[Freeswitch-users] Can't quite get call screening to work

Gilad Abada ga at steadfasttelecom.com
Fri Dec 23 21:12:13 MSK 2011


Thanks got it working and updated thre wiki

On Fri, Dec 23, 2011 at 1:08 PM, Michael Collins <msc at freeswitch.org> wrote:
> try setting this var to true prior to the bridge:
> http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge
>
>
> On Thu, Dec 22, 2011 at 2:09 PM, Gilad Abada <ga at steadfasttelecom.com>
> wrote:
>>
>> Hi
>> I am trying to get call screening to work too.
>>
>> http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_13:_Call_Screening
>>
>> The issue I am having is that if the called device hangs up even after
>> a full conversation (accepting the call by pressing 1) the calling
>> party gets sent to voicemail.
>>
>> Seems like being sent to voicemail should be an anti-action?
>> Also the recording says to reject the call press 2 or send it to VM
>> press 3. These are not defined in the dial plan and I am not sure how
>> to do that.
>>
>> Thanks in advance!
>> Gill
>>
>> On Wed, Dec 7, 2011 at 5:51 PM, Michael Collins <msc at freeswitch.org>
>> wrote:
>> > it looks like the person who posted that example did not post their
>> > sample
>> > phrase macro file. However, do a git pull...
>> >
>> > commit 9ea3ce666fa7f021b5c2a7e2fbe153eb351c5734
>> > Author: Michael S Collins <msc at freeswitch.org>
>> > Date:   Wed Dec 7 14:49:16 2011 -0800
>> >
>> >     config: add screen_confirm macro to lang/en/ivr/sounds.xml
>> >
>> > snag that config file and drop it into conf/lang/en/ivr/ and then
>> > reloadxml.
>> > I did this on the fly w/o testing so be sure to test it thoroughly to
>> > make
>> > sure it works! Also, be sure to use the full path name to the sound file
>> > that you are playing back.
>> >
>> > -MC
>> >
>> > On Wed, Dec 7, 2011 at 12:49 PM, Phil Quesinberry
>> > <philq at qsystemsengineering.com> wrote:
>> >>
>> >> I'm trying to use the call screening example in the wiki and can't get
>> >> FS
>> >> to
>> >> play the caller's name back to the destination extension.
>> >> example here:
>> >>
>> >>
>> >> http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_13:_Call_Scr
>> >> eening
>> >>
>> >> The initial announcement asking for the caller's name works fine:
>> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>> >> phrase(voicemail_record_name)
>> >> 2011-12-07 14:52:13.715494 [DEBUG] mod_dptools.c:2362 Execute
>> >> voicemail_record_name() lang
>> >> 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:67 No language
>> >> specified - Using [en]
>> >> 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:244 Handle
>> >> play-file:[voicemail/vm-record_name1.wav] (en:en)
>> >> 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:1302 Codec
>> >> Activated L16 at 8000hz 1 channels 20ms
>> >> 2011-12-07 14:52:18.695891 [DEBUG] switch_ivr_play_say.c:1672 done
>> >> playing
>> >> file
>> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_name1.wav
>> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>> >> playback(tone_stream://%(500,
>> >> 0,
>> >> 640))
>> >> 2011-12-07 14:52:18.799791 [DEBUG] switch_ivr_play_say.c:1302 Codec
>> >> Activated L16 at 8000hz 1 channels 20ms
>> >> 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:1672 done
>> >> playing
>> >> file tone_stream://%(500, 0, 640)
>> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>> >> set(playback_terminators=#*0123456789)
>> >> 2011-12-07 14:52:19.299814 [DEBUG] mod_dptools.c:1263
>> >> sofia/internal/102 at 192.168.1.6:5060 SET
>> >> [playback_terminators]=[#*0123456789]
>> >>
>> >> Then the caller's name is recorded, and I've verified that the
>> >> recording
>> >> is
>> >> indeed saved in /tmp:
>> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 record(/tmp/102-name.wav 7
>> >> 200
>> >> 2)
>> >> 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:585 Raw Codec
>> >> Activated
>> >> 2011-12-07 14:52:19.299814 [DEBUG] switch_core_codec.c:116
>> >> sofia/internal/102 at 192.168.1.6:5060 Push codec L16:70
>> >> .
>> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(group_confirm_key=1)
>> >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263
>> >> sofia/internal/102 at 192.168.1.6:5060 SET [group_confirm_key]=[1]
>> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>> >> set(fail_on_single_reject=true)
>> >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263
