I¡¯m trying to add a new function to my callcenter server which allow peers transfer their call to other operator.
<extension name="att_xfer">
<condition field="destination_number" expression="^att_xfer$">
<action application="read" data="3 6 'tone_stream://%(10000,0,350,440)' digits 30000 #"/>
<action application="set" data="origination_cancel_key=#"/>
<action application="att_xfer" data="user/AA-${digits}@$${domain}"/>
This should works when *4 was pressed and att_xfer activated, however when I test this dialplan, freeswitch showed the log as:2019-05-08 18:31:46.546807 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF *:1280
2019-05-08 18:31:46.546807 [INFO] switch_channel.c:515 RECV DTMF *:1280
2019-05-08 18:31:46.986810 [DEBUG] switch_rtp.c:7789 RTP RECV DTMF 4:1280
2019-05-08 18:31:46.986810 [INFO] switch_channel.c:515 RECV DTMF 4:1280
2019-05-08 18:31:47.226816 [DEBUG] mod_local_stream.c:907 Opening Stream [moh/8000] 8000hz
2019-05-08 18:31:47.226816 [DEBUG] switch_ivr_play_say.c:1497 Codec Activated L16@8000hz 1 channels 20ms
2019-05-08 18:31:47.226816 [ERR] mod_native_file.c:74 Error opening /usr/local/freeswitch/sounds/en/us/callie/execute_extension:att_xfer XML features.PCMU
2019-05-08 18:31:47.226816 [ERR] mod_sndfile.c:204 Error Opening File [/usr/local/freeswitch/sounds/en/us/callie/execute_extension:att_xfer XML features.wav] [System error : No such file or directory.]
2019-05-08 18:31:47.246814 [DEBUG] switch_ivr_play_say.c:1941 done playing file local_stream://moh/8000
The anonymous is my test phone and BB-000004 is operator. I pressed *4 on operator sip phone and got the error above. Freeswitch seems to execute the playback on bind_meta_app.FreeSWITCH: FreeSWITCH Version 1.8.5+git~20190124T225359Z~31281a0bf1~64bit (git 31281a0 2019-01-24 22:53:59Z 64bit)
Regards.