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Any way to block on "freeswitch.xml.fsxml" being generated?
by Nathan Neulinger 13 Mar '14

13 Mar '14

Right now, if you run reloadxml, it returns completion, but the merged config is often not fully written. Is there any way to wait for that to fully generate? Or alternatively, could this code be changed to have that file updated atomically (written to tmp file + renamed into place)? Seems like it would be a relatively trivial change to switch_xml_parse_file to do it atomically. -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Lua variables
by Alex Lake 13 Mar '14

13 Mar '14

I'm looking into moving a more Lua-orientated view of call handling. An upshot of this is that some variables are more neatly (and efficiently) retained as Lua local or global variables instead of being channel variables. One of the things that would be very nice would be to have access from one Lua script to another to look at a master-leg's data. Is this at all possible? There are other options (eg. for the Lua script to export its data into channel variable(s) and then the other one to look at that).

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Setting jitterbuffer via jitterbuffer_msec variable in dialplan
by Markus von Arx 13 Mar '14

13 Mar '14

Hi I have three questions about the jitterbuffer (with FreeSwithc 1.2.22). 1) I seem to fail configuring the jitterbuffer via the dialplan using the channel variable jitterbuffer_msec as described here: https://wiki.freeswitch.org/wiki/Jitterbuffer Here's the part of my dialplan: <extension name="conference"> <condition field="destination_number" expression="^18(.+)$"> <action application="set" data="jitterbuffer_msec=60:200:20"/> <action application="answer"/> <action application="conference" data="$1@audioswitch_conference"/> </condition> </extension> When I open a channel using this dialplan, the jitterbuffer does not seem to get configured. I assume this because I cannot find a log message "Setting Jitterbuffer to..." from sofia_glue_activate_rtp() in my log file, so I assume that the part of the code setting the jitterbuffer does not execute. I think this might have something to do that the variable jitterbuffer_msec is not available yet when sofia_glue_activate_rtp() gets executed. Correct? Maybe related to using inline="true"? But even if I use <action inline="true" application="set" data="jitterbuffer_msec=60:200:20"/>, the jitterbuffer does not seem to get configured. => question 1: how can I configure the jitterbuffer via the dialplan using the channel variable jitterbuffer_msec? Btw: enabling <param name="auto-jitterbuffer-msec" value="20"/> in the SIP profile works fine and produces the expected log statement. But I'd rather use the channel variable as it also allows configuring the max buffer length. 2) How can I get the current status (size) of the jitterbuffer at runtime, e.g. from a console (mod_cli) command or via mod_event_socket? 3) Do I need a jitterbuffer at all for the setup with many SIP channels meeting in a FreeSwitch audio conference? My tests with a network with a big jitter indicate yes, because audio quality markedly improves when enabling the jitterbuffer. Thanks and best regards, Markus

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Clarificaton on record_post_process_exec_api
by Sean Devoy 13 Mar '14

13 Mar '14

Hi, I am trying to run a bash shell script when a recording completes. I tried it like I do for fax completion, but it either does not trigger or fails to run. For faxes I have: <action application="set" data="api_hangup_hook=system /usr/local/freeswitch/scripts/emailfax.sh 410-420-7778 /tmp/${uuid}.rxfax.tiff sdevoy(a)bizfocused.com" /> So I guessed recordings would work like: <action application="set" data="record_post_process_exec_api=system /usr/local/freeswitch/scripts/emailwav.sh 410-420-7777 /tmp/recordings/${destination_number}.wav sdevoy(a)bizfocused.com" /> I tried with a space and with a ":" following the "system" api name. How can I run achieve this? I will update the wiki if someone will explain it. The complete wiki contents are currently: Post Processing Recordings in the Dialplan * record_post_process_exec_api * record_post_process_exec_app These two variables allow the postprocessing of recorded audio. The reason this is required is if the A leg hangs up first in a call, the dialplan stops being processed, and then you aren't able to take action on the file that was recorded. These variables take the form of: <action application="set" data="record_post_process_exec_api=some_api_app:api_app args" /> <action application="set" data="record_post_process_exec_app=some_app:app args" /> Thanks, Sean

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mod_fifo and group_confirm_key
by Johny Kadarisman Kwan 12 Mar '14

