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[anatoliy at kounitskiy...]

Hello everybody!
I have few questions about the att_xfer application. First, what i want
to accomplish is: user A calls user B, after that user B makes attended
transfer to user C.
In the dialplan i have:

<context name="vpbx">
<extension name="local_number">
...
<action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>
<action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>
<action application="bind_meta_app" data="4 b s
execute_extension::attented_xfer XML features"/>
....
</condition>
</extension>

So when user B answers the call, he sends *4 and the extensions for the
attended transfer is started - the usual - plays message and read the
input dtmf:

features.xml
...
<extension name="attented_xfer">
<condition field="${toll_allow}" expression="local"/>
<condition field="destination_number" expression="^attented_xfer$">
<action application="info"/>
<action application="read" data="3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 30000 #"/>
<action application="set" data="call_timeout=15"/>
<action application="att_xfer"
data="user/${attxfer_callthis}@${domain_name}"/>
</condition>
</extension>
...

To this problems everything is perfect. But here comes the questions, so
if you can give some tips would be great.

1) when user B enters the extension number of C - the C's phone starts
ringing in the tcpdump i can see that the phone is sending 180 ringing,
BUT user B does not hear the ringing.
2) as mentioned in the
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
quote: "If the other leg is a voicemail or doesn't answered you can
hangup that leg by pressing dtmf # (fixed in r14438) "
It doesn't work. The option 0 is working even before C answering the
phone - after he answers it's a threeway conference - i like this
feature.

I'm using FreeSWITCH Version 1.0.trunk (14633M)

Also I tried to set call timeout to see if I can go back the user A, who
is listening to MOH - no luck here.

Probably I'm missing something. Tried to look in the source of att_xfer
to understand why the feature i want is not working - but it seems my
C/C++ skills are not so good, as i want .

Thank you in advance,
Anatoliy Kounitskiy

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[anatoliy at kounitskiy...]

After several hours of testing I was able to answer myself the
previous mentioned questions.

It appears that # and the 0 option work _only_ if user C has answered
the call OR voicemail system answers it.

user A ---call---> user B----attended xfer---> user C

At this point I have new question. In example user C does not have a
voicemail and the call timeout is not an option to wait for. How can
user B go back to the user A, who is listening to MOH?
Could someone help me with an advice/tip?

At the moment I have just one idea for accomplishing it:
1) try to use bind_meta_app in the extension with the att_xfer (not
sure if it can be done). To have a key feature that takes the user A
call leg id and bridging it with user B

Thank you in advnace,
Anatoliy Kounitskiy


On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy
Kounitskiy<anatoliy@kounitskiy.com> wrote:

[anthony.minessale at g...]

maybe we can make an origination_cancel_key=# you could set on the dial string to be able to cancel that originate with dtmf


On Wed, Aug 26, 2009 at 1:27 PM, Anatoliy Kounitskiy <anatoliy@kounitskiy.com (anatoliy@kounitskiy.com)> wrote:

[anthony.minessale at g...]

i added a patch to attempt to do this so try adding

{origination_cancel_key=#} before the dial string
or
<action application="set" data="origination_cancel_key=#"/>
before you bridge.

On Wed, Aug 26, 2009 at 1:39 PM, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:

[msc at freeswitch.org]

On Wed, Aug 26, 2009 at 12:59 PM, Anthony Minessale <anthony.minessale@gmail.com (anthony.minessale@gmail.com)> wrote:

[anatoliy at kounitskiy...]

Thank you for the fast patch :)

I'm going to test it right away and report you if there are problems or not.


On Wed, Aug 26, 2009 at 10:59 PM, Anthony
Minessale<anthony.minessale@gmail.com> wrote:

[anatoliy at kounitskiy...]

Just for information - the idea :
A---calls---> B ---att_xfer---> C , B hangs C up and goes back to A
( I`m sorry C is not answering )

dialplan:
-----------
<extension name="local_number">
<condition field="${toll_allow}" expression="local"/>
<condition field="destination_number" expression="^(\d{3})$">
<action application="set" data="dialed_extension=$1"/>
<action application="export" data="dialed_extension=$1"/>
<action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>
<action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
<action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>
<action application="bind_meta_app" data="4 b s
execute_extension::attented_xfer XML features"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="set" data="call_timeout=10"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="set"
data="continue_on_fail=USER_NOT_REGISTERED,SUBSCRIBER_ABSENT"/>
<action application="info"/>
<action application="set"
data="bringback=${user_data(${dialed_extension}@${domain_name} var
cringback )}"/>
<action application="set" data="ringback=${${bringback}}"/>
<action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>
<action application="info"/>
...
</condition>
</extension>

features.xml
---------------------
<extension name="attented_xfer">
<condition field="${toll_allow}" expression="local"/>
<condition field="destination_number" expression="^attented_xfer$">
<action application="set" data="continue_on_fail=true"/>
<action application="read" data="3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 30000 #"/>
<action application="set"
data="bringback=${user_data(${attxfer_callthis}@${domain_name} var
cringback )}"/>
<action application="set" data="ringback=${${bringback}}"/>
<action application="info"/>
<action application="set" data="origination_cancel_key=#"/>
<action application="att_xfer"
data="user/${attxfer_callthis}@${domain_name}"/>
</condition>
</extension>

After testing the patch the results are:

1) B answers the call from A and executes the feature code *4
(attented_xfer), sends the desired number ( in this case the number of
C). A goes to MusicOnHold, B is waiting for C to pick up - B sends # -
sends SIP CANCEL to C and SIP BUY to A - so all calls are dropped.

