March 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Javascript call recording revisited
by Francesco Facco de Lagarda 10 Mar '18

10 Mar '18

'Evening one and all. Im going crazy trying to record calls from javascript scripting This is what I have done: I have installed mod_shout $ apt-get install freeswitch-mod-shout enable in /autoload ../modules.xml reloaded freeswitch $ service freeswitch restart Checked that module was loaded fs_cli -x "show modules" | grep shout And set it up before bridge: set "bridge_pre_execute_aleg_app record_session /tmp/test.mp3" Simplyfied code I have: if (session.ready()) { var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060") if (sessOut.ready()) { sessOut.execute("set", "bridge_pre_execute_aleg_app record_session /tmp/test.mp3"); bridge(session, sessOut); } } But nothing appears in "/tmp" !!!! Where am I going wrong?

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 141, Issue 12
by Melek Oktay 09 Mar '18

09 Mar '18

Hi again, I am planing to use mod_http_cache to request (there will be posting recording file also) music/mp3/vaw files nginx&php web server that stores these files to Cassandra ! Proof of concept study has already done, ~3.5 MB file is requested by FreeSwitch, and I log file say that [INFO] mod_http_cache.c: X file downloaded in 265 ms. Cassandra store these files in 1 MB of chunks.. My question is about idea!! Cassandra is the new topic for me.. Am I pushing forward these technologies, in order to achieve my goal.. Is there any other better/easy way to do it ? My goal is providing redundancy to my media server, multi FreeSwitch could read/write file to theses media server, no single point of failure (VRRP will be used for my new Webserver), Regards, Melek On Sat, Mar 3, 2018 at 1:00 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > Send FreeSWITCH-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of FreeSWITCH-users digest..." > > Today's Topics: > > 1. Re: Best way for storing Media for Freeswitch (Bilal Abbasi) > > > ---------- Forwarded message ---------- > From: Bilal Abbasi <bilaln018(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Sat, 03 Mar 2018 08:56:58 +0000 > Subject: Re: [Freeswitch-users] Best way for storing Media for Freeswitch > Hi, > I am using amazon S3 and mod_http_cache, works good for me. > > Regards > Abbasi > > On Fri, 2 Mar 2018 at 11:48 PM, Sergey Safarov <s.safarov(a)gmail.com> > wrote: > >> You can play media files from http server >> >> <action application="playback" data="http://example.com/media/hello_world.wav"/> >> >> >> пт, 2 мар. 2018 г. в 19:57, Emrah Ciftcibasi <emrahciftcibasi(a)gmail.com>: >> >>> Hi, >>> >>> I need a solution to store and read media files from Freeswitch servers >>> (I have multiple Freeswitch). >>> Media files are recorded voicemails, recorded calls and user files like >>> welcome sounds, >>> IVR sounds, waiting music, etc. >>> >>> Currently this is solved by using a mount to a SMB share (NFS) on local >>> storage. This works good but I think it has a drawbacks like it is not >>> scalable.. >>> >>> I wonder, what kind of solution exist in the literature ? What are the >>> best practices that you face? >>> >>> Best Regards, >>> Melek >>> >>> P.S. At first a solution with a distributed file system was tried with >>> GlusterFS. This had some drawbacks, especially with high latency >>> synchronization between nodes. This approach was dropped. >>> ____________________________________________________________ >>> _____________ >>> 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 >> >> _________________________________________________________________________ >> 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 > > > _______________________________________________ > 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 > >

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Custom Headers with mod_callcenter
by Ari Sonesh 08 Mar '18

08 Mar '18

Can anyone advise when using the mod callcenter ... how to pass custom sip headers from the calling party call leg to the agent call leg? thanks! *Ari Sonesh *

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NAT / UDP hole punching issue
by Tihomir Culjaga 07 Mar '18

