys.. as soon as it hits # it
just dies and stoppes... anyone know a way around this?
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Re: [Asterisk-Users] Attended transfer and queue

[aston martin]

Are there any patches that would fix the agent staying in 'busy' state problem as well? I couldn't find any so far..Dinesh Nair [EMAIL PROTECTED] wrote:  On 06/12/06 20:21 Matt said the following: AHHH! We use the Xfer button on our Aastra 9133is to do transfers for some reason (see another post I just made) when I hit * queue calls disconnect.take a look at http://bugs.digium.com/view.php?id=6897 which solves this problem. also, since this has been committed to 1.2 and trunk, i would think that 1.2.9.1 would also have this patch applied.-- Regards, /\_/\ "All dogs go to heaven."[EMAIL PROTECTED] (0 0) http://www.openmalaysiablog.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 |+=+___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit:http://lists.digium.com/mailman/listinfo/asterisk-users __Do You Yahoo!?Tired of spam?  Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___
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Re: [Asterisk-Users] Attended transfer and queue

[BJ Weschke]

On 6/12/06, aston martin [EMAIL PROTECTED] wrote:

Are there any patches that would fix the agent staying in 'busy' state
problem as well? I couldn't find any so far..



That would depend on what's causing this to happen. If you open up a
bug at bugs.digium.com with the configs and trace/logs, we'll take a
look at what's going on.

BJ

--
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Re: [Asterisk-Users] Attended transfer and queue

[Matt]

What version of Asterisk are you running, that you are able to dial *2
and the * isn't hanging up like it is for me?

On 6/12/06, aston martin [EMAIL PROTECTED] wrote:

Are there any patches that would fix the agent staying in 'busy' state
problem as well? I couldn't find any so far..


Dinesh Nair [EMAIL PROTECTED] wrote:


On 06/12/06 20:21 Matt said the following:
 AHHH! We use the Xfer button on our Aastra 9133is to do transfers
 for some reason (see another post I just made) when I hit * queue
 calls disconnect.

take a look at http://bugs.digium.com/view.php?id=6897
which solves this
problem. also, since this has been committed to 1.2 and trunk, i would
think that 1.2.9.1 would also have this patch applied.

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Re: [Asterisk-Users] Attended transfer and queue

[aston martin]

I'm using 1.2.9.1, and it isn't hanging up... (at leastTHAT seems to be okay, hehe). I wonder how nobody else came across this before, cause I'm not using queues pretty much by the bookthe only thing is that agents stay busy after transfer.Here are part of my configs, if somebody gets any ideas, what could be causing that:That's the sip client (defined in sip.conf):[Agent001]username=Agent001secret=Agent001type=friendhost=dynamiccontext=from-sipdisallow=allallow=alawallow=ulaw-  That's the extension with the queue:exten = _995,1,Answer()exten = _995,2,LookupBlacklist(j)exten = _995,3,Set(MONITOR_FILENAME=${CALLERIDNUM}_${UNIQUEID}_${EXTEN}_wav128)exten
 = _995,4,Queue(MainQueue|tT|||14400)-  That's the extension where agent gets actually called:exten = _0XXX.,1,Dial(SIP/Agent${EXTEN:1:3},8,tTj)  And that's pretty much all.  Matt [EMAIL PROTECTED] wrote:  What version of Asterisk are you running, that you are able to dial *2and the * isn't hanging up like it is for me?On 6/12/06, aston martin <[EMAIL PROTECTED]>wrote: Are there any patches that would fix the agent staying in 'busy' state problem as well? I couldn't find any so far.. Dinesh Nair <[EMAIL PROTECTED]>wrote: On 06/12/06
 20:21 Matt said the following:  AHHH! We use the Xfer button on our Aastra 9133is to do transfers  for some reason (see another post I just made) when I hit * queue  calls disconnect. take a look at http://bugs.digium.com/view.php?id=6897 which solves this problem. also, since this has been committed to 1.2 and trunk, i would think that 1.2.9.1 would also have this patch applied. -- Regards, /\_/\ "All dogs go to heaven." [EMAIL PROTECTED] (0 0) http://www.openmalaysiablog.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] Attended transfer and queue

[Dinesh Nair]

On 06/12/06 20:42 Dinesh Nair said the following:



i would 
think that 1.2.9.1 would also have this patch applied.


not it doesnt. my patch was only committed for trunk, though mantis does 
have the patch that works on 1.2.x as well.