2) B answers the call from A and executes the feature code *4
(attented_xfer), but without entering the number of C - so the read
aplication timeouts after 30 sec - after that att_xfer is executed
with empty string (nothing is entered) - B and A are bridged together.

On the console without entering the extension of C:
2009-08-27 16:00:58.929756 [NOTICE] switch_core_session.c:1576 Execute
set(origination_cancel_key=#)
EXECUTE sofia/internal/sip:102@10.17.4.107:5060 set(origination_cancel_key=#)
2009-08-27 16:00:58.929756 [DEBUG] mod_dptools.c:748
sofia/internal/sip:userB@10.17.4.107:5060 SET
[origination_cancel_key]=[#]
2009-08-27 16:00:58.929756 [NOTICE] switch_core_session.c:1576 Execute
att_xfer(user/${attxfer_callthis}@${domain_name})
EXECUTE sofia/internal/sip:102@10.17.4.107:5060 att_xfer(user/@1.1.1.1)
2009-08-27 16:00:58.929756 [WARNING] mod_dptools.c:2373 Can't find
user [@1.1.1.1]
2009-08-27 16:00:58.929756 [ERR] switch_ivr_originate.c:1527 Cannot
create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2009-08-27 16:00:58.929756 [DEBUG] switch_ivr_originate.c:2167
Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT]
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_play_say.c:1402 done playing file
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:231
sofia/internal/userA@1.1.1.1 receive message [BRIDGE]
2009-08-27 16:00:58.949842 [DEBUG] switch_core_session.c:630 Send
signal sofia/internal/userA@1.1.1.1 [BREAK]
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:233 Send signal
sofia/internal/sip:userB@10.17.4.107:5060 [BREAK]
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:231
sofia/internal/sip:userB@10.17.4.107:5060 receive message [BRIDGE]
2009-08-27 16:00:58.949842 [DEBUG] switch_core_session.c:630 Send
signal sofia/internal/sip:userB@10.17.4.107:5060 [BREAK]
2009-08-27 16:00:58.954052 [DEBUG] switch_ivr_bridge.c:233 Send signal
sofia/internal/userA@1.1.1.1 [BREAK]
2009-08-27 16:01:22.881501 [DEBUG] sofia.c:3302 Channel
sofia/internal/userA@1.1.1.1 entering state [calling][0]
2009-08-27 16:01:22.886122 [DEBUG] sofia.c:3302 Channel
sofia/internal/userA@1.1.1.1 entering state [ready][200]

I'll try to summarize my idea - could it be possible when the # (in
the att_xfer) is executed to behave as if you're trying to make
attended transfer to non existing subscriber( SUBSCRIBER_ABSENT ), so
B can be bridged back with A.

Thank you in advance,
Anatoliy Kounitskiy

On Wed, Aug 26, 2009 at 11:11 PM, Michael Collins<msc@freeswitch.org> wrote:

[anthony.minessale at g...]

please retest with latest SVN trunk

On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy <anatoliy@kounitskiy.com (anatoliy@kounitskiy.com)> wrote:

[brian at freeswitch.org]

/me points to the first question on the FAQ

/b

On Aug 27, 2009, at 1:18 PM, Anatoliy Kounitskiy wrote:

[anatoliy at kounitskiy...]

Thank you all for the great job
Now it works as I wanted!!

Tomorrow I'll will try to update the wiki :)

But with few words ( I tried to strip the extensions to minimum for
more easy read for users).
In the dialplan I have created the extension showed below:

<extension name="local_number">
<condition field="destination_number" expression="^(\d{3})$">
<action application="set" data="dialed_extension=$1"/>
<action application="export" data="dialed_extension=$1"/>
<action application="bind_meta_app" data="1 b s
execute_extension::attented_xfer XML features"/>
<action application="set" data="transfer_ringback=$${hold_music}"/>
<action application="set" data="call_timeout=10"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>
</condition>
</extension>

In the features.xml file I have extension attented_xfer, that is using
the att_xfer application :

<extension name="attented_xfer">
<condition field="destination_number" expression="^attented_xfer$">
<action application="set" data="continue_on_fail=true"/>
<action application="read" data="3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 30000 #"/>
<action application="set" data="origination_cancel_key=#"/>
<action application="att_xfer"
data="user/${attxfer_callthis}@${domain_name}"/>
</condition>
</extension>

Also I have set the the origination cancel key will be # - so if B
wants to cancel the call to C and to go back to A - just uses the #
key :)

Again - thank you very much :)

Anatoliy Kounitskiy

On Thu, Aug 27, 2009 at 7:48 PM, Anthony
Minessale<anthony.minessale@gmail.com> wrote:

[msc at freeswitch.org]

Thanks! I'll be in touch to help you if you need any wiki editing assistance.
-MC

On Thu, Aug 27, 2009 at 11:18 AM, Anatoliy Kounitskiy <anatoliy@kounitskiy.com (anatoliy@kounitskiy.com)> wrote:

[anatoliy at kounitskiy...]

The wiki page is updated (
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer ). Hope
it is understandable :)

Anatoliy Kounitskiy

On Thu, Aug 27, 2009 at 9:30 PM, Michael Collins<msc@freeswitch.org> wrote:

[msc at freeswitch.org]

Thanks! I made very minor changes. We appreciate you helping out with documenting new features.
-MC

On Fri, Aug 28, 2009 at 4:13 AM, Anatoliy Kounitskiy <anatoliy@kounitskiy.com (anatoliy@kounitskiy.com)> wrote: