February 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Kernel Hyperthreading ON or OFF?
by Raimund Sacherer 06 Oct '16

06 Oct '16

Hi, we have a new Freeswitch cluster with Dell R410. It has 2 CPUs (Xeon E5620, QuadCore. it's 8 cores withouth HT and 16 cores with HT. I googled with hyperthreading and freeswitch but some say turn it on it will give you a boost, others say turn it off, you will gain performance. Does anyone know if it's better to leave it on or turn it off? And some small explanation? I would be very gratefull for your help, best regards Raimund --

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freeswitch 1.6 without video support
by Grant Bagdasarian 19 Aug '16

19 Aug '16

Hello, Is it possible to compile and install fs 1.6 without support for video? Or is the 1.4 version the same as 1.6 with bug fixes and security updates? Regards, Grant

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Re- End Lua script after HangupHook handled without all the extra code to handle the return to the function
by Andrew Keil 08 Jul '16

08 Jul '16

To FreeSWITCH Users, See below for a sample template for a Lua Service Script running inside FreeSWITCH. The issue I have is fairly straightforward. I need a function to run when hangup is detected (ie. at the end of the call) however I understand this must not delay ending the script. This function is CleanUp(). Then I would like the service to end. The problem I am having is if the caller hangs up during the playback of "intro.wav" (as shown inside the MainService() function below), then the code jumps to the myHangupHook which calls CleanUp() perfectly, the issue is once CleanUp() is complete I would like the Lua script to end there and then (ie. at the bottom of CleanUp()). What actually happens is it returns to MainService() and continues to try and play "info.wav", unless I either check for session:ready() everywhere or add a goto as shown below under each streamFile() function call. My aim is to reduce extra code and to make the Lua script simpler and easier to read. Also I would like to try and avoid goto statements, which I know can be done with if (session:ready()) etc.... So is there a way to stop a Lua script running inside FreeSWITCH cleanly? I have tried the os.exit() this is barred from use by FreeSWITCH. I have also tried session:destroy() which crashes FreeSWITCH (version 1.6.5 on CentOS 6.7, CentOS 7 and windows) 100% of the time! I could look further into the Lua additions done by the FreeSWITCH team in the source code, however if someone has already solved this then that would be the best solution. FYI: Obviously the script below is simple, however I am sure that you understand if the script was complicated having to use "if (session:ready()) then ...." or "if (not session:ready()) then goto HANGUPEXIT end" makes the code ugly. Thanks in advance, Andrew Keil Visytel Pty Ltd ------------------------------------------------------------------------------ Sample Lua Service ----------------------------------------------------------------------- -- Lua template for FreeSWITCH service -- By: Andrew Keil (Visytel Pty Ltd) -- Email: support(a)visytel.com -- Setup script wide variables here function PreAnswer() freeswitch.consoleLog("INFO", "PRE ANSWER SECTION\n"); -- Add your pre answer code from here -- End of your pre answer code freeswitch.consoleLog("INFO", "PRE ANSWER SECTION COMPLETE\n"); end function AnswerCaller() session:answer() session:sleep(1000) end function MainService() freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION\n"); if (session:ready()) then -- Note (1): If you wish to end the call then simply use: goto ENDSERVICE -- Note (2): To terminate the service sooner when HANGUP is detected use: if (not session:ready()) then goto HANGUPEXIT end -- Add your main service code from here (caller would have been answered) session:streamFile("intro.wav") if (not session:ready()) then goto HANGUPEXIT end session:streamFile("info.wav") if (not session:ready()) then goto HANGUPEXIT end session:streamFile("outro.wav") if (not session:ready()) then goto HANGUPEXIT end -- End of your main service code end ::ENDSERVICE:: if (session:ready()) then -- End of service so hangup session:hangup() -- Should automatically jump to CleanUp() via hangup handler if caller still online at this stage end goto END ::HANGUPEXIT:: freeswitch.consoleLog("INFO", "END OF SERVICE (HANGUP DETECTED)\n"); ::END:: freeswitch.consoleLog("INFO", "MAIN SERVICE SECTION COMPLETE\n"); end function CleanUp() freeswitch.consoleLog("INFO", "CLEANUP SECTION\n"); -- Add your cleanup code from here (caller would have been disconnected) -- End of your cleanup code freeswitch.consoleLog("INFO", "CLEANUP SECTION COMPLETE\n"); end function myHangupHook(s, status, arg) session:hangup() CleanUp() -- Run CleanUp function now since the caller has disconnected end -- Setup Hangup event handler here v_hangup = "HANGUP" session:setHangupHook("myHangupHook", "v_hangup") -- Call service functions in order PreAnswer() AnswerCaller() MainService() -- End of Lua service

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Sharing presence between FS boxes
by Chris B. Ware 04 Jun '16

04 Jun '16

Hi, On advice by Brian and Anthony I'm writing to mailing list, after a jira (FS-6358). If I have two Freeswitch servers, sharing the same DB as backend, how should I configure presence such that if a phone send SUBSCRIBE to box A, and call is received on box B, I get Notify from box B? By now I've set manage_presence=true on both sip profiles (internal,external) and presence_hosts="_DISABLED_" as I read on wiki. Of course dial_string contains presence_id=${dialed_user}@${dialed_domain}. Sip registrations are correctly shared between boxes, and even presence works sometimes, but is not stable. Here my internal sip profiles config: <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="configuration" description="Various Configuration"> <configuration name="sofia.conf" description="Sofia Endpoint"> <profiles> <profile name="internal"> <domains> <domain name="all" alias="true" parse="false"/> </domains> <settings> <param name="debug" value="0"/> <param name="sip-trace" value="no"/> <param name="sip-capture" value="no"/> <param name="watchdog-enabled" value="no"/> <param name="watchdog-step-timeout" value="30000"/> <param name="watchdog-event-timeout" value="30000"/> <param name="log-auth-failures" value="true"/> <param name="forward-unsolicited-mwi-notify" value="false"/> <param name="context" value="default"/> <param name="rfc2833-pt" value="101"/> <param name="sip-port" value="$${internal_sip_port}"/> <param name="dialplan" value="XML"/> <param name="dtmf-duration" value="2000"/> <param name="inbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="outbound-codec-prefs" value="$${global_codec_prefs}"/> <param name="rtp-timer-name" value="soft"/> <param name="rtp-ip" value="$${local_ip_v4}"/> <param name="sip-ip" value="$${local_ip_v4}"/> <param name="hold-music" value="$${hold_music}"/> <param name="apply-nat-acl" value="nat.auto"/> <param name="apply-inbound-acl" value="domains"/> <param name="local-network-acl" value="localnet.auto"/> <param name="record-path" value="$${recordings_dir}"/> <param name="record-template" value="${caller_id_number}.${target_domain}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/> <param name="manage-presence" value="true"/> <param name="inbound-codec-negotiation" value="generous"/> <param name="tls" value="$${internal_ssl_enable}"/> <param name="tls-bind-params" value="transport=tls"/> <param name="tls-sip-port" value="$${internal_tls_port}"/> <param name="tls-cert-dir" value="$${internal_ssl_dir}"/> <param name="tls-version" value="$${sip_tls_version}"/> <param name="nonce-ttl" value="60"/> <param name="auth-calls" value="$${internal_auth_calls}"/> <param name="inbound-reg-force-matching-username" value="true"/> <param name="auth-all-packets" value="false"/> <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> <param name="rtp-timeout-sec" value="300"/> <param name="rtp-hold-timeout-sec" value="1800"/> <param name="presence-hosts" value="_DISABLED_"/> <param name="odbc-dsn" value="systemDB:freeswitch:pazzword"/> <param name="challenge-realm" value="auto_from"/> <param name="outbound-proxy" value="sip::5060"/> </settings> </profile> </profiles> </configuration> </section> </document> Can somebody help?