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Re: [Asterisk-Users] Attended transfer and queue

[Dinesh Nair]

On 06/12/06 21:11 Matt said the following:

What version of Asterisk are you running, that you are able to dial *2
and the * isn't hanging up like it is for me?


because i wrote and applied the patch ? :)

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Re: [Asterisk-Users] attended transfer issue

[Olivier Krief]

2006/4/20, John Novack (port) [EMAIL PROTECTED]:
There should be no need forTWO feature codes.I fully second that : what matters most is to satisfy users.Unified transfer method offer :- simplicity,- hardware independance (think about mobile phones, or people occasionnaly using foreign language configured phones when visiting a sister company abroad)
- and above all, it keeps calls from being lost.So it should be implemented in Asterisk and it's up to Polycom, Snom and others to design phones that at least do not prevent people to use # sign based unified transfer method if they wish to.
For the sake of behaviour consistency, maybe :- this unified transfer method (let's say U for unified)  should be introduced in features.conf independently of previous t or T methods and it's up developpers to reuse, factorize or rewrite existing transfer code and as long as those 3 methods as supported,
- and previous t or T methods should be droped sometimes later on to simplify code support.Cheers
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Re: [Asterisk-Users] attended transfer issue

[Don Pobanz]

Olivier Krief wrote:

For the sake of behaviour consistency, maybe :
- this unified transfer method (let's say U for unified) should be 
introduced in features.conf independently of previous t or T methods and 
it's up developpers to reuse, factorize or rewrite existing transfer 
code and as long as those 3 methods as supported,


- and previous t or T methods should be droped sometimes later on to 
simplify code support.




the t or T is used to determine whether someone can transfer a call, be 
it blink or attended, not HOW that transfer occurs. So the 'tT' 
discussion is completely separate from the how discussion.


Don Pobanz
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Re: [Asterisk-Users] attended transfer issue

[John Novack]

Olivier Krief wrote:


snip
So it should be implemented in Asterisk and it's up to Polycom, Snom 
and others to design phones that at least do not prevent people to use 
# sign based unified transfer method if they wish to.


I would HOPE that either the transfer key could be reprogrammed or the 
transfer function in Asterisk could be changed to match one another, and 
it also work with  other than SIP phones.


John Novack



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Re: [Asterisk-Users] attended transfer issue

[Eric \ManxPower\ Wieling]

John Novack wrote:



Eric ManxPower Wieling wrote:


John Novack wrote:




Damon Estep wrote:

There is some kind of issue with SIP transfer interaction between 
some SIP phones and asterisk, I have personal experience with 
Polycom phones not being able to do a blind xfer using the feature key.




Our receptionist does both blind and attended transfers with Polycoms 
all the time.  In later versions of the Polycom SIP firmware: Attended 
transfer: while on a call press the Transfer key, dial the 
destination, talk, then press the transfer key again.  For Blind: 
while on call, press transfer key, then the Blind key, dial, hangup.


All well and good, but that is awkward.
The point is/was that doing a transfer, talk or not, transferee answer 
or not, the transferer should be able to hang up and the call NOT get 
lost, the transfer complete and ring then overflow to voice mail.

Easier on ALL users and the way hybrid systems have worked for many a year


This is just the way Polycom does transfers.  Older versions of the 
Polycom firmware worked as you describe.  I think it was in 1.6.x where 
they changed things to separate blind and attended transfers.  It really 
has nothing to do with Asterisk at all.

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RE: [Asterisk-Users] attended transfer issue

[Damon Estep]

 



From: [EMAIL PROTECTED] on behalf of Eric ManxPower Wieling
Sent: Wed 4/19/2006 8:47 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] attended transfer issue



John Novack wrote:


 Eric ManxPower Wieling wrote:

 John Novack wrote:



 Damon Estep wrote:

 There is some kind of issue with SIP transfer interaction between
 some SIP phones and asterisk, I have personal experience with
 Polycom phones not being able to do a blind xfer using the feature key.


