- FreeSWITCH-users - FreeSWITCH Mailing Lists

SRTP Encryption
by NOx-WHV 01 Sep '09

01 Sep '09

Hi, i have a problem using SRTP Encrytion. All intern calls are SRTP encrypted. Some of my Gateway don´t support SRTP encryption. In my dialplan I now set the sip_secure_media to false. <action application="set" data="sip_secure_media=false"/> It works. But is there any chance to encrypt the call on one side and use a unencrypted call on the other side of the freeswitch? Phone -->(SRTP)--> Freeswitch -->(RTP)--> Gateway Thanks for help NOx -- View this message in context: http://www.nabble.com/SRTP-Encryption-tp25237296p25237296.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Can an Instant Message (IM) be sent to a cell phone with FS under Windows?
by Merle J. Ebbert 01 Sep '09

01 Sep '09

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Load Balacing
by Juan Manuel Vicente 01 Sep '09

01 Sep '09

Hello: I am using two boxes, with same domain, to register about 1000 users working together a DNS box doing round robin to resolve two differentes Ips one to each box. This two boxes are using the same database (through unixodbc). When I use the funcion sofia_contact or sofia profile status both boxes show the status of the user. Does not matter where the resgistration was did. But when I try to call the user this call fails, the call only works if the call is sent through the same box where the user is registered. Any ideas where i need to looking for ? Thx in Advance.

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FreeSWITCH HA + Loadbalancing
by Raimund Sacherer 01 Sep '09

01 Sep '09

Hello List, I have read the current thread about scalability and I would need some advice about a callcenter setup: First where I come from: I have lot's of problems with an asterisk solution. I have regular crash's and lock-ups, with downgrading and other stuff i got it somewhat stable, but have nevertheless regular hickups. I am desperate and want to get rid of asterisk and I hope that freeSwitch will provide me with a more stable solution. Our Setup (really nothing special): * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3 GIG of RAM, Xircom analog switch connected to mobile stations for 4 different providers, Digium 4port cards TP400<something> * 8 queues * ~60 agents (which logon, logoff, pause, unpause), not more than 40 concurrently online * ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx) * Music on Hold in the call-queues * No special announcement * Transfers between calls in queues and different agents as well as non agents (i mention this because we have transfer related chrashes in asterisk) The current Problems: * Lockups with different causes (ranging from calls not terminated to heavy thread locking through the AMI interface) * Crashes and library aborts (pthread, libc, crashes related to music on hold, app_queue, transfers) We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are now back to 1.4.21.2 (stock debian) as anything beyond that is for whatever reason highly unstable for our szenario. Maybe we should have been segmenting the box into one asterisk dedicted to talking to the hardware, one especial for queue/sip handling, i do not know. (all issues are well documented in issues.asterisk.org, but it seems to be very, very difficult to get to the bottom of them as they exist since 1.4.23 as it seems and are open until know and not fixable since month.) Now, I really would appreciate some success-stories on how you guys managed to get a stable pbx system with freeSWITCH in regard of HA and scalability: * How to segment freeSWITCH? Or is it stable enough to handle all in one for such a szenario as outlined above? * What would be the best strategie for High Availability / Failover? -> I read in the WIKI (featurelist) that Livemigration of calls from one box to another should be possible? -> I was thinking about using memcached for storing all state information so another freeswitch box can takeover calls from the first box if it dies, is this possible? If so, how? -> Is there anotherway to somehow configure freeSWITCH that in the event of a crash i do not loose the current established calls? Basically I just want a stable PBX where I do not have to fear every day it will core-dump or abort or Lock up. Is freeSWITCH mature enough so i can sleep at night for at least 3 month without a crash? Thank you for your Time and help in advance, and I am more than willing to take all the information gathered here and create a wiki page to help other people with the same questions/problems. best Ray -- Raimund Sacherer - RunSolutions Open Source It Consulting - Parc Bit - Centro Empresarial Son Espanyol Edificio Estel - Local 3D 07121 - Palma de Mallorca Baleares

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Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
by vmorales 31 Aug '09

