- FreeSWITCH-users - FreeSWITCH Mailing Lists

After a system reboot a skype client can't start
by Xenia Obolenskaya 18 Nov '17

18 Nov '17

Dear Mr Sergey Safarov, Thank you for your interesting reply. I know of the last unexpected move from Microsoft to discontinue skype service for old Linux versions. However, on another server the same version of the skype client is currently working. Of course, this may be just a matter of time. The problem is not only that the skype client is not registered in Microsoft server, but also that the process in the system can't be initiated properly. Once again, thank you for help. Regards, Xenia Obolenskaya -------------------------------- *Sergey Safarov* s.safarov at gmail.com wrote: <freeswitch-users%40lists.freeswitch.org?Subject=Re%3A%20%5BFreeswitch-users%5D%20After%20a%20system%20reboot%20a%20skype%20client%20can%27t%0A%09start&In-Reply-To=%3CCAHtxdDe_p41Fa%2BxMJt-jEga3aLDnPbk%2B8q1s7X6%2B_ZLN_UCzmA%40mail.gmail.com%3E> >Some time ago Microsoft announsed that old skype version is not more >supported and will stop working in future. >Think no way get back FreeSwith+Skype at present time

1 0

Group call limit on answering extension
by Piccinin Francesco 17 Nov '17

17 Nov '17

Hi all, I need you help in order to solve a problem with group_call function. I would like to set limit (on db backend) for an extension that answer at a group_call bridge. I'm trying to explain the problem with an example. Ext A is calling a group that contains extensions B and C. C answers the call and I would like to insert limit for busy and max call for C. I'm not able to find a channel's variable that ensures me call has been answered by C. All of channels variables are store on DB on my config (freeswich cluster); it seems that channels are going to be updated slower then "execute_on_answer" trigger. I'm using a lua script running with "execute_on_answer" that looks for callstate=ACTIVE on call uuid but does not retrieve any data. Thanks for help Regards Francesco

1 0

Re: [Freeswitch-users] caller control options for outbound conference calls
by Shishir Pokharel 17 Nov '17