12 Mar '14

I recently try mod_fifo which are working great for the most cases, but I come by scenario that I'm not sure if that can be supported by mod_fifo. There are 3 static agent for a queue, dialstring contains 'group_confirm_key=1'. When call come into the queue, Freeswitch dial first member, and asking for digit confirmation. When this agent did not press confirmation key, Freeswitch keep re-dialing the same agent over and over again. Is above the correct behaviour? Is there any setting that would make FS to move on and try next agent in sequence. Thank in advance. Johny Kwan

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switch_rtp "stun binding response 487 Role Conflict"
by D D 12 Mar '14

12 Mar '14

Hi, Can you explain what this message means? 2014-03-11 15:30:08.520609 [WARNING] switch_rtp.c:857 sofia/internal/1003(a)192.168.1.79 got stun binding response 487 Role Conflict 2014-03-11 15:30:08.520609 [WARNING] switch_rtp.c:868 Changing role to CONTROLLED What does switch_rtp do in this case? Thanks, David

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freeswitch, Mavericks and MacPorts
by terry＠digital-outpost.com 12 Mar '14

12 Mar '14

For other MacPorts users, here are the steps I took to get freeswitch working on a Macmini running OS X 10.9.2, Xcode 5.0.2 & MacPorts: - autoconf, automake, libtool, pkgconfig, jpeg and openssl installed with MacPorts - mkdir -p /usr/local/src/freeswitch - cd /usr/local/src - git clone git://git.freeswitch.org/freeswitch.git - cd freeswitch - Edit configure.in and comment out darwin lines with hardcoded paths to homebrew openssl: lines 490, 491, 501, 502, 511 & 512 - ./bootstrap.sh - Edit modules.conf to enable mod_rtmp, mod_directory, mod_callcenter, mod_tts_commandline, mod_dingaling, mod_flite, mod_shout, mod_pocketsphinx & mod_cidlookup - ./configure -C CFLAGS="-I/opt/local/include" LDFLAGS="-L/opt/local/lib - make - sudo make install - sudo make cd-sounds-install cd-moh-install Configured a phone, a gateway and am making calls! -Terry

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RTP DESTINATION UNREACHABLE
by Doron Kruh 12 Mar '14

12 Mar '14

Hello, I am using Freeswitch version 1.2.12 on Debian 3.2.46-1 x86_64. My main use case includes terminating calls through gateways. In most of my gateways everything is working 100% right, but I have a private integration with a termination gateway provider which isn't working. The problem is that when performing a bridge , the SDP negotiation is finishing but no RTP packet are sent to the gateway and I see the following lines in the log: [NOTICE] switch_ivr_bridge.c:274 Hangup sofia/external/999112123752658 [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER] If I capture the traffic I can see the following ICMP packet: 6 17.453667000 144.76.108.53 85.92.157.36 ICMP 242 Destination unreachable (Port unreachable) The strange thing is that if I reboot the machine, the problem is solved for one session and than keep reproduces on the following calls. I eliminated all firewalls issue by disabling the firewall , and tried changing the hardware to Ubuntu but still the same behavior.keeps coming back. Is there any know issue which might explain this? Best regards, Doron

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Any way to tell?
by Blake Priddy 12 Mar '14

12 Mar '14

Using this board what do I need to look for in specs that will give me an estimate of concurrent calls? Will this handle 100 phones? http://www.amazon.com/gp/product/B0038GVR82/ref=oh_details_o02_s01_i01?ie=U… -- *Blakelund Priddy* Network & Systems Engineer Bryant Public School District Bryant, Arkansas 72022 http://www.bryantschools.org p 501-653-5038 f 501-847-5656

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Playing media from buffer
by prashanth d 12 Mar '14

12 Mar '14

Hi, Our product uses Freeswitch for call and media control. I have a requirement from an application consuming our product to play media from buffer instead of a file. Application want to provide the data to be played to a channel. Are there any way to achieve this with FreeSwitch ? It appears to be possible to provide such data using “switch_core_session_write_frame” . But I would end up doing what “switch_ivr_play_file” does . Am I correct ? Being an application I would prefer not to do what Freeswitch does but prefer an API for that. Thanks in advance

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