 Our receptionist does both blind and attended transfers with Polycoms
 all the time.  In later versions of the Polycom SIP firmware: Attended
 transfer: while on a call press the Transfer key, dial the
 destination, talk, then press the transfer key again.  For Blind:
 while on call, press transfer key, then the Blind key, dial, hangup.

 All well and good, but that is awkward.
 The point is/was that doing a transfer, talk or not, transferee answer
 or not, the transferer should be able to hang up and the call NOT get
 lost, the transfer complete and ring then overflow to voice mail.
 Easier on ALL users and the way hybrid systems have worked for many a year

This is just the way Polycom does transfers.  Older versions of the
Polycom firmware worked as you describe.  I think it was in 1.6.x where
they changed things to separate blind and attended transfers.  It really
has nothing to do with Asterisk at all.


Understand, but the point was that every SIP device has its own method, and it 
would be nice if asterisk had a blind/attend transfer feature as described so 
we are not dependent on the SIP UA vendors to try and normalize the world. If 
asterisk had the featuer we would not care that every SIP phone does it 
differently ,there would be one sure method that users could learn that wouild 
be consistent from device to device.

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Re: [Asterisk-Users] attended transfer issue

[Andrew Kohlsmith]

On Wednesday 19 April 2006 13:32, Damon Estep wrote:
 Understand, but the point was that every SIP device has its own method, and
 it would be nice if asterisk had a blind/attend transfer feature as
 described so we are not dependent on the SIP UA vendors to try and
 normalize the world. If asterisk had the featuer we would not care that
 every SIP phone does it differently ,there would be one sure method that
 users could learn that wouild be consistent from device to device.

We do have that, it's called the t and T options for app_dial(), and using 
#  :-)

-A.
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RE: [Asterisk-Users] attended transfer issue

[Damon Estep]

 -Original Message-
 From: [EMAIL PROTECTED] [mailto:asterisk-users-
 [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith
 Sent: Wednesday, April 19, 2006 11:53 AM
 To: asterisk-users@lists.digium.com
 Subject: Re: [Asterisk-Users] attended transfer issue
 
 On Wednesday 19 April 2006 13:32, Damon Estep wrote:
  Understand, but the point was that every SIP device has its own
method,
 and
  it would be nice if asterisk had a blind/attend transfer feature as
  described so we are not dependent on the SIP UA vendors to try and
  normalize the world. If asterisk had the featuer we would not care
that
  every SIP phone does it differently ,there would be one sure method
that
  users could learn that wouild be consistent from device to device.
 
 We do have that, it's called the t and T options for app_dial(),
and
 using
 #  :-)
 
Is the current release different than what I am running, # transfer on
my systems are all blind, no attended option.
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Re: [Asterisk-Users] attended transfer issue

[Eric \ManxPower\ Wieling]

Damon Estep wrote:


Is the current release different than what I am running, # transfer on
my systems are all blind, no attended option.


1.0.x only supported blind DTMF transfer hack.  1.2.x supports both 
blind and supervised DTMF transfer hacks.  See features.conf in 1.2.x

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Re: [Asterisk-Users] attended transfer issue

[John Novack (port)]

Eric ManxPower Wieling wrote:


Damon Estep wrote:


Is the current release different than what I am running, # transfer on
my systems are all blind, no attended option.



1.0.x only supported blind DTMF transfer hack.  1.2.x supports both 
blind and supervised DTMF transfer hacks.  See features.conf in 1.2.x


Jf you read back through the thread, Damon's ( and my ) point is, 
however, that ONE function code should BEGIN as an attended transfer, 
then if the transferer so chooses, have it turn into a blind transfer 
wit the simple act of hanging up ( in many systems a release )
Currently it seems that an attended transfer  that goes unanswered 
results in a lost or dropped call

There should be no need for  TWO feature codes.

John Novack


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Re: RES: [Asterisk-Users] attended transfer issue

[Kevin Bockman]

dovb wrote:

That fix would be great!!!

To press # and be able to get the call back and terminate the transfer...
I had to implement an horrible workaround to emulate this functionality


Well, as it stands now, to hangup while you are doing a transfer, you 
using the hangup feature code (in features.conf).  That will put you 
back to the original person.