17 Nov '17

Try setting moderator-controls along with caller-control. For some reason if moderator-controls is not included in conference XML it won’t recognize caller-control. Anyone knows the reason behind it? /Shishir On Thu, Nov 16, 2017 at 7:26 PM, Shishir Pokharel < shishir.pokharel(a)gmail.com> wrote: > Try setting moderator-controls along with caller-control. For some reason > if moderator-controls is not included in conference XML it won’t recognize > caller-control. > > Anyone knows the reason behind it? > > /Shishir > > > On Nov 16, 2017, at 10:33, John Covici <covici(a)ccs.covici.com> wrote: > > > > To change the caller control options, I am pretty sure you have to > > unload mod_conference, so I would do this programatically instead. > > > > On Thu, 16 Nov 2017 08:10:06 -0500, > > Ketan Kothari wrote: > >> > >> [1 <multipart/alternative (7bit)>] > >> [1.1 <text/plain; UTF-8 (7bit)>] > >> [1.2 <text/html; UTF-8 (base64)>] > >> Hello Guys, > >> > >> Can i have any suggestion on the same ? > >> > >> On Wed, Nov 15, 2017 at 5:11 PM, Ketan Kothari <kkothari157(a)gmail.com> > wrote: > >> > >> Hello there, > >> > >> We are trying to use caller control options for outbound conference > calls. > >> We are using originate command to originate conference call with PSTN > numbers. > >> I am using mod_XML for write conference. > >> I use default caller controls as well as also added custom caller > controls. > >> My caller control name is plain. I have added it in dialplan parameter > conference_control=plain as well as add it in conference parameter still > caller mute option is not working. > >> Below is my conference XML: > >> > >> <document type="freeswitch/xml"> > >> <section name="Configuration" description="Configuration"> > >> <configuration name="conference.conf" description="Audio Conference"> > >> <advertise> > >> <room name="0676338702(a)xxx.xxx.xxx.xx" status="FreeSWITCH"/> > >> </advertise> > >> <caller-controls> > >> <group name="plain"> > >> <control action="mute" digits="0"/> > >> <control action="vol talk dn" digits="1"/> > >> <control action="vol talk zero" digits="2"/> > >> <control action="vol talk up" digits="3"/> > >> <control action="vol listen dn" digits="4"/> > >> <control action="vol listen zero" digits="5"/> > >> <control action="vol listen up" digits="6"/> > >> <control action="energy dn" digits="7"/> > >> <control action="energy equ" digits="8"/> > >> <control action="energy up" digits="9"/> > >> <control action="deaf mute" digits="*"/> > >> <control action="execute_application" digits="#" > data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default" /> > >> </group> > >> </caller-controls> > >> <profiles> > >> <profile name="test"> > >> <param name="caller-id-number" value="xxxxxxxxxx"/> > >> <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/> > >> <param name="exit-sound" value="tone_stream://%(500,0, > 300,200,100,50,25)"/> > >> <param name="auto-record" value="${recordings_dir}/ > testrushika_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> > >> <param name="member-flags" value="dist-dtmf"/> > >> <param name="pin-retries" value="10"/> > >> <param name="max-members" value="1000"/> > >> <param name="status" value="0"/> > >> <param name="caller-controls" value="plain"/> > >> </profile> > >> </profiles> > >> </configuration> > >> </section> > >> </document> > >> > >> Please let me know if anything is not correct or any new configuration > need to go. > >> > >> [2 <text/plain; utf-8 (base64)>] > >> ____________________________________________________________ > _____________ > >> Professional FreeSWITCH Consulting Services: > >> consulting(a)freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://confluence.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users(a)lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/ > options/freeswitch-users > >> http://www.freeswitch.org > > > > -- > > Your life is like a penny. You're going to lose it. The question is: > > How do > > you spend it? > > > > John Covici > > covici(a)ccs.covici.com > > > > ____________________________________________________________ > _____________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://confluence.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > -- Shishir Pokharel

3 2

caller control options for outbound conference calls
by Ketan Kothari 17 Nov '17

17 Nov '17

Hello there, We are trying to use caller control options for outbound conference calls. We are using originate command to originate conference call with PSTN numbers. I am using mod_XML for write conference. I use default caller controls as well as also added custom caller controls. My caller control name is plain. I have added it in dialplan parameter conference_control=plain as well as add it in conference parameter still caller mute option is not working. Below is my conference XML: <document type="freeswitch/xml"> <section name="Configuration" description="Configuration"> <configuration name="conference.conf" description="Audio Conference"> <advertise> <room name="0676338702(a)xxx.xxx.xxx.xx" status="FreeSWITCH"/> </advertise> <caller-controls> <group name="plain"> <control action="mute" digits="0"/> <control action="vol talk dn" digits="1"/> <control action="vol talk zero" digits="2"/> <control action="vol talk up" digits="3"/> <control action="vol listen dn" digits="4"/> <control action="vol listen zero" digits="5"/> <control action="vol listen up" digits="6"/> <control action="energy dn" digits="7"/> <control action="energy equ" digits="8"/> <control action="energy up" digits="9"/> <control action="deaf mute" digits="*"/> <control action="execute_application" digits="#" data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default" /> </group> </caller-controls> <profiles> <profile name="test"> <param name="caller-id-number" value="xxxxxxxxxx"/> <param name="enter-sound" value="tone_stream://%(200,0,500,600,700)"/> <param name="exit-sound" value="tone_stream://%(500,0,300,200,100,50,25)"/> <param name="auto-record" value="${recordings_dir}/testrushika_${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="member-flags" value="dist-dtmf"/> <param name="pin-retries" value="10"/> <param name="max-members" value="1000"/> <param name="status" value="0"/> <param name="caller-controls" value="plain"/> </profile> </profiles> </configuration> </section> </document> Please let me know if anything is not correct or any new configuration need to go.

