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freeswitch + flashphoner web call server
by jdayola＠spingine.com 02 Mar '15

02 Mar '15

Hi Guys, Have anyone tried using flashphoner web call server? I already have a freeswitch and a flashponer webcallserver running. My problem is how to make freeswitch use the flashphoner webcallserver. Which freeswitch xml file do I need to edit to make use of the flashphoner webcallserver. Im using a flex phone client for my flex desktop application. Do you have any suggestions on the best approach for sip to rtmp gateway? Regards, jdayola

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Mod_shout libfacc support
by Eloy Coto Pereiro 02 Mar '15

02 Mar '15

Hi, I'm trying to play a libfaac[0] files into my Freeswitch, but I can't get it working. Play mp3 files work ok, but encoding with libfacc doesn't work. This is my config in mod_shout: <action application="playback" data="shout://localhost/suite.mp4"/> In the other hand, I tried mod_vlc to play mp4 files, but It didn't work too. I compiled from sources, and tried with debian backports too. <action application="playback" data="vlc://http://localhost/suite.m4a"/> In both cases http request work, reply 200 ok and with data. Play mp3 with both modules work ok. In the other hand, mod_shout and mod_vlc are loaded correctly, and module_exists return always True. Any idea? Is libfaac supported? [0] https://trac.ffmpeg.org/wiki/Encode/AAC#libfaac Regards

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supporting multiple dtmf protocols
by ik 01 Mar '15

01 Mar '15

Hello, I have the misfortune of forced to use a telco named Bezeq - the biggest telco in Israel. The SIP trunk they provide is very problematic. The one that I cannot overcome is that sometimes doing the call, it switches DTMF between Inband and rfc2833, and sometimes the whole DTMF sending is either inband or rfc2833 doing the entire call. I cannot make them to be more stable in this matter (I have tried talking with them), and I cannot replace them (I have tried to do so as well). How can I deal with such mess ? Thanks, Ido

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Announce position in queue
by Shai Perelman 01 Mar '15

01 Mar '15

I found this code for adding position annoucement ability to fusionpbx. https://code.google.com/p/fusionpbx/issues/detail?id=658 can some one point me the steps to integrate it to my fpbx installation? it looks like the code files is replacing some original code , is that good? what about version updates, isnt it going to overwrite it? are there other , better options to acheive this? Im new to this so forgive my newbie questions, I find this list to be the only solution trying to learn fs and fusion, as the documentation is very limited. Thanks Shai

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FreeSWITCH Cookbook Free Today
by Brian West 01 Mar '15

01 Mar '15

https://twitter.com/packtpub/status/572041101076381696 Go get it! Start your FreeSWITCHing today! /b -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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originating a call from event socket
by Royce Mitchell III 01 Mar '15

01 Mar '15

I need to initiate a call on an agent's behalf, but I can't seem to get it to work. This is the closest I've gotten ( 7777 puts agent back in the call-queue ) uuid_setvar <agent-uuid> transfer_after_bridge 7777 uuid_transfer <agent-uuid> <pstn-number> I have 2 problems: 1) the transfer_after_bridge isn't happening, or it's not going to the right place 2) I have to wait forever to get a uuid to control that call. I want to be able to allocate a uuid for the outbound leg before I initiate the call, but I can't figure out the syntax to make it work. I've tried this, but I get a fast busy: uuid_transfer <agent-uuid> {origination_uuid=<new-uuid>}<pstn-number> I've also tried this and it doesn't work either: originate {origination_uuid=<new-uuid>}sofia/external/<pstn-number> &bridge(<agent-extension>) Please help, thanks Royce Mitchell, IT Consultant ITAS Solutions royce3(a)itas-solutions.com

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mod_callcenter - cc_warning_tone not working
by bedgar＠vseinc.com 27 Feb '15

27 Feb '15

Hello Experts, I have been working on setting up a call center in FS and am having a problem with the warning tone notification. It does not play any tones before the caller is bridged to the agent. I am using the latest stable build for 1.2 and I am using the example in the wiki: https://wiki.freeswitch.org/wiki/Mod_callcenter#uuid-standby Snippet: <extension> <condition field="destination_number" expression="^(4099)$"> <action application="set" data="transfer_after_bridge=4099"/>  <action application="set" data="res=${callcenter_config(agent set uuid ${caller_id_number}@${domain_name} '${uuid}')}" /> <action application="set" data="res=${callcenter_config(agent set type ${caller_id_number}@${domain_name} 'uuid-standby')}" /> <action application="set" data="res=${callcenter_config(agent set status ${caller_id_number}@${domain_name} 'Available (On Demand)')}" /> <action application="set" data="res=${callcenter_config(agent set state ${caller_id_number}@${domain_name} 'Waiting')}" /> <action application="set" data="cc_warning_tone=tone_stream://%(200,0,500,600,700)"/> <action application="answer" /> <action application="playback" data="$${hold_music}"/> </condition> </extension> I have tried changing the tone values to see if making it repeat or play longer would produce a positive result. No dice... On a second note, is there a way to play a file to the agent before the call is bridged? My call center services multiple brands and would like to have the agent greet the caller appropriately. Thank you, Brian Edgar

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Freeswitch 1.4.15 and a sudden jump in CPU usage
by Ali Pey 27 Feb '15

27 Feb '15

Hello, Normally our CPU usage during peak hours are less than %10; however, time to time, freeswitch uses more than %95 of the CPU and it stays there forever. The only way to get it back to normal is to restart freeswitch service. During the high CPU usage, freeswitch continues to operate but very very slow. If I lower the traffic quite a bit, the CPU comes down, but adding back the traffic, the CPU usage goes right back up. After freeswitch restart, it becomes normal again with the same traffic. Any ideas or suggestions? Much appreciated. Thanks, Ali Pey

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Nibblebill round to minute
by Zoltán Szabó 27 Feb '15

27 Feb '15

Hi, Is there any way to configure nibblebill to round up the billing of a call leg to minutes? So for example if the call is just 25 seconds long, I would like to bill one minute. If the call is 80 seconds, 2 minutes, etc. Many thanks, Zoltan

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 27 Feb '15

27 Feb '15

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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