April 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Video Conference in freeswitch
by Veerabhadrarao Kankatala 27 Apr '16

27 Apr '16

Hello , I am using Freeswitch 1.6.5V to accomplish video conference through veto client provided by Freeswitch. When i am dialing 3500 (default dialplan extension) the user is connecting into video conference but if I increase the number of users upto 3 itself CPU usage crosses 160% + Currently i am using Centos 6.7 , 4GB RAM and Intel Pentium Dual 2GHz processor. Can anyone suggest me --> what is the system requirements (like OS, RAM, Processor..........) ? --> i would like to know how many conferences(with max members in each conference) was supported by freeswitch. Please tell me above information. Thanks in advance Veerabhadrarao Kankatala

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sip_h_X into verto signaling
by Tulipa Frigo 27 Apr '16

27 Apr '16

Is there a way that I can insert the sip_h_X headers, received from the sip provider , into the verto signaling to the agents via application callcenter? We have a callcenter solution where the agents are VERTO endpoints. The Freeswitch receives SIP calls from a sip provider and transfer it to the verto agents via application callcenter.

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Running FreeSWITCH error
by amani mansour 27 Apr '16

27 Apr '16

Hi all , I have a big problem ,i didn't understand but this morning when i start by running my freeSWITCH it tel me error ./freeswitch -ncwait -nonat ==>)1095 Backgrounding. FreeSWITCH[1094] Error starting system! pid:1095 but when i check freeSWITCH status i have : root@ubuntu-srv:/usr/local/freeswitch/bin# systemctl status freeswitch.service ● freeswitch.service - LSB: Freeswitch debian init script. Loaded: loaded (/etc/init.d/freeswitch) Active: active (exited) since mer. 2016-04-27 11:32:25 CET; 19min ago Docs: man:systemd-sysv-generator(8) Process: 938 ExecStop=/etc/init.d/freeswitch stop (code=exited, status=0/SUCCESS) Process: 946 ExecStart=/etc/init.d/freeswitch start (code=exited, status=0/SUCCESS) avril 27 11:32:25 ubuntu-srv systemd[1]: Stopped LSB: Freeswitch debian init script.. avril 27 11:32:25 ubuntu-srv systemd[1]: Starting LSB: Freeswitch debian init script.... avril 27 11:32:25 ubuntu-srv systemd[1]: Started LSB: Freeswitch debian init script.. avril 27 11:39:45 ubuntu-srv systemd[1]: Started LSB: Freeswitch debian init script.. avril 27 11:39:50 ubuntu-srv systemd[1]: Started LSB: Freeswitch debian init script.. can any one help me please it is urgent please Best regards Amani

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Codec list
by huseyin kalyoncu 26 Apr '16

