March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

date format in FS log
by Madovsky 28 Mar '10

28 Mar '10

Is there any way to add date in FS log amd modify the format like asterisk [general] dateformat=%F %Tbecause I'd like to create a filterwith fail2ban (an app that scan logs with a filter)to avoid any brute force attacks.there's an asterisk example herehttp://www.voip-info.org/wiki/index.php?page_id=5348I would like to test it with FS, and if it works well so why not wikiRegardsFranck

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Which module is responsible for directory
by Vitalii Colosov 28 Mar '10

28 Mar '10

Hi, I am using mod_xml_curl to get users from database and I don't need ./conf/directory/*.xml at all. Is there is a module that I can unload (I would like to keep only needed modules). For example the module responsible for diaplan is "mod_dialplan_xml". Which one is for the users directory? Thank you, vIT

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new users in directory
by Madovsky 27 Mar '10

27 Mar '10

good evening, is it really necessary to do a "reloadxml" everytime a new user is created ? Thanks

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 45, Issue 280
by Juan Carlos Gonzalez Cardona 27 Mar '10

27 Mar '10

Hi, the problem is time limit for call to some extensions, not for all. 2010/3/23 <freeswitch-users-request(a)lists.freeswitch.org> > 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. Time limit (Juan Carlos Gonzalez Cardona) > 2. Re: Compile issue with openzap (Joseph L. Casale) > 3. Re: Time limit (Rupa Schomaker) > 4. Re: Compile issue with openzap (Brian West) > 5. donating german prompts (Christian L?schenkohl) > > > ---------- Mensaje reenviado ---------- > From: Juan Carlos Gonzalez Cardona <juancarlosg6(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Tue, 23 Mar 2010 15:14:42 -0500 > Subject: [Freeswitch-users] Time limit > > Hi everyone, is posible use call limite for calls, to some extensiones? > > In Asterisk is possible creating custom context with set time limit to > specific trunk, but i see that Freeswitch is more stable code, i need to > know if is possible create groups for limit calls and assign extensiones to > that group, if is possible, how i do that, thank´s. > > -- > Juan Carlos Gonzalez C. > > > ---------- Mensaje reenviado ---------- > From: "Joseph L. Casale" <jcasale(a)activenetwerx.com> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Date: Tue, 23 Mar 2010 21:01:20 +0000 > Subject: Re: [Freeswitch-users] Compile issue with openzap > >What is the output of the build when you type make? > > Ok, I didn't have access to that machine today, but on another > following my notes from previous builds, I got the same errors > again. > > I presume I missing a dep (CentOS 5 x64) but I compared my notes > with the wiki and they match. > > make (both stdout/stderr) are pastebin'ed: > http://pastebin.freeswitch.org/12529 > > Thanks! > jlc > > > > > ---------- Mensaje reenviado ---------- > From: Rupa Schomaker <rupa(a)rupa.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Tue, 23 Mar 2010 16:03:12 -0500 > Subject: Re: [Freeswitch-users] Time limit > for time limit, look at sched_hangup or sched_transfer > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_transfer > > If you want to limit the number of calls, look at mod_limit: > http://wiki.freeswitch.org/wiki/Mod_limit > > On Tue, Mar 23, 2010 at 3:14 PM, Juan Carlos Gonzalez Cardona < > juancarlosg6(a)gmail.com> wrote: > >> >> Hi everyone, is posible use call limite for calls, to some extensiones? >> >> In Asterisk is possible creating custom context with set time limit to >> specific trunk, but i see that Freeswitch is more stable code, i need to >> know if is possible create groups for limit calls and assign extensiones to >> that group, if is possible, how i do that, thank´s. >> >> -- >> Juan Carlos Gonzalez C. >> >> _______________________________________________ >> 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 >> >> > > > -- > -Rupa > > > ---------- Mensaje reenviado ---------- > From: Brian West <brian(a)freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Date: Tue, 23 Mar 2010 16:13:12 -0500 > Subject: Re: [Freeswitch-users] Compile issue with openzap > /usr/bin/ld: cannot find -ljpeg > > install libjpeg-devel and you should be set.. its not an OpenZAP compile > issue you're simply failing to pay closer attention to the errors. > > /b > > On Mar 23, 2010, at 4:01 PM, Joseph L. Casale wrote: > > >> What is the output of the build when you type make? > > > > Ok, I didn't have access to that machine today, but on another > > following my notes from previous builds, I got the same errors > > again. > > > > I presume I missing a dep (CentOS 5 x64) but I compared my notes > > with the wiki and they match. > > > > make (both stdout/stderr) are pastebin'ed: > > http://pastebin.freeswitch.org/12529 > > > > Thanks! > > jlc > > > > _______________________________________________ > > 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 > > > > > > ---------- Mensaje reenviado ---------- > From: "Christian Löschenkohl" <christian.loeschenkohl(a)xpirio.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Tue, 23 Mar 2010 22:38:10 +0100 > Subject: [Freeswitch-users] donating german prompts > hello > > we are going to produce a full set of german audio prompts and donate them > to this wonderful project. > i have attached a prompt list with the phrases we are going to record. > > the source of this prompt list is the file docs/phrase/phrase_de.xml. i > have also compared it with > the english file under docs/phrase/phrase_en.xml an merged this two + > translated what was needed/had > not been present in the german langunge file from english to german. > > i would be happy if someone could give me feedback on this and at best > review the phrase list in terms > of correctness and completeness. > > br > > > -- > Ing. Christian Löschenkohl > Technische Leitung, Forschung & Entwicklung VoIP > > xpirio > Telekommunikation & Service GmbH > Lakeside B04 > 9020 Klagenfurt > Austria > > T +43 (0) 5 77 11 - 1000 > F +43 (0) 5 77 11 - 1002 > E christian.loeschenkohl(a)xpirio.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 > > -- Juan Carlos Gonzalez C.