07 Mar '18

hi, I have "no audio" issue with TLS and i hope someone could help as Im getting crazy ... literally :( my setup is like this: Phone <> NAT <> INTERNET <> NAT <FreeSWITCH> FreeSWITCH version: 1.6.12~64bit ( 64bit) I have a separate profile configured for TLS: <param name="rtp-ip" value="192.168.100.60"/> <param name="sip-ip" value="192.168.100.60"/> <param name="apply-nat-acl" value="rfc1918"/> <param name="ext-sip-ip" value="stun:stun.freeswitch.org"/> <param name="ext-rtp-ip" value="stun:stun.freeswitch.org"/>  <param name="tls-only" value="true"/> <param name="tls-sip-port" value="15061"/> ================================================================================================= Name tls-public Domain Name N/A Auto-NAT false DBName sofia_reg_tls-public Pres Hosts 192.168.100.60,192.168.100.60 Dialplan XML Context public Challenge Realm auto_from RTP-IP 192.168.100.60 Ext-RTP-IP stun:stun.freeswitch.org SIP-IP 192.168.100.60 Ext-SIP-IP 85.114.41.180 TLS-URL sip:mod_sofia@85.114.41.180:15061 TLS-BIND-URL sips:mod_sofia@85.114.41.180:15061 ;maddr=192.168.100.60;transport=tls WS-BIND-URL sip:mod_sofia@192.168.100.60:5066;transport=ws WSS-BIND-URL sips:mod_sofia@192.168.100.60:7443;transport=wss HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA CODECS OUT PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG true PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 2 REGISTRATIONS 0 i manage to register the phone with no problems but when i call the phone i get no audio; bgapi expand originate ${sofia_contact(tls-profile/agent2/ nexios(a)192.168.100.60)} &echo() FS sends the invite as: SDP in INVITE message from FS v=0 o=FreeSWITCH 1519708899 1519708900 IN IP4 *85.114.41.180* s=FreeSWITCH c=IN IP4 *85.114.41.180* t=0 0 m=audio *17480* RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 SIP Client responds with: SDP in 200 OK from the client v=0 o=- 3728718779 3728718780 IN IP4 *213.147.96.240* s=pjmedia b=AS:84 t=0 0 a=X-nat:0 m=audio *4002 *RTP/AVP 8 101 c=IN IP4 213.147.96.240 b=TIAS:64000 a=rtcp:4003 IN IP4 *213.147.96.240* a=sendrecv a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 So the UDP stream is: client( *4002 * ) <> ( *17480* )FS when i sniff the traffic (on both sides client/FS) using wireshark, i see RTP packets ( src:4002, dst:17480 ) leaving the client towards FS, but I don't see RTP packets ( src:17480, dst:4002 ) from FS side leaving towards the client. so my question, of course, is why FS is not sending RTP packets to the IP:PORT notified in 200 OK SDP from the client ? Did i miss some needed configuration ? in FS logs i see *192.168.100.60 port 17480 -> 213.147.96.240 port 4002* but nothing is actually being sent out from FS 2018-02-27 11:13:02.733376 [DEBUG] sofia.c:6965 Channel sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 entering state [ready][200] 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4311 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1] 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4366 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4227 Set telephone-event payload to 101@8000 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:3021 Set Codec sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 PCMA/8000 20 ms 160 samples 64000 bits 1 channels 2018-02-27 11:13:02.733376 [DEBUG] switch_core_codec.c:111 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Original read codec set to PCMA:8 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4572 Set telephone-event payload to 101@8000 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:4631 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Set 2833 dtmf send payload to 101 recv payload to 101 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6588 AUDIO RTP [sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551] *192.168.100.60 port 17480 -> 213.147.96.240 port 4002 *codec: 8 ms: 20 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:3875 Starting timer [soft] 160 bytes per 20ms 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6806 Activating RTCP PORT 4003 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:4261 RTCP send rate is: 5000 and packet rate is: 20000 Remote Port: 4003 2018-02-27 11:13:02.733376 [DEBUG] switch_rtp.c:2534 Setting RTCP remote addr to 213.147.96.240:4003 2 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6887 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Set 2833 dtmf send payload to 101 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6894 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Set 2833 dtmf receive payload to 101 2018-02-27 11:13:02.733376 [DEBUG] switch_core_media.c:6917 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Set rtp dtmf delay to 40 2018-02-27 11:13:02.733376 [NOTICE] sofia.c:8023 Channel [sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551] has been answered 2018-02-27 11:13:02.733376 [DEBUG] switch_channel.c:3770 (sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551) Callstate Change RINGING -> ACTIVE 2018-02-27 11:13:02.753388 [DEBUG] switch_ivr_originate.c:3686 Originate Resulted in Success: [sofia/tls-public/sip:agent2/ nexios@213.147.96.240:10551] 2018-02-27 11:13:02.753388 [INFO] switch_channel.c:3127 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Flipping CID from "" <0000000000> to "Outbound Call" <nexios> 2018-02-27 11:13:02.753388 [DEBUG] mod_commands.c:4788 (sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551) State Change CS_CONSUME_MEDIA -> CS_EXECUTE 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:584 (sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551) Running State Change CS_EXECUTE 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:650 (sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551) State EXECUTE 2018-02-27 11:13:02.753388 [DEBUG] mod_sofia.c:198 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 SOFIA EXECUTE 2018-02-27 11:13:02.753388 [DEBUG] switch_core_state_machine.c:328 sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 Standard EXECUTE EXECUTE sofia/tls-public/sip:agent2/nexios@213.147.96.240:10551 echo() Regards, Tihomir.

