March 2018 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Debian 9 and Freeswitch 1.8
by William Colls 11 Jul '18

11 Jul '18

Is there an expected date when Freeswitch 1.8/Debian 9 will become the preferred production configuration?

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Sending SIP Messages from LUA
by Ronnie Beck 19 Jun '18

19 Jun '18

Hi All, I am trying to write a LUA script which will slot into a chatplan whose goal is to simple take a simple text message (SIMPLE MESSAGE) and then distribute it to multiple recipients. I wrote a script which takes some arguments (sender, recipieant list and the body) then would generate new SIP Messages to be sent to each of the recipient. The code I use to generate the message (which is more or less the example given in mod_sms for sending an SMS, not exactly what I want to do): local event = freeswitch.Event("CUSTOM", "SMS::SEND_MESSAGE"); event:addHeader("proto", "global"); event:addHeader("dest_proto", "sip"); event:addHeader("from", source); event:addHeader("to", domain.."/"..recipient); event:addHeader("type", "text/plain"); event:addHeader("skip_global_process", "true"); event:addBody( body ); event:fire(); This gives the error: [WARNING] sofia_presence.c:221 Not sending to local box for XXXX(a)10.11.12.13 [ERR] sofia_presence.c:272 Chat proto [sip] from [<sip:YYYY@10.11.12.13>;tag=uUQudJKEl1SKjq8wdjqwi68768] to [XXXX(a)10.11.12.13] Some message text Nobody to send to: Profile internal But the message actually arrives successfully. This is fine but I would prefer not to have lots of error messages. So I tried to find a more correct way to do this. I tried the following (based on what I could find in freeswitch 1.6.20 source code): local event = freeswitch.Event("SWITCH_EVENT_SEND_MESSAGE"); event:addHeader("user", recipient ); event:addHeader("host", domain ); event:addHeader("profile", domain ); event:addHeader("subject", "SIMPLE MESSAGE"); event:addHeader("content-type", "text/plain"); event:addHeader("from", source); event:addBody( body ); event:fire(); But this gives the same error. How can I send a simple SIP Message with LUA to a locally registered user? Many thanks, Aaron

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Re: [Freeswitch-users] stream a file multicast with mod_esf
by Sluschny, Thomas 10 May '18

10 May '18

Hi Brian, i thought if i can stream from portaudio it is almost the same with streaming from file, so it should working already now. Is this not the design idea of channels and media to do so? regards, thomas PS: sry for improper formatted mail, i cant reply at the moment and have to copy mail from archive :( ________________________________ >From brian at freeswitch.org Thu Feb 12 07:07:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 12 Feb 2009 09:07:36 -0600 Subject: [Freeswitch-users] stream a file multicast with mod_esf Message-ID: <6F68FEDB-8976-44D4-9A67-F871EC7D5120(a)freeswitch.org <mailto:6F68FEDB-8976-44D4-9A67-F871EC7D5120@freeswitch.org> > esf is for multi cast paging... it currently won't let you play files... we would have to create a multicast playback application. /b On Feb 12, 2009, at 8:00 AM, Sluschny, Thomas wrote: > Hi, > > i want to stream a file per IP multicast with mod_esf. > > I can stream IP multicast with: > pa call stream XML > and in XML dialplan: > <extension name="stream"> > <condition field="destination_number" expression="^stream$"> > <action application="answer"/> > <action application="esf_page_group" data="224.168.168.168 34567"/> > <action application="esf_page_group" data=""/> > </condition> > </extension> > and i can also play files with 'playback' app, > > BUT: how can put these 2 things together? > > May be its trivial, but i cant get it make working with 'originate' > or 'uuid_broadcast' or 'bridge', 'transfer' and so on ... > > Thanks in advance, > Thomas

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Voicemail via web interface
by John 24 Apr '18

24 Apr '18

Hello, Can you point me to any additional information about the voice mail via web interface? I have it up and running; however if you click the play button there is no playback, if you click download it will play in MS media player. Thanks John

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mod_conference scalability
by Brian 19 Apr '18