3 3

After a system reboot a skype client can't start
by Xenia Obolenskaya 17 Nov '17

17 Nov '17

After executing sh /usr/local/freeswitch/skypopen/skype-clients- startup-dir/start_skype_clients.sh I can see that the skype client process is absent # ps -aux: root 4353 0.0 0.3 126244 3092 ? Ss 10:10 0:00 /usr/sbin/crond -n pulse 30755 0.0 0.6 296340 6480 ? S<l 10:54 0:00 /usr/bin/pulseaudio --system=true --daemonize=true --log-level=info --log-target=sys root 30908 0.0 1.2 113140 12556 pts/0 S 11:01 0:00 /usr/bin/Xvfb :101 -ac -nolisten tcp -screen 0 640x480x8 root 30967 1.5 5.4 884616 55260 ? S<l 11:11 0:19 freeswitch -nc root 31041 0.0 0.0 0 0 ? S 11:22 0:00 [kworker/0:0] root 31048 0.0 0.0 0 0 ? R 11:28 0:00 [kworker/0:2] root 31054 0.0 0.3 151072 3964 pts/0 R+ 11:33 0:00 ps -aux I tried several times to kill process /usr/bin/Xvfb and start skype client again but failed. I rebooted a system. I deleted completely directories: /tmp /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir /usr/local/freeswitch/skypopen/skype-clients-configuration-dir /usr/local/freeswitch/skypopen/skype-clients-startup-dir and executed the "install.pl" script cd /usr/src/freeswitch/src/mod/endpoints/mod_skypopen/install ./install.pl However a skype client don't start. Of course, mod_skypopen also failed: 2017-11-17 11:13:08.741603 [DEBUG] mod_skypopen.c:1542 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1542 ][none ][N/A,N/A] globals.dialplan=XML 2017-11-17 11:13:08.741603 [DEBUG] mod_skypopen.c:1539 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1539 ][none ][N/A,N/A] globals.context=default 2017-11-17 11:13:08.741603 [DEBUG] mod_skypopen.c:1545 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1545 ][none ][N/A,N/A] globals.destination=5000 2017-11-17 11:13:08.741603 [DEBUG] mod_skypopen.c:1548 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1548 ][none ][N/A,N/A] globals.skype_user=xenia.obolenskya 2017-11-17 11:13:08.741603 [DEBUG] mod_skypopen.c:1551 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1551 ][none ][N/A,N/A] globals.report_incoming_chatmessages=false 2017-11-17 11:13:08.741603 [DEBUG] mod_skypopen.c:1557 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1557 ][none ][N/A,N/A] globals.write_silence_when_idle=false 2017-11-17 11:13:08.741603 [DEBUG] mod_skypopen.c:1560 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1560 ][none ][N/A,N/A] globals.setsockopt=true 2017-11-17 11:13:08.741603 [DEBUG] mod_skypopen.c:1653 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1653 ][none ][N/A,N/A] interface_id=1 2017-11-17 11:13:08.741603 [DEBUG] mod_skypopen.c:1664 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1664 ][none ][N/A,N/A] name=skype101 2017-11-17 11:13:08.741603 [DEBUG] mod_skypopen.c:1670 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1670 ][none ][N/A,N/A] Initialized XInitThreads! 2017-11-17 11:13:08.761605 [DEBUG] mod_skypopen.c:1693 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1693 ][skype101 ][IDLE,IDLE] CONFIGURING interface_id=1 2017-11-17 11:13:08.761605 [DEBUG] mod_skypopen.c:1730 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1730 ][skype101 ][IDLE,IDLE] interface_id=1 globals.SKYPOPEN_INTERFACES[ interface_id].name=skype101 2017-11-17 11:13:08.761605 [DEBUG] mod_skypopen.c:1733 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1733 ][skype101 ][IDLE,IDLE] interface_id=1 globals.SKYPOPEN_INTERFACES[ interface_id].context=default 2017-11-17 11:13:08.761605 [DEBUG] mod_skypopen.c:1736 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1736 ][skype101 ][IDLE,IDLE] interface_id=1 globals.SKYPOPEN_INTERFACES[ interface_id].dialplan=XML 2017-11-17 11:13:08.761605 [DEBUG] mod_skypopen.c:1739 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1739 ][skype101 ][IDLE,IDLE] interface_id=1 globals.SKYPOPEN_INTERFACES[ interface_id].destination=5000 2017-11-17 11:13:08.761605 [DEBUG] mod_skypopen.c:1742 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1742 ][skype101 ][IDLE,IDLE] interface_id=1 globals.SKYPOPEN_INTERFACES[ interface_id].X11_display=:101 2017-11-17 11:13:08.761605 [DEBUG] mod_skypopen.c:1745 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1745 ][skype101 ][IDLE,IDLE] interface_id=1 globals.SKYPOPEN_INTERFACES[ interface_id].skype_user=xenia.obolenskaya 2017-11-17 11:13:08.761605 [DEBUG] mod_skypopen.c:1748 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1748 ][skype101 ][IDLE,IDLE] interface_id=1 globals.SKYPOPEN_INTERFACES[ interface_id].report_incoming_chatmessages=0 2017-11-17 11:13:08.761605 [DEBUG] mod_skypopen.c:1751 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1751 ][skype101 ][IDLE,IDLE] interface_id=1 globals.SKYPOPEN_INTERFACES[ interface_id].silent_mode=0 2017-11-17 11:13:08.761605 [DEBUG] mod_skypopen.c:1754 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1754 ][skype101 ][IDLE,IDLE] interface_id=1 globals.SKYPOPEN_INTERFACES[ interface_id].write_silence_when_idle=0 2017-11-17 11:13:08.761605 [DEBUG] mod_skypopen.c:1757 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1757 ][skype101 ][IDLE,IDLE] interface_id=1 globals.SKYPOPEN_INTERFACES[ interface_id].setsockopt=1 2017-11-17 11:13:08.761605 [WARNING] mod_skypopen.c:1759 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [WARNINGA 1759 ][skype101 ][IDLE,IDLE] STARTING interface_id=1 2017-11-17 11:13:08.761605 [DEBUG] skypopen_protocol.c:1869 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1869 ][skype101 ][IDLE,IDLE] X Display ':101' opened 2017-11-17 11:13:08.761605 [ERR] skypopen_protocol.c:1686 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [ERRORA 1686 ][none ][N/A,N/A] Received error code 5 from X Server 2017-11-17 11:13:08.761605 [ERR] skypopen_protocol.c:1687 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [ERRORA 1687 ][none ][N/A,N/A] Display error for 1, :101 2017-11-17 11:13:08.761605 [ERR] skypopen_protocol.c:1884 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [ERRORA 1884 ][skype101 ][DEAD,IDLE] Fatal display error for :101 - successed to jump 2017-11-17 11:13:08.761605 [WARNING] skypopen_protocol.c:1910 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [WARNINGA 1910 ][skype101 ][DEAD,IDLE] Removing skype interface #skype101 2017-11-17 11:13:08.841659 [DEBUG] mod_skypopen.c:1422 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1422 ][skype101 ][DEAD,IDLE] In skypopen_signaling_thread_func: started, p=0x7fa2cccb1d70 2017-11-17 11:13:08.841659 [DEBUG] mod_skypopen.c:1506 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 1506 ][skype101 ][DEAD,IDLE] EXITING 2017-11-17 11:13:08.961605 [NOTICE] mod_skypopen.c:1788 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [NOTICA 1788 ][skype101 ][DEAD,IDLE] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=1 2017-11-17 11:13:18.981616 [ERR] mod_skypopen.c:1801 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [ERRORA 1801 ][skype101 ][DEAD,IDLE] Failed to connect to a SKYPE API for interface_id=1, no SKYPE client running, please (re)start Skype client. Skypopen exiting 2017-11-17 11:13:18.981616 [DEBUG] mod_skypopen.c:421 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 421 ][skype101 ][DEAD,IDLE] interface 'skype101' STILL console 2017-11-17 11:13:18.981616 [DEBUG] mod_skypopen.c:431 [1.6.19+git~20171019T182535Z~f51b5d2dce~64bit ] [DEBUG_SKYPE 431 ][ ][IDLE,IDLE] interface 'skype101' deleted successfully 2017-11-17 11:13:19.961652 [CRIT] switch_loadable_module.c:1522 Error Loading module /usr/lib/freeswitch/mod/mod_skypopen.so **Module load routine returned an error**

