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reroute conference audio
by Joli Martinez 13 Nov '13

13 Nov '13

Hello, I would like to send the conference bridge audio to another sip server. So basically Caller calls freeswitch free switch places user to conference1 and then sends audio stream via SIP to a separate SIP server which will then record and keep track of all the reporting for the conference. I know freeswitch can do the recording and reporting, but there is a dedicated box already for that and that is what y boss wants. Could someone give me some suggestion as to how I can accomplish this? Thanks,

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TEST
by Brian West 12 Nov '13

12 Nov '13

-- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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spaces in params to uuid_broadcast
by Alex Lake 12 Nov '13

12 Nov '13

I want to use uuid_broadcast to run a lua app with a few parameters. http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast Doesn't give an example of this, and the few things I've tried didn't really work. This isn't even worth bothering with: uuid_broadcast f6db28c5-1703-4819-a285-d4b197983bb6 lua::mcm3.lua param1 param2 bleg This one didn't seem to work: uuid_broadcast f6db28c5-1703-4819-a285-d4b197983bb6 "lua::mcm3.lua param1 param2" bleg Any clues? ;-)

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Nibblebiil "Pause" causing session locked
by doremo chin 12 Nov '13

12 Nov '13

Hi All, I'm trying to use nibblebill "pause" function for a particular call but the call session locked after the call hangup. Dialplan, <extension name="outofcredit"> <condition field="destination_number" expression="^(outofcredit)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="nibble_rate=0"/> <action application="nibblebill" data="pause"/> <action application="answer"/> <action application="playback" data="$${sounds_dir}/en/us/callie/misc/8000/misc-your_call_has_been_terminated.wav"/> <action application="hangup"/> </condition> </extension> Call logs, 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:48 sofia/sipinterface_2/4668(a)192.168.0.106 Standard HANGUP, cause: NONE 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:676 (sofia/sipinterface_2/4668(a)192.168.0.106) State HANGUP going to sleep 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:689 (sofia/sipinterface_2/4668(a)192.168.0.106) Callstate Change ACTIVE -> HANGUP 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:446 (sofia/sipinterface_2/4668(a)192.168.0.106) State Change CS_HANGUP -> CS_REPORTING 2013-07-30 20:38:33.360312 [DEBUG] switch_core_session.c:1340 Send signal sofia/sipinterface_2/4668(a)192.168.0.106 [BREAK] 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:415 (sofia/sipinterface_2/4668(a)192.168.0.106) Running State Change CS_REPORTING 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:761 (sofia/sipinterface_2/4668(a)192.168.0.106) State REPORTING 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:92 sofia/sipinterface_2/4668(a)192.168.0.106 Standard REPORTING, cause: NONE 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:761 (sofia/sipinterface_2/4668(a)192.168.0.106) State REPORTING going to sleep 2013-07-30 20:38:33.360312 [DEBUG] switch_core_state_machine.c:440 (sofia/sipinterface_2/4668(a)192.168.0.106) State Change CS_REPORTING -> CS_DESTROY 2013-07-30 20:38:33.360312 [DEBUG] switch_core_session.c:1340 Send signal sofia/sipinterface_2/4668(a)192.168.0.106 [BREAK] 2013-07-30 20:38:33.360312 [DEBUG] switch_core_session.c:1548 Session 3 (sofia/sipinterface_2/4668(a)192.168.0.106) Locked, Waiting on external entities show channels & uuid_kill, freeswitch@internal> show channels uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid,sent_callee_name,sent_callee_num f161cb1c-f914-11e2-92d1-35a97464e755,inbound,2013-07-30 20:38:31,1375187911,sofia/sipinterface_2/4668@192.168.0.106 ,CS_EXECUTE,4668,4668,192.168.0.133,0,playback,/usr/local/freeswitch/sounds/en/us/callie/misc/8000/misc-your_call_has_been_terminated.wav,XML,out_bound,PCMA,8000,64000,PCMA,8000,64000,, sip.wit.net.my,4668(a)192.168.0.106,,ACTIVE,,,,,, 1 total. freeswitch@internal> uuid_kill f161cb1c-f914-11e2-92d1-35a97464e755 -ERR No such channel! freeswitch@internal> May i know is there any way to release the locked session? or is it a bug in freeswitch ? Thanks, dc