Kevin
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Re: [Asterisk-Users] attended transfer issue

[Eric \ManxPower\ Wieling]

John Novack wrote:



Damon Estep wrote:

There is some kind of issue with SIP transfer interaction between some 
SIP phones and asterisk, I have personal experience with Polycom 
phones not being able to do a blind xfer using the feature key.


Our receptionist does both blind and attended transfers with Polycoms 
all the time.  In later versions of the Polycom SIP firmware: Attended 
transfer: while on a call press the Transfer key, dial the destination, 
talk, then press the transfer key again.  For Blind: while on call, 
press transfer key, then the Blind key, dial, hangup.

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Re: [Asterisk-Users] attended transfer issue

[John Novack]

Eric ManxPower Wieling wrote:


John Novack wrote:




Damon Estep wrote:

There is some kind of issue with SIP transfer interaction between 
some SIP phones and asterisk, I have personal experience with 
Polycom phones not being able to do a blind xfer using the feature key.




Our receptionist does both blind and attended transfers with Polycoms 
all the time.  In later versions of the Polycom SIP firmware: Attended 
transfer: while on a call press the Transfer key, dial the 
destination, talk, then press the transfer key again.  For Blind: 
while on call, press transfer key, then the Blind key, dial, hangup.


All well and good, but that is awkward.
The point is/was that doing a transfer, talk or not, transferee answer 
or not, the transferer should be able to hang up and the call NOT get 
lost, the transfer complete and ring then overflow to voice mail.

Easier on ALL users and the way hybrid systems have worked for many a year

John Novack


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Re: [Asterisk-Users] attended transfer issue

[Eric \ManxPower\ Wieling]

Damon Estep wrote:

There is some kind of issue with SIP transfer interaction between some
SIP phones and asterisk, I have personal experience with Polycom phones
not being able to do a blind xfer using the feature key.

We have to use the asterisk # blind xfrer functionality for blind
transfers

The phones will drop the call if you initiate a transfer with the
feature key but do not wait for the remote line to answer before
releasing the call. In other words, if you hit transfer on the phone,
wait for the remote phone to ring, and hang up, you will drop the call.

If you wait for the remote phone to answer (live or voicemail) the
transfer will complete.


My users also reported this.  It turns out Polycom added a Blind soft 
key to continue with a blind transfer.  If you don't select that then a 
blind transfer will fail.

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Re: [Asterisk-Users] attended transfer issue

[Thomas Artner]

Hi!

I decided to open an issue about this case in the mantis database!
I am not very familiar with the bug/issue tracking procedure at the
asterisk project, but I think i can make it.

Is there something that would speak against it?

cheers,
tom




Thomas Artner wrote:
 Hi!
 
 
 A few months ago I needed some help for the following issue:
 
 .) a call comes in
 .) Person A takes the call and does an attended transfer to Person B
 .) Person A hangs up the phone without waiting for Person B taking the call
 .) the caller get lost at this point !!
 
 At this point the attended transfer should go into a blind transfer. The
 phone of Person B should still be ringing and the caller shouldnt get lost.
 
 I think this is the most usual behaviour of a call transfer also on the
 cheapest systems on the market.
 
 Why doesnt this work well with asterisk? Will there be a solution for
 that in the near future?
 
 I am thankful for any kind of help!
 
 
 thx,
 Tom
 
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Re: [Asterisk-Users] attended transfer issue

[John Novack]

Thank you for doing that.

With a little luck someone will be able to fix it, assuming they 
understand it NEEDS to be fixed.


Damon Estep stated proper transfer operation well yesterday
Sustitution of another key for pound in features.conf might also be 
desirable


To be typical it would act like this;

Press pound to get secondary dial tone
Dial the number for the transfer
Either hang up or stay on the line after progress (ring)
If you stay on the line the transfer completes when you hang up
If you hang up during the ring the call is blind transferred
If you press the same feature access key (#) again you get the call back
and terminate the transfer.


John Novack
Thomas Artner wrote:


Hi!

I decided to open an issue about this case in the mantis database!
I am not very familiar with the bug/issue tracking procedure at the
asterisk project, but I think i can make it.

Is there something that would speak against it?

cheers,
tom




Thomas Artner wrote:
 


Hi!


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the call
.) the caller get lost at this point !!

At this point the attended transfer should go into a blind transfer. The
phone of Person B should still be ringing and the caller shouldnt get lost.

