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Shared Presence Sofia profile config
by Nandy Dagondon 29 Sep '14

29 Sep '14

Hello, I'm trying to monitor PRESENCE of remote extension phones. In this wiki https://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Presence it says - - - - - *manage-presence* "Enable presence. If you want to share your presence (see dbname and presence-hosts) set this to "true" on the first profile and enable the shared presence database. Then on subsequent profiles that share presence set this variable to "passive" and enable the shared presence database there as well. - - - - - - If i have the ff profiles: doublenat.xml internal.xml which one is the "first profile" that I should set manage-presence=true? Does parsing of XML config files in alphabetical order? Tks. /Nandy

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Export Var
by Andre 28 Sep '14

28 Sep '14

Can you export the var to the B leg using this code? api.ExecuteString("uuid_setvar ".. Would the syntax be Usage: uuid_setvar <uuid> Export <varname> [value]

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BLF Lamps flashing orange
by Cj B 28 Sep '14

28 Sep '14

Hello list, I have a Cisco 525G and 504G both with side cars. They were working fine for a while (green lights, flash when ringing, etc) and then all of a sudden they are now just blinking orange constantly. I haven't upgraded freeswitch (1.4.6+git~20140603T193516Z~9479729340~32bit). The sidecar still seems to work, I can transfer to other lines, I can dial them, but the lights just stay flashing orange. Here's a packet capture: http://pastebin.com/vDreS2bm Does anyone know what could cause this? Thanks for any help!

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mod_v8 vs mod_lua overhead?
by Avi Marcus 28 Sep '14

28 Sep '14

Hi - I need to do an sql query with some logic and manipulation, and then route a call. I think it's a bit much to do with mod_odbc_query and the dialplan, so I'll be using mod_lua or mod_v8. Does anyone have information about the overhead about running a script in either language? I know mod_lua has been the goto scripting language for low overhead, but how does the newer mod_v8 perform? I'd rather use javascript, but only if there isn't a significant performance difference. Thanks, -Avi Marcus

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Mod_sofia registration records
by Carlos Ruiz Díaz 27 Sep '14

27 Sep '14

Hi all, is there any way for FS to consume registration records from a custom database, for example, location table of Kamailio? Or on a on-demand basis, making a Redis query for example. Thanks, -- Carlos http://caruizdiaz.com

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Mod_translate Issue
by Ave ebock 27 Sep '14

27 Sep '14

Hi, Please im trying to use mod_translate. I have this profile in my translate.conf.xml <profile name="NG"> <rule regex="^0([1-9]\d+)$" replace="234$1"/> </profile> How do i please apply it, such that before calls reach my dialplan, the number is translated accordingly. Any help would be highly appreciated Thank You

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Opus codec issue
by jonathan augenstine 26 Sep '14

26 Sep '14

I have been working with WebRTC and utilizing the opus codec. The sound quality on calls is great. I have been encountering some file issues however and want to get some input on how to best proceed. When the call is has both ends in Opus, or if Freeswitch is transcoding the call has no issues. The only issues I encounter are the following. 1. fs_encode fails to create a valid opus file When fs_encode executes, you do see mod_opus load successfully. However, the file that is created cannot be read by opus.org utilities, specifically opusinfo. The command line I use is: fs_encode -l mod_opus infile.wav outfile.opus 2. Streaming a file that is encoded in opus format fails If I encode a file using opus.org tools, opusenc, I have a valid opus file and I am able to open it with other audio applications. If I call into freeswitch and try to stream out that file, there is no error reported in the console logs, but the audio heard is garbled. Basically the mod_opus appears to work but reading, writing files appears to have an issue. My questions. 1. Is there anything I might have configured incorrectly? 2. What information can I grab to help troubleshoot this issue? Jonathan

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FreeSWITCH Friday FreeForAll Reminder!
by Ken Rice 26 Sep '14

26 Sep '14

FreeSWITCHers, Do not forget to join us at 2PM CST for the FreeSWITCH Friday FreeFor All Visit http://ift.tt/1n3h0Pf and Click Call 888 with your WebRTC enabled Browser and headset, Call sip:888@conference.freeswitch.org or see http://ift.tt/1prwIZL for access info! -- Ken FreeSWITCH.org ClueCon.com OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH @ClueCon

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is there any way to apply mod_shout "volume" setting on a per stream basis?
by Nathan Neulinger 26 Sep '14

26 Sep '14

i.e. in the param section of a local_stream? It doesn't look like anything in place that would allow this - is it something that might reasonably be possible to add, or would it be involved? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Re: [Freeswitch-users] Internal Sip Profile public or default context?
by Matthew Shealy 26 Sep '14

26 Sep '14

My Freeswitch is hosted on an external vps. Would I want to put the Internal profile behind the firewall and not allow connections to it. Or since I will be using Bria to make and receive my calls would I open it up and connect my registered users that way? - Matthew ---------- Forwarded message ---------- > From: Michael Collins <msc(a)freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 22 Sep 2014 14:03:10 -0700 > Subject: Re: [Freeswitch-users] Internal Sip Profile public or default > context? > This would be the default context to use for any unauthenticated calls. In > other words, that context parameter is telling the Internal SIP profile > which dialplan context to use when an unauth'd call comes in. Typically > this would be inbound SIP provider calls, such as calls in to a DID number. > You definitely want unauth'd calls to be handled by your > public/untrusted/paranoid dialplan context. To paraphrase Brian West: > "paranoid by default" is a good policy. If the user is authenticated then > the user_context value is used. > > -MC > > On Sat, Sep 20, 2014 at 6:21 PM, Matthew Shealy <shealyw2(a)gmail.com> > wrote: > >> Hi guys, >> >> In the documentation it says: >> >> "Authenticated users of this profile are normally set (via the variable >> user_context) to use the dialplan defined in the default context." >> >> >> https://confluence.freeswitch.org/display/FREESWITCH/Configuring+FreeSWITCH >> >> So why after a fresh compile is the $conf/sip_profiles/internal.xml >> Profile set to <param name="context" value="public"/> >> >> >> Thanks, >> Matthew >> >> _________________________________________________________________________ >> 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-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.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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