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mute/unmute conference
by Joli Martinez 12 Nov '13

12 Nov '13

Hello, I need the ability to have first person to join the conference have the ability to mute/unmute the other callers. Any idea how to acomplish this. Thanks,

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OPUS parameters
by Jan Sverre Andreassen 11 Nov '13

11 Nov '13

I want to control OPUS bandwidth and sound quality. Is it an easy way to send parameters to mod_opus? Best Regards, Jan Sverre Andreassen ________________________________ This email and any files contained therein is confidential and may contain privileged information. If you are not the named addressee(s) or you have otherwise received this in error, you should not distribute or copy this e-mail or use any of its content for any purpose. Please notify the sender immediately by e-mail if you have received this e-mail in error and delete it from your system.

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Multi-access to lua scripts
by Alex Lake 11 Nov '13

11 Nov '13

Got a dialplan which calls a lua script that does a bridge. That bridge may well end up calling some bit of dialplan that calls the same lua script. I've found that fs seems to complain that there is "No such file or directory" when the script is being called for the second time. Am I smoking something odd here, or is this WAD?

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ORIGINATOR_CANCEL doesn't stop the ringing at the called party.
by Vincent Xia 11 Nov '13

11 Nov '13

for a vanilla call process, the called party will hangup immediately after the caller cancels the call before it's answered, this is true when im using <action application="bridge" data="user/${destination_number}" /> in the dialplan xml. however, if the call is handled by a lua script with a pattern like sessionB = freeswitch.Session('user/xxxx'); if sessionB:ready() then freeswitch.bridge(session, sessionB); else cause = sessionB:hangupCause(); if (cause == "NO_USER_RESPONSE" or cause=="NO_ANSWER") then session:streamFile("user_no_answer.wav"); end session:hangup(); end the called party is still ringing after the caller cancels the call before it's answered，this lua call process pattern is from wiki so im pretty sure it's being widely used, but why im having this problem? could someone figure this out?

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About the hardware fax detection on Sangoma cads
by sparklezou 11 Nov '13

11 Nov '13

Hi All, When configure the card and freetdm via wancfg_fs command, there is a question. Would you like to enable hardware fax detection? I always choose YES. I don't know how to use it. Is there any predefined dialplan context,such as "fax",to handle the incoming the fax? Thanks! 2013-11-11 sparklezou

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blind transfer back to caller on failure
by Tiago Sampaio 11 Nov '13

11 Nov '13

Hi all. I'm wondering if is possible to return the call if no answer or user busy? i tryed this: <context name="INTERNAL"> <extension name="INTER_RAMAL"> <condition field="destination_number" expression="^(\d{4})$"> <action application="set" data="continue_on_fail=true" /> <action application="bridge" data="[originate_timeout=10]user/$1" /> </condition> </extension> <extension name="BLIND_TRANSFER"> <condition field="destination_number" expression="^blind_transfer$"> <action application="answer"/> <action application="read" data="3 4 'tone_stream://%(10000,0,350,440)' digits 5000 #"/> <action application="set" data="continue_on_fail=true" /> <action application="execute_extension" data="is_transfer XML INTERNAL" />  <action application="bridge" data="user/${caller_id_number}" /> <action application="hangup" /> </condition> </extension> <extension name="is_transfer"> <condition field="destination_number" expression="^is_transfer$"/> <condition field="${digits}" expression="^(\d+)$"> <action application="set" data="continue_on_fail=true" /> <action application="transfer" data="-bleg ${digits} XML INTERNAL"/> <anti-action application="eval" data="cancel transfer"/> </condition> </extension> </context> Example: User A call user B, user B answer and do blind transfer to user C. User B hangup when call blind transfer and C rings 30 seconds and then transfer again to B before hangup A... Now if i do blind transfer and no answer or user busy, freeswitch hangup all sides...

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