>> >> sofia/internal/102 at 192.168.1.6:5060 SET [fail_on_single_reject]=[true]
>> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060
>> >> set(group_confirm_file=phrase:screen_confirm:/tmp/102-name.wav)
>> >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263
>> >> sofia/internal/102 at 192.168.1.6:5060 SET
>> >> [group_confirm_file]=[phrase:screen_confirm:/tmp/102-name.wav]
>> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true)
>> >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263
>> >> sofia/internal/102 at 192.168.1.6:5060 SET [continue_on_fail]=[true]
>> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/102)
>> >> .
>> >> Then when attempting to play back the Output from the console showing
>> >> the
>> >> error is here:
>> >> 2011-12-07 14:52:26.348406 [ERR] switch_ivr_play_say.c:142 Can't find
>> >> macro
>> >> screen_confirm.
>> >> 2011-12-07 14:52:26.348406 [WARNING] switch_ivr_play_say.c:339 Macro
>> >> [screen_confirm]: '/tmp/102-name.wav' did not match any patterns
>> >> 2011-12-07 14:52:26.348406 [ERR] switch_ivr_originate.c:219
>> >> sofia/internal/sip:102 at 192.168.1.4:5060 Error Playing File!
>> >>
>> >> The call goes right to voicemail once the destination extension
>> >> attempts
>> >> to
>> >> answer it.
>> >>
>> >> Where are these macros supposed to be stored?  Somewhere under
>> >> /usr/local/freeswitch/conf/lang/en?  Do I need to create a macro for
>> >> screen_confirm or is it just named incorrectly or in the wrong place?
>> >>
>> >> Thanks,
>> >>
>> >> Phil Quesinberry
>> >> Q Systems Engineering, Inc.
>> >> Electronic Controls and Embedded Systems Development
>> >> (410) 969-8002
>> >> http://www.qsystemsengineering.com
>> >>
>> >>
>> >>
>> >>
>> >> _________________________________________________________________________
>> >> Professional FreeSWITCH Consulting Services:
>> >> consulting at freeswitch.org
>> >> http://www.freeswitchsolutions.com
>> >>
>> >> 
>> >> 
>> >>
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>> >> http://www.freeswitch.org
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>> >>
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>> >>
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>> >
>> >
>> >
>> >
>> > _________________________________________________________________________
>> > Professional FreeSWITCH Consulting Services:
>> > consulting at freeswitch.org
>> > http://www.freeswitchsolutions.com
>> >
>> > 
>> > 
>> >
>> > Official FreeSWITCH Sites
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>> >
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>> >
>>
>>
>>
>> --
>> Gilad Abada
>>
>> SteadFast Telecommunications, Inc.
>>
>> Call us to find out how much you can save with VoIP!
>>
>> V: 212.589.1001
>> F: 212.589.1011
>>
>>
>> For 35 years, Steadfast Telecommunications has been providing
>> state-of-the-art communications technology to businesses and
>> government agencies - large and small. Steadfast Telecommunications
>> tailors Unified Communications and Voice-Over IP Solutions to
>> single-site offices or multi-site and worldwide enterprises.   Make
>> your virtual office a reality.  Enjoy the freedom to travel while
>> remaining connected to your office.
>>
>> _________________________________________________________________________
>> Professional FreeSWITCH Consulting Services:
>> consulting at freeswitch.org
>> http://www.freeswitchsolutions.com
>>
>> 
>> 
>>
>> Official FreeSWITCH Sites
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>>
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>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> 
> 
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://wiki.freeswitch.org
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>
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-- 
Gilad Abada

SteadFast Telecommunications, Inc.

Call us to find out how much you can save with VoIP!

V: 212.589.1001
F: 212.589.1011


For 35 years, Steadfast Telecommunications has been providing
state-of-the-art communications technology to businesses and
government agencies - large and small. Steadfast Telecommunications
tailors Unified Communications and Voice-Over IP Solutions to
single-site offices or multi-site and worldwide enterprises.   Make
your virtual office a reality.  Enjoy the freedom to travel while
remaining connected to your office.



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