2 1

B-Leg Missing SPD payloads
by Juan Pablo Godoy Macari 17 Nov '17

17 Nov '17

Hello Everyone, I'm Trying to establish a SIP Trunk with a provider, but I'm encountering a strange problem. When I receive an INVITE from my provider its SDP have this format: v=0. o=- 1134042028 1134042029 IN IP4 XXXX. s=SBC call. c=IN IP4 XXXXX. t=0 0. m=audio 10042 RTP/AVP 8 0 116. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:116 telephone-event/8000. a=ptime:5. a=curr:qos local none. a=curr:qos remote none. a=des:qos mandatory local sendrecv. a=des:qos optional remote sendrecv. a=3gOoBTC. The problem is when my freeswitch is trying to call to the B-Leg its sending the SDP without any payloads: v=0. o=SBC 1510738948 1510738949 IN IP4 XXXXX. s=SBC. c=IN IP4 XXXXXX. t=0 0. Why is that? I debugged from the CLI (level 10) and I can't see any errors from Freeswitch. I tested this on versions 1.6.12 and 1.6.17 with the same results. I'm guessing that Sofia or Freeswitch don't support the qos values, it's the only different thing from my others providers, I'll thank you any feedback guys. Regards, Juan Pablo Godoy.

2 4

[ERR] mod_lua.cpp:203 error loading module 'socket.core' from file '/usr/local/lib/lua/5.2/socket/core.so'
by Akshaya j 17 Nov '17

17 Nov '17

Hi team, I run fs-version 1.6.19+ in centos 7. I installed Lua5.2 and tried installing luasocket-3.0-rc1 from freeswitch doc. I can run independent lua script and send mail, but when I use the same script inside FS I get this error, ERR] mod_lua.cpp:203 error loading module 'socket.core' from file '/usr/local/lib/lua/5.2/socket/core.so': /usr/local/lib/lua/5.2/socket/core.so: undefined symbol: luaL_setfuncs stack traceback: [C]: ? [C]: in function 'require' /usr/local/share/lua/5.2/socket.lua:12: in main chunk [C]: in function 'require' /usr/local/share/lua/5.2/socket/smtp.lua:15: in main chunk [C]: in function 'require' /usr/local/freeswitch/scripts/sendmail.lua:2: in main chunk >From the freeswitch doc luasocket installation I can witness last make command with LUAINC_linux=/usr/local/src/freeswitch/src/mod/languages/mod-lua/lua where I don't find any folder "lua" inside the source. I have LUAINC_linux=/usr/local/include/ (lua include dir) and this works for independent luasocket to send mail. Please help me in fixing this workaround.

1 0

Any way to record only one leg of the call? (without using stereo recording)
by Björn Bylander 17 Nov '17

17 Nov '17

Hello, Is there any way, using FreeSWITCH, to record only one leg of a bridged call? I know about stereo recording and removing one channel but that's unfortunately not an option. The audio for the leg which mustn't be recorded must never be written to disk. This is to be able to comply with https://en.wikipedia.org/wiki/General_Data_Protection_Regulation. Regards, Björn Bylander

2 2

RxFax incomplete session
by Volodymyr Fedorov 16 Nov '17

16 Nov '17

Hi All, quick question in case of interruption of fax call are there any way to keep partially received file. What I noticed that RxFax seems like does not keep it. Thanks for any advice. -- Best regards, Volodymyr

2 1

fifo off-hook agent on multiple queues ?
by Khalil Khamlichi 16 Nov '17

16 Nov '17

Hi, In FIFO, Can we put an agent in off-hook mode in multiple queues so that he receives calls from multiple queues on same off-hook session ? Thanks in advance. Khalil

2 4