26 Apr '16

Hello I have problem with codec list which can be seen with 'show codec' cli command. This is what i see after fresh reboot : freeswitch@fs1> show codec type,name,ikey codec,ADPCM (IMA),mod_spandsp codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.722,mod_spandsp codec,G.726 16k,mod_spandsp codec,G.726 16k (AAL2),mod_spandsp codec,G.726 24k,mod_spandsp codec,G.726 24k (AAL2),mod_spandsp codec,G.726 32k,mod_spandsp codec,G.726 32k (AAL2),mod_spandsp codec,G.726 40k,mod_spandsp codec,G.726 40k (AAL2),mod_spandsp codec,G.729,mod_com_g729 codec,GSM,mod_spandsp codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_spandsp codec,MP3,mod_shout codec,MP4V Video (passthru),mod_mp4v codec,OPUS (STANDARD),mod_opus codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,Polycom(R) G722.1/G722.1C,mod_siren codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE codec,SILK,mod_silk codec,Speex,CORE_SPEEX_MODULE codec,THEORA Video (passthru),mod_theora codec,VP8 Video,CORE_VPX_MODULE codec,VP9 Video,CORE_VPX_MODULE codec,iLBC,mod_ilbc 33 total. this is what i see after service freeswitch restart command : freeswitch@fs1> show codec type,name,ikey codec,ADPCM (IMA),mod_spandsp codec,G.722,mod_spandsp codec,G.726 16k,mod_spandsp codec,G.726 16k (AAL2),mod_spandsp codec,G.726 24k,mod_spandsp codec,G.726 24k (AAL2),mod_spandsp codec,G.726 32k,mod_spandsp codec,G.726 32k (AAL2),mod_spandsp codec,G.726 40k,mod_spandsp codec,G.726 40k (AAL2),mod_spandsp codec,G.729,mod_com_g729 codec,GSM,mod_spandsp codec,H.261 Video (passthru),mod_h26x codec,H.263 Video (passthru),mod_h26x codec,H.263+ Video (passthru),mod_h26x codec,H.263++ Video (passthru),mod_h26x codec,H.264 Video (passthru),mod_h26x codec,LPC-10,mod_spandsp codec,MP3,mod_shout codec,MP4V Video (passthru),mod_mp4v codec,OPUS (STANDARD),mod_opus codec,Polycom(R) G722.1/G722.1C,mod_siren codec,SILK,mod_silk codec,THEORA Video (passthru),mod_theora codec,iLBC,mod_ilbc 25 total. freeswitch@fs1> version FreeSWITCH Version 1.6.7+git~20160401T134007Z~f0c3870be3~64bit (git f0c3870 2016-04-01 13:40:07Z 64bit) root@fs1:~# uname -a Linux fs1 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt25-2 (2016-04-08) x86_64 GNU/Linux What would be the cause of this difference?

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Using Adhearsion with mod_rayo
by Colin Morelli 25 Apr '16

25 Apr '16

Hey all, As I'm looking into better ways to remotely control FS, adhearson and moho (over rayo) both look very powerful. Right now I'm tracking call state in the form of URLs and custom-crafted use of httapi. It works, but it's far from ideal. Has anyone had experience with using adhearsion and mod_rayo? How do you handle making sure each FS server in a cluster is actually connected to by a rayo client? mod_httapi is nice because FS initiates the outbound call, so as long as your service is up things should work. Rayo, on the other hand, appears to require that your application be connected to every running FS instance. Would love anyones' input/experience here. Best, Colin

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Call Recording On Demand for Call Center Queues (mod_callcenter)
by Shaun Stokes 25 Apr '16