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Re: [Freeswitch-users] Presence with shared database
by Adam Kuśmirek 04 Jun '16

04 Jun '16

Hello, Yes we enabled manage-presence. According to documentation dbname is used to share presence between profiles. What we want to achieve is to share presence between hosts. We investigated some presence source code and found that there are sql queries that ask for specific FS host. We modified SELECT statements and UPDATES of version parameter, so they have no host in WHERE clause. After mod_sofia recompilation everything works fine. It seems that the only way to share presence between hosts is to set the same hostname in switch.xml for every FS node, but we cannot do this because we need host information for other services. I wonder why FS checks whether the subscription was made on a host that sets up the connection. Maybe this is a bug in the concept and it should be removed. Regards Adam Kusmirek ---------- Wiadomość przekazana dalej ---------- >> > From: "Ítalo Rossi" <italorossib(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Fri, 7 Aug 2015 22:42:37 -0500 > Subject: Re: [Freeswitch-users] Presence with shared database > Hard to say without more info, where's your configs? > > Did you enabled the manage-presence in your profile? and set the dbname? > > >

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BEARERCAPABILITY_NOTAVAIL with 1.6.6 on Debian 8
by Thomas Løcke 31 May '16

31 May '16

Hi all, Using 1.6.2 this works from fs_cli and dialplan: originate loopback/park/default &bridge(sofia/gateway/server/xxxxxxxx) Using 1.6.6 it fails with BEARERCAPABILITY_NOTAVAIL. Has something changed between 1.6.2 and 1.6.6 that may be the cause of this, and if so, what can I do to fix it? :o) Thomas

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FreeSWITCH using outbound proxy but to bypass media
by Muhammad Naseer Bhatti 28 May '16

28 May '16

I am trying to use an outbound SIP proxy for FreeSWITCH by adding fs_path to the bridge. It’s working fine and I am able to send calls using the proxy. On the proxy (which is also FreeSWITCH) I am receiving the call in a lua script, create a new session with the destination gateway and bridge two sessions (old, incoming and the newly created) The idea is to have multiple FreeSWITCH servers (all public IP address) originating calls while all calls going to the carrier via the proxy. The problem is that all media is also passing through the proxy. Is there a way I can only send signaling through proxy and media to flow directly between the endpoints? I have tried bypass_media and bypass_media_after_bridge but it’s not working. Is this even doable? Thanks. -- Muhammad Naseer Bhatti

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Fax....How I hate you.
by Blake Priddy 24 May '16

24 May '16

Is anyone out there having success with sending/receiving faxes with Flowroute? I have an SPA 112 and I am getting just the tops of the faxes. I have referred myself to this document https://wiki.freeswitch.org/wiki/SPA2102_T38_Howto with no success. I was just wanting to see what everyone is doing out there with flowroute for the people who still must use faxing.. -- *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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Using freeswitch as Generic SIP SBC/RtpProxy without predefined endpoints
by Taisto Qvist 14 May '16

14 May '16

Hi, I am an OpenSIPS/Kamailio user that has spent the whole day trying to figure out which software is most suitable to build an SBC that only acts as a full b2bua with rtp-relay, with an undefined number of destinations(entire domains with lots of unknown proxies) on each side. I've been looking at yate, freeswitch, SEMS, but I am overwhelmed by the gazillions of configuration-files that seem to be needed. I have absolutely NO need of pbx/user functionality, just a B2BUA and rtp-proxy capabilities(with full codec-transparency), with the addition that I must be able to add some custom logic to decide the DNS or IP(v6) address of the next-hop sip-node. All examples I've found on freeswitch SBC config, has predefined gateways/trunks/proxies as "predefined next-hops", and I want the SBC to be able to send anywhere, depending on custom application logic. I'd stick with kamailio/rtp-proxy and the standard topology-hiding module, but unfortunately I need the full topology hiding of a b2bua. Is this easily achievable with Freeswitch? Any help with be immensely appreciated! Regards Taisto Qvist

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Mastering FreeSWITCH Book
by Anthony Minessale 09 Apr '16

09 Apr '16

Hi, We've been asked by packt to produce a 3rd book called "Mastering FreeSWITCH" Would anyone be interested in working on the authoring of that book and get their name in print? We need help with creating the content as well as formatting the text into the word templates they require etc. Please contact consulting(a)freeswitch.org if you are interested. -- 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

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