31 Aug '09

By "./compile" I was referring to "./configure" Vladimir -----Original Message----- From: vmorales [mailto:email.list.subscriber@gmail.com] Sent: Monday, August 10, 2009 11:49 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86 Thanks for the response(s): I ran the "./compile" script with a set PREFIX. This took a few attempts with errors before it was able to complete error-free, as I had to install libtool. Since then, I have tried running 'make', 'gmake', and '/opt/gnu/bin/make', but each results with an error. This is the error when running 'make' or 'gmake': <snip> make: Fatal error: Command failed for target `all-recursive' Current working directory /home/vmorales/freeswitch-1.0.4 *** Error code 1 make: Fatal error: Command failed for target `all' </snip> This is the error when running '/opt/gnu/bin/make': <snip> make[5]: *** [mod_amr.so] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_amr-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /opt/gnu/bin/make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 </snip> I re-untar'd before each compile/make attempt. Let me know if this is something that I can resolve. Vladimir -----Original Message----- From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Saturday, August 08, 2009 12:37 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86 This is not currently a supported platform, it only builds on 64 bit right now I think on solaris. Mike On Aug 6, 2009, at 6:03 PM, vmorales wrote: > Hello, > > Does anyone have, or know where to get, a pre-compiled copy of > FreeSwitch for Solaris 10/x86? _______________________________________________ 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-use rs http://www.freeswitch.org

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Can I stream a file to a parked call.
by Phillip Jones 31 Aug '09

31 Aug '09

Hi there, I know there are other ways of doing this. I am just trying get to know have fun with the FreeSWITCH API. I am using originate and park a call: fsApi.Execute("originate", string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2} &park", blegSession.Uuid, OutgoingCallerID, NumberToDial)); That works great, the phone rings. I want to play something to this called party when they pick up. while (!blegSession.answered()) { Log.WriteLine(LogLevel.Alert, "Inside::CallReturns:!Session.answered::Loop"); blegSession.sleep(500, 1); } string promptFile = "prompts/whisper.wav"; blegSession.StreamFile(promptFile, 0); This works - but the audio is choppy and slow. Is there something I need to do to that parked call before streaming that file? Thanks for any input. Phil

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T.38 with V.34 support
by Steve Underwood 31 Aug '09

31 Aug '09

Hi, Does anyone have access to a system that supports T.38 FAXing at V.34 rates. These things seem to be as rare as bugless software. Steve

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prompt playback volume using mod_shout
by Marc Orenberg 31 Aug '09

31 Aug '09

Hi, I'm finding that when I record a caller's voice using mod_shout in MP3 format, the recording is too soft when I play it back. I see there's a "volume" setting in shout.conf.xml, but it says, "Dont change these unless you are insane". When I insanely try to increase this value, I don't here any difference in volume. Is this the right knob, or is there some other way to do this? Thanks, Marc

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some errors from FS startup
by Erwin Huang 31 Aug '09

31 Aug '09

Hi, I was able to build the FS successfully and make a couple of test calls. But when I looked at the startup logs, I saw some errors. Please point out how to fix them. Thanks, Regards, e 2009-08-31 06:59:55.54063 [ERR] mod_xml_curl.c:404 Binding has no url! 2009-08-31 06:59:55.54094 [CRIT] switch_loadable_module.c:871 Error Loading modu le /usr/local/freeswitch/mod/mod_xml_curl.so **Module load routine returned an error** 2009-08-31 06:59:57.393133 [CONSOLE] mod_local_stream.c:142 Can't open directory : /usr/local/freeswitch/sounds/music/8000 2009-08-31 06:59:57.404228 [CONSOLE] mod_local_stream.c:142 Can't open directory : /usr/local/freeswitch/sounds/music/8000 2009-08-31 06:59:57.414708 [CONSOLE] mod_local_stream.c:142 Can't open directory : /usr/local/freeswitch/sounds/music/16000 2009-08-31 06:59:57.424568 [CONSOLE] mod_local_stream.c:142 Can't open directory : /usr/local/freeswitch/sounds/music/32000 // The errors below indicate the files donot exist. How to include these two files into FS make build? 2009-08-31 06:59:57.477374 [CRIT] switch_loadable_module.c:871 Error Loading mod ule /usr/local/freeswitch/mod/mod_java.so **/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No such file or directory** 2009-08-31 06:59:57.485242 [CRIT] switch_loadable_module.c:871 Error Loading mod ule /usr/local/freeswitch/mod/mod_flite.so **/usr/local/freeswitch/mod/mod_flite.so: cannot open shared object file: No suc h file or directory**

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Re: [Freeswitch-users] Troubles build with mod_cepstral
by Pete Mueller 31 Aug '09

31 Aug '09

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