I think this is the most usual behaviour of a call transfer also on the
cheapest systems on the market.

Why doesnt this work well with asterisk? Will there be a solution for
that in the near future?

I am thankful for any kind of help!


thx,
Tom

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RES: [Asterisk-Users] attended transfer issue

[dovb]

That fix would be great!!!

To press # and be able to get the call back and terminate the transfer...
I had to implement an horrible workaround to emulate this functionality

Dov 

-Mensagem original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Em nome de John Novack
Enviada em: sábado, 15 de abril de 2006 13:19
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Assunto: Re: [Asterisk-Users] attended transfer issue

Thank you for doing that.

With a little luck someone will be able to fix it, assuming they understand
it NEEDS to be fixed.

Damon Estep stated proper transfer operation well yesterday Sustitution of
another key for pound in features.conf might also be desirable

To be typical it would act like this;

Press pound to get secondary dial tone
Dial the number for the transfer
Either hang up or stay on the line after progress (ring) If you stay on the
line the transfer completes when you hang up If you hang up during the ring
the call is blind transferred If you press the same feature access key (#)
again you get the call back and terminate the transfer.


John Novack
Thomas Artner wrote:

Hi!

I decided to open an issue about this case in the mantis database!
I am not very familiar with the bug/issue tracking procedure at the 
asterisk project, but I think i can make it.

Is there something that would speak against it?

cheers,
tom




Thomas Artner wrote:
  

Hi!


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the 
call
.) the caller get lost at this point !!

At this point the attended transfer should go into a blind transfer. 
The phone of Person B should still be ringing and the caller shouldnt get
lost.

I think this is the most usual behaviour of a call transfer also on 
the cheapest systems on the market.

Why doesnt this work well with asterisk? Will there be a solution for 
that in the near future?

I am thankful for any kind of help!


thx,
Tom

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Re: [Asterisk-Users] attended transfer issue

[Thomas Artner]

here's the reported issue: http://bugs.digium.com/view.php?id=6973


cheers,
tom


Thomas Artner wrote:
 Hi!
 
 I decided to open an issue about this case in the mantis database!
 I am not very familiar with the bug/issue tracking procedure at the
 asterisk project, but I think i can make it.
 
 Is there something that would speak against it?
 
 cheers,
 tom
 
 
 
 
 Thomas Artner wrote:
 Hi!


 A few months ago I needed some help for the following issue:

 .) a call comes in
 .) Person A takes the call and does an attended transfer to Person B
 .) Person A hangs up the phone without waiting for Person B taking the call
 .) the caller get lost at this point !!

 At this point the attended transfer should go into a blind transfer. The
 phone of Person B should still be ringing and the caller shouldnt get lost.

 I think this is the most usual behaviour of a call transfer also on the
 cheapest systems on the market.

 Why doesnt this work well with asterisk? Will there be a solution for
 that in the near future?

 I am thankful for any kind of help!


 thx,
 Tom

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[Asterisk-Users] attended transfer issue

[Thomas Artner]

Hi!


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the call
.) the caller get lost at this point !!

At this point the attended transfer should go into a blind transfer. The
phone of Person B should still be ringing and the caller shouldnt get lost.

I think this is the most usual behaviour of a call transfer also on the
cheapest systems on the market.

Why doesnt this work well with asterisk? Will there be a solution for
that in the near future?

I am thankful for any kind of help!


thx,
Tom

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RE: [Asterisk-Users] attended transfer issue

[Michael Collins]

 A few months ago I needed some help for the following issue:
 
 .) a call comes in
 .) Person A takes the call and does an attended transfer to Person B
 .) Person A hangs up the phone without waiting for Person B taking the
 call
 .) the caller get lost at this point !!
 
 At this point the attended transfer should go into a blind transfer.
The
 phone of Person B should still be ringing and the caller shouldnt get
 lost.
 
 I think this is the most usual behaviour of a call transfer also on
the
 cheapest systems on the market.


Could you remind us of what kinds of phones you are using, and whether
you're using SIP, Zap or something else?

Thanks!

-MC
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Re: [Asterisk-Users] attended transfer issue

[John Novack]

Michael Collins wrote:


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B taking the call
.) the caller get lost at this point !!

At this point the attended transfer should go into a blind transfer.
   