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Media storage for Freeswitch
by Melek Oktay 06 Mar '18

06 Mar '18

Hi, I need a solution to store and read media files from Freeswitch servers (I have multiple Freeswitch). Media files are recorded voicemails, recorded calls and user files like welcome sounds, IVR sounds, waiting music, etc. Currently this is solved by using a mount to a SMB share (NFS) on local storage. This works good but I think it has a drawbacks like it is not scalable.. I wonder, what kind of solution exist in the literature ? What are the best practices that you face? Best Regards, Melek P.S. At first a solution with a distributed file system was tried with GlusterFS. This had some drawbacks, especially with high latency synchronization between nodes. This approach was dropped.

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Internal Sip profile error
by Social Boh 06 Mar '18

06 Mar '18

Hello list, I'm experimenting a "strange" issue with FreeSWITCH. If I Start Freeswitch from command line: *freeswitch -nonat -nf* IPv4 internal profile working fine: * sofia status* Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) 45.77.161.19 alias internal ALIASED external profile sip:mod_sofia@45.77.161.19:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG internal-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) internal profile sip:mod_sofia@45.77.161.19:5060 RUNNING (0) ================================================================================================= If I use script daemon *systemctl start freeswitch* with this parameters: *-nonat -nf -nc* I have this error: Error Creating SIP UA for profile: internal (sip:mod_sofia@45.77.161.19:5060;transport=udp,tcp) The likely causes for this are: 1) Another application is already listening on the specified address. 2) The IP the profile is attempting to bind to is not local to this system sofia status Name Type Data State ================================================================================================= external-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5080 RUNNING (0) external profile sip:mod_sofia@45.77.161.19:5080 RUNNING (0) external::example.com gateway sip:joeuser@example.com NOREG internal-ipv6 profile sip:mod_sofia@[2001:19f0:9002:11f6:5400:1ff:fe64:1503]:5060 RUNNING (0) ================================================================================================= I compiled 1.6 version from source (git) on CentOS 7.4. Remote VPS with Public and Private IPv4 Any Hint? Thank you Regards -- --- I'm SoCIaL, MayBe

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[mod_conference] playback .wav after x seconds of silence
by Mititelu Stefan 06 Mar '18

06 Mar '18

Hi, Is there a way to playback a random .wav message after silence threshold has been kept for more than X seconds, in a conference? Thank you, Stefan

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pgsql core and odbc
by Madovsky 06 Mar '18

06 Mar '18

Hi, I noticed that if I use pgsql core protocol in switch.conf.xml (pgsql://) to a remote postgresql through VPN ps shows an idle connection taking between 15 and 50% of cpu. with odbc there is no such kind of issue. jira or a special setting needed? thanks

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(no subject)
by Ali Haider 05 Mar '18

05 Mar '18

[ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error] Usage: fs_cli [-H <host>] [-P <port>] [-p <se how you can ressolve this error kindly tell me please Sent from my iPhone

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Re: [Freeswitch-users] (no subject)
by Raimundo Pérez Nieves 05 Mar '18

05 Mar '18

Try this: Go to path /usr/bin and execute ./freeswitch It starts FS in secure mode, and you can debug the error. In my case, it was bad close tag in html and I just shutdown FS with that error, so It did not open. Doing this gave the idea of the problem. Hope solves Enviado desde mi iPhone El 05/03/2018, a la(s) 17:52, Ali Haider <alihaider.4189(a)gmail.com <mailto:alihaider.4189@gmail.com>> escribió: > > [ERROR] fs_cli.c:1659 main() Error Connecting [Socket Connection Error] > Usage: fs_cli [-H <host>] [-P <port>] [-p <se > how you can ressolve this error kindly tell me please > Sent from my iPhone > >

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