25 Apr '16

Hi, We’re trying to setup call recording on demand for call center queues (mod_callcenter) but are experiencing tremendous trouble. Is it possible for the agent to perform pause\resume call recording on mod_callcenter calls which are recorded by default upon answer? Here’s the important elements of our dialplan: <condition field="${rec_file}" expression="^(.)+"> <action application="set" inline="true" data="RECORD_APPEND=true"/> <anti-action application="set" inline="true" data="RECORD_APPEND=true"/> <anti-action application="export" inline="true" data="rec_file=$${recordings_dir}/${domain_name}/archive/${strftime(%Y)}/${strftime(%b)}/${strftime(%d)}/${uuid}.${record_ext}"/> </condition> <condition field="destination_number" expression="^SETUP_RECORDING$"> <action application="log" data="INFO Configuring bind_digit_action to do recording on this session..."/> <action application="bind_digit_action" data="features,*1,exec:execute_extension,START_RECORDING XML ${domain_name}"/> <action application="bind_digit_action" data="features,*0,exec:execute_extension,STOP_RECORDING XML ${domain_name}"/> <action application="digit_action_set_realm" data=" features"/> </condition> <condition field="destination_number" expression="^START_RECORDING$"> <action application="log" data="INFO Starting recording..."/> <action application="record_session" data="${rec_file}"/> </condition> <condition field="destination_number" expression="^STOP_RECORDING$"> <action application="log" data="INFO Stopping recording..."/> <action application="stop_record_session" data="${rec_file}"/> </condition> <condition field="destination_number" expression="^837$"> <action application="export" data="nolocal:execute_on_answer=record_session ${rec_file}"/> <action application="set" data="cc_export_vars=rec_file"/> <action application="set" data="bridge_pre_execute_bleg_app=execute_extension"/> <action application="set" data="bridge_pre_execute_bleg_data=SETUP_RECORDING XML ${domain_name}"/> <action application="callcenter" data="Test-bench@tester"/> </condition> Here’s the log of the call as it comes in: SET [RECORD_APPEND]=[true] Configuring bind_digit_action to do recording on this session... Digit parser DPTOOLS: binding *1/features/0 callback: 0x7f37720eed10 data: 0x7f367c7dd918 Digit parser DPTOOLS: binding *0/features/0 callback: 0x7f37720eed10 data: 0x7f367c7ddab8 Digit parser DPTOOLS: Setting realm to 'features' EXPORT (export_vars) (REMOTE ONLY) [execute_on_answer]=[record_session /usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3] The call is then answered by the agent, we see the recording file has been created, the agent presses *0 but the call continues recording, log as follows: SET [RECORD_APPEND]=[true] RTP RECV DTMF *:1536 RTP RECV DTMF 0:1536 Stopping recording... stop_record_session(/usr/local/freeswitch/recordings/tester/archive/2016/Apr/22/cce7b636-088c-11e6-9458-f7ab40b1257a.mp3) The configuration looks as though it should work, any ideas? Is there another method we should be using? Many Thanks, Shaun ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

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Freeswitch-1.6.5 (video Conference)
by Veerabhadrarao Kankatala 25 Apr '16

25 Apr '16

Hello, -- I am currently working on video conference module in freeswitch-1.6.5 -- I am using verto client for confernece (i configured just 3 verto users in different hosts) and when i dial 3500 extension from the verto user, the conference is starting but the main problem is if 3 users connected with the 3500 conference, the audio is gone(too much noice, no audio clearence) and CPU usage is reached to 150%+ and i follow below configuration also ulimit -c unlimited # The maximum size of core files created. ulimit -d unlimited # The maximum size of a process's data segment. ulimit -f unlimited # The maximum size of files created by the shell (default option) ulimit -i unlimited # The maximum number of pending signals ulimit -n 999999 # The maximum number of open file descriptors. ulimit -q unlimited # The maximum POSIX message queue size ulimit -u unlimited # The maximum number of processes available to a single user. ulimit -v unlimited # The maximum amount of virtual memory available to the process. ulimit -x unlimited # ??? ulimit -s 240 # The maximum stack size ulimit -l unlimited # The maximum size that may be locked into memory. ulimit -a # All current limits are reported. i am using using Centos 6.5 with 4GB Ram. Could you any one help me to resolve this problem, i want to know following thigs --> Video confernece configuration --> Required system configuration --> and also i would like to know how many possible video conferences can be handle by freeswitch-1.6.5 Please help me. -- Thanks & Regards Veerabhadrarao Kankatala

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switch_rtp not changing IP with SRTP
by fs＠hellea.net 25 Apr '16

25 Apr '16

Hi, when calling to a linphone on a VPN without SRTP enabled on the leg-B, FS does [INFO] switch_rtp.c:6411 Auto Changing audio port from 192.168.1.35:7076 to 172.16.23.207:7076 But it doesn't when SRTP is enabled on the leg-B. How to force that behaviour when SRTP is enabled? Kind regards, JCh

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Don't report bugs or issues here please use jira.freeswitch.org
by Anthony Minessale 25 Apr '16

25 Apr '16

Friendly Reminder! -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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Verto registrations
by Gregor Nanger 25 Apr '16

25 Apr '16

Is there a way to flush specific verto registration. When I show status: verto status, somehow one registration is hanging. Can I flush specific verto registration from cli or ESL? When does session expire?

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