Can't quite get call screening to work
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@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@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@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@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@192.168.1.6:5060 set(playback_terminators=#*0123456789) 2011-12-07 14:52:19.299814 [DEBUG] mod_dptools.c:1263 sofia/internal/102@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@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@192.168.1.6:5060 Push codec L16:70 . EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [group_confirm_key]=[1] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [fail_on_single_reject]=[true] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [group_confirm_file]=[phrase:screen_confirm:/tmp/102-name.wav] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [continue_on_fail]=[true] EXECUTE sofia/internal/102@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@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
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@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@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@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@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@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@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@192.168.1.6:5060 set(playback_terminators=#*0123456789) 2011-12-07 14:52:19.299814 [DEBUG] mod_dptools.c:1263 sofia/internal/102@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@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@192.168.1.6:5060 Push codec L16:70 . EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [group_confirm_key]=[1] EXECUTE sofia/internal/102@192.168.1.6:5060set(fail_on_single_reject=true) 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 sofia/internal/102@192.168.1.6:5060 SET [fail_on_single_reject]=[true] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [group_confirm_file]=[phrase:screen_confirm:/tmp/102-name.wav] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [continue_on_fail]=[true] EXECUTE sofia/internal/102@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@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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Hi I am trying to get call screening to work too. http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_13:_Call_Scr... 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@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@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@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@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@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@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@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@192.168.1.6:5060 set(playback_terminators=#*0123456789) 2011-12-07 14:52:19.299814 [DEBUG] mod_dptools.c:1263 sofia/internal/102@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@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@192.168.1.6:5060 Push codec L16:70 . EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [group_confirm_key]=[1] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [fail_on_single_reject]=[true] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [group_confirm_file]=[phrase:screen_confirm:/tmp/102-name.wav] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [continue_on_fail]=[true] EXECUTE sofia/internal/102@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@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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- 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.
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@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_Scr...
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@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@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@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@192.168.1.6:5060phrase(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@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@192.168.1.6:5060playback(tone_stream://%(500, 0, 640)) 2011-12-07 14:52:18.799791 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16@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@192.168.1.6:5060 set(playback_terminators=#*0123456789) 2011-12-07 14:52:19.299814 [DEBUG] mod_dptools.c:1263 sofia/internal/102@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@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@192.168.1.6:5060 Push codec L16:70 . EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [group_confirm_key]=[1] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [fail_on_single_reject]=[true] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [group_confirm_file]=[phrase:screen_confirm:/tmp/102-name.wav] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [continue_on_fail]=[true] EXECUTE sofia/internal/102@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@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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- 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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thanks got it working and updated thre wiki On Fri, Dec 23, 2011 at 1:08 PM, Michael Collins <msc@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@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_Scr...
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@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@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@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@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@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@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@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@192.168.1.6:5060 set(playback_terminators=#*0123456789) 2011-12-07 14:52:19.299814 [DEBUG] mod_dptools.c:1263 sofia/internal/102@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@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@192.168.1.6:5060 Push codec L16:70 . EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [group_confirm_key]=[1] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [fail_on_single_reject]=[true] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [group_confirm_file]=[phrase:screen_confirm:/tmp/102-name.wav] EXECUTE sofia/internal/102@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@192.168.1.6:5060 SET [continue_on_fail]=[true] EXECUTE sofia/internal/102@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@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@freeswitch.org http://www.freeswitchsolutions.com
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UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- 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@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com
Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- 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.
participants (3)
-
Gilad Abada
-
Michael Collins
-
Phil Quesinberry