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Sunday, April 25th - FreeSWITCH Users Group Install Fest - San Francisco, CA
by Darren Schreiber 27 Mar '10

27 Mar '10

Hi folks, Just an update for ya... I will be hosting our first FreeSWITCH Users Group Install Fest on April 25th, 2010 in downtown San Francisco. I don't have a location for you yet, but please sign up if the time works for you so I can gauge how big a place I need to find. http://www.meetup.com/fsusers/ for details, which I'll copy for you here, too: Hi folks! Welcome telecomm enthusiasts, current Asterisk/FreeSWITCH users and anyone else who's interested in SIP, FreeSWITCH or any other VoIP services! This is an exciting time in VoIP - here's your chance to network with others and learn more about FreeSWITCH and what others are doing with it. We'll be having our first official meet-up on Sunday, April 25th 2010 at 4:00pm at a location downtown (TBD). We'll show you: * Installing FreeSWITCH * Setting up devices, trunks and general settings via XML * Using at least two cool features in FreeSWITCH (vote:) - mod_skypiax - Skype Integration - mod_event_socket - FreeSWITCH event sockets - mod_tts - Text to Speech Engine - mod_sofia - Advanced sofia w/ trunking/failover/etc. Please plan to bring a laptop so you can get the most from this experience with hands-on interaction between the organizer and your peers. The first 45 minutes of the event will be structured (so please try to be on time) and lead by the organizer, after which we'll share amongst each other. You'll also have the chance to network with other individuals who are working on similar projects. A location will announced shortly but let us first know if you can make the date! This is an independent event and is not sponsored by any other company nor officially endorsed by FreeSWITCH. It's more like a local users group. Thanks much, and welcome! - Darren Schreiber

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Multiplexing multiple rtp relaying incoming port
by Fabio Pietrosanti (naif) 27 Mar '10

27 Mar '10

Hi all, found our that UDP/53 port is prioritized on most mobile operators as DNS traffic. It would be really effective to be able to use only port 53 udp for incoming RTP traffic of FS while acting as Media Relay. I tried with: <param name="rtp-start-port" value="53"/> <param name="rtp-end-port" value="53"/> But i got error: 2010-03-27 15:54:57.566241 [WARNING] switch_core_port_allocator.c:74 Rounding odd start port 53 to 54 2010-03-27 15:54:57.566241 [WARNING] switch_core_port_allocator.c:78 Rounding odd end port 53 to 52 2010-03-27 15:54:57.566241 [CRIT] sofia_glue.c:712 No RTP ports available! 2010-03-27 15:54:57.566241 [ERR] sofia_glue.c:1203 sofia/internal/sip:546547426@109.112.56.148:39664 I/O Error If FS would be able to support a mode where it can receive multiple RTP streams coming from different peers on the same local port it would be very useful for mobile voip environment. Fabio

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session nb
by Jean-Yves F. Barbier 27 Mar '10

27 Mar '10

Hi list, On my old athlonXP-2600+, FS sessions max is 1000 & session limit is 30 (I guess connection per second?.) How does it translate in the real world? Is it going down a lot if I've got many transcoding channels (let's say iLBC internal transcoding to G711 external) I know it couldn't be a sharp answer, but I need to have an idea of what is possible (or not.) -- This fortune is false.

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Bad Video in latest SVN
by Yuriy Ivzhenko 27 Mar '10

27 Mar '10

Hi all, After upgrade from SVN 16013 to latest SVN 17109 Video quality become very bad. I'm use H264 video codec and Grandstream GXV3140 phones Anyone knows why this may happen?

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suggestions
by Jean-Yves F. Barbier 27 Mar '10

27 Mar '10

Hi, what is the best way to make requests for features &| suggestions? -- Given my druthers, I'd druther not.

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Attention all FreeSWITCH and Openzap committors. ACTION REQUIRED
by Michael Jerris 26 Mar '10

26 Mar '10

We are in process of our migration from svn to git for our main source control. We will likely still have a read only svn mirror, but all future commits will go to git after the migration. In order to proceed with this process, I will need your full name, public key, email address, and svn username. If you have outstanding svn branches that you would like to keep, please contact me with these details as well, I do not currently plan on merging any branches from svn to the new git repo. For information on how to use git: http://git-scm.com/documentation , http://git-scm.com/course/svn.html For information on how to generate a public key if you do not already have one: http://hkn.eecs.berkeley.edu/~dhsu/ssh_public_key_howto.html For general users, more information will follow on these changes. For now, nothing has changed. Around April 1, there will likely be a 1 time need to do a full checkout if you plan on continuing to use the svn mirror, or a fresh git checkout if you plan on switching to git. If you have any other questions about the process or changes, feel free to ask. Mike

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