19 Apr '18

Hi, I'm new to FreeSWITCH and I'm testing the scalability of mod_conference to see if it will scale better that other solutions. My scenario is to have one speaker, and many listeners (mute). Since I have only one speaker, I was expecting this to scale well because there is no audio mixing required, just send each frame of the single speaker to each listener. Unfortunately, my testing was disappointing, and it didn't scale nearly as well as I'd hoped (based on what I've read on how FreeSWITCH is supposed to be generally very scalable). Here's my server setup is this: FreeSWITCH 1.0.4, 64 bit CentOS 5.3, on a quad-core Xeon server, 4 Gig of RAM. I've set file logging to "notice" level. My conference profile is configured to suppress several events, hoping that it would improve performance. Here are a few scenarios I tested, and roughly where I reached the point of audio failure on the conferences: Scenario 1: 1 conference, 1 speaker, audio failed at approx 300 listeners (mute) Scenario 2: 4 conferences, 1 speaker per conference, audio failed approx 110 listeners per conference (so just over 400 total channels on the system). Scenario 3: 16 conferences, 1 speaker per conference, audio failed at 32 listeners per conference (so just over 500 total channels on the system). Looking at the output from "top", it seems that in all 3 scenarios, the audio quality failed when the % CPU for the FreeSWITCH process exceeded 300%. I was hoping maybe someone else might have done similar testing, or maybe has suggestions on how to improve the performance. Or perhaps an alternate solution to the one speaker, many listener case? Thanks, Brian.

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Send RTP to external server
by Mickael Hubert 18 Apr '18

18 Apr '18

Hi list, I want to record each call through freeswitch. But i want record only caller (SSRC 1) OR callee (SSRC 2) voice (not both). I read about SIPREC, Jack, etc ... not interesting Do you have a idea for me please ? Thanks in advance

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how to access the XML_REQUEST in mod_python
by Do Nguyen Ha 09 Apr '18

09 Apr '18

hi list i want to use mod_python to dynamic load user directory and read the wiki but no documents for access XML_REQUEST how to get the XML_REQUEST in python script thank you

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Adding a Parked Call into a conference
by Bob McCarthy 07 Apr '18

07 Apr '18

Anyone have a best practice for taking a channel out of a parking lot and adding it to a conference? I have successfully transferred into the parking lot but the results are less than desired. The BLF remains active until the conference ends and pressing it give you hold music. Looking for an API / lua solution. Thanks in advance, Bob McCarthy

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Extract variables from 300 redirect
by Joseph Waite 04 Apr '18

04 Apr '18

Hi Guys I am wondering if its possible to extract a custom variable from a 300 Redirect response from a billing system? Effectively I am trying to add a custom variable to the 300 message to give a maximum call length to then use in the dial plan for a Scheduled Hangup. Any pointers of how to do this or the magic words to search confluence/google with? Regards

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Using lcr in lua session and fail_on_single_reject problem
by Mimiko 04 Apr '18

04 Apr '18

Hello. I want to originate a call using lcr from lua script: freeswitch.Session("{ignore_early_media=true,originate_timeout=15,fail_on_single_reject=^^:NO_ANSWER:CALL_REJECTED:NORMAL_CLEARING:USER_BUSY}lcr/default/"..number) Using this command, originate_timeout is not taken into account and all routes are tried regardless of response reason, for example NO_ANSWER. If I remove fail_on_single_reject, then originate_timeout works but same problem: all routes are tried, except when the call was answered. In diaplan <action application="lcr" data="${destination_number} default"/> <action application="bridge" data="{fail_on_single_reject=^^:NO_ANSWER:CALL_REJECTED:NORMAL_CLEARING:USER_BUSY}${lcr_auto_route}"/> this works fine. Maybe its due to using bridge, not lcr directly. But in lua ${lcr_auto_route} variable is not available. Any suggestion? FreeSWITCH Version 1.5.6b+git~20130928T022323Z~6b9382290d (git 6b93822 2013-09-28 02:23:23Z) Yes, it is old, but can't update at this time.

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