The phone of Person B should still be ringing and the caller shouldnt get lost.

I think this is the most usual behaviour of a call transfer also on the 
cheapest systems on the market.
 




Could you remind us of what kinds of phones you are using, and whether you're 
using SIP, Zap or something else?

Thanks!

-MC

I think the point of this post and other related ones is the fact that 
there are attended and blind transfers, initiated by different actions, 
where phone systems for at least the last 20 years have one action, or 
transfer.
The person initiating the transfer starts the procedure, and if the 
destination extension answers, either through the facilities of 
handsfree intercom or picking up the phone, the initiator and the 
receiver can confer BEFORE the transfer is complete.
If, on the other hand the initiator either chooses to hang up after 
starting the transfer, the transfer is then complete, and the 
destination extension rings until answered or overflows into voice mail.
In NO case should the call get lost. Attended and blind transfer SHOULD 
start with the same action and be considered as ONE function

Irrelevant what phones are being used.

JMO

John Novack

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Re: [Asterisk-Users] attended transfer issue

[Jerry Jones]

Yes it should all behave the way we are used to. However SIP IS  
different. The exact behavior will be dependant upon the individual  
hard phone.


This of course is if using SIP which we do not know yet...

On Apr 14, 2006, at 1:43 PM, John Novack wrote:




Michael Collins wrote:


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B  
taking the call

.) the caller get lost at this point !!

At this point the attended transfer should go into a blind transfer.

The phone of Person B should still be ringing and the caller  
shouldnt get lost.


I think this is the most usual behaviour of a call transfer also  
on the cheapest systems on the market.




Could you remind us of what kinds of phones you are using, and  
whether you're using SIP, Zap or something else?


Thanks!

-MC

I think the point of this post and other related ones is the fact  
that there are attended and blind transfers, initiated by different  
actions, where phone systems for at least the last 20 years have  
one action, or transfer.
The person initiating the transfer starts the procedure, and if the  
destination extension answers, either through the facilities of  
handsfree intercom or picking up the phone, the initiator and the  
receiver can confer BEFORE the transfer is complete.
If, on the other hand the initiator either chooses to hang up after  
starting the transfer, the transfer is then complete, and the  
destination extension rings until answered or overflows into voice  
mail.
In NO case should the call get lost. Attended and blind transfer  
SHOULD start with the same action and be considered as ONE function

Irrelevant what phones are being used.

JMO

John Novack

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Re: [Asterisk-Users] attended transfer issue

[John Novack]

Jerry Jones wrote:

Yes it should all behave the way we are used to. However SIP IS  
different. The exact behavior will be dependant upon the individual  
hard phone.



Isn't that true only if it has a preprogrammed transfer key?
an Asterisk feature code should work as discussed.
There SHOULD be a way to make SIP phones work the same.
( easy to say, perhaps not so easy to do )

John Novack


This of course is if using SIP which we do not know yet...

On Apr 14, 2006, at 1:43 PM, John Novack wrote:




Michael Collins wrote:


A few months ago I needed some help for the following issue:

.) a call comes in
.) Person A takes the call and does an attended transfer to Person B
.) Person A hangs up the phone without waiting for Person B  taking 
the call

.) the caller get lost at this point !!

At this point the attended transfer should go into a blind transfer.

The phone of Person B should still be ringing and the caller  
shouldnt get lost.


I think this is the most usual behaviour of a call transfer also  on 
the cheapest systems on the market.




Could you remind us of what kinds of phones you are using, and  
whether you're using SIP, Zap or something else?


Thanks!

-MC

I think the point of this post and other related ones is the fact  
that there are attended and blind transfers, initiated by different  
actions, where phone systems for at least the last 20 years have  one 
action, or transfer.
The person initiating the transfer starts the procedure, and if the  
destination extension answers, either through the facilities of  
handsfree intercom or picking up the phone, the initiator and the  
receiver can confer BEFORE the transfer is complete.
If, on the other hand the initiator either chooses to hang up after  
starting the transfer, the transfer is then complete, and the  
destination extension rings until answered or overflows into voice  
mail.
In NO case should the call get lost. Attended and blind transfer  
SHOULD start with the same action and be considered as ONE function

Irrelevant what phones are being used.

JMO

John Novack

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