February 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Nagios Plugin
by Kristian Kielhofner 28 Oct '10

28 Oct '10

Hello everyone, I've been looking for a FreeSWITCH Nagios plugin. Ideally I'd like to connect to the event socket and run some api commands and return them (as opposed to checking SIP, for example). I haven't found anything and I've started to write one in perl using ESL. I'm sure whatever I come up with is going to be pretty ugly and I'd much rather use something else. Has this been done? Thanks! -- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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Freeswitch Video Capture and Playback
by Esben Stien 18 Oct '10

18 Oct '10

I'm using ekiga with mod_fsv, trying to record and play back video. When I dial the record extension, it seems to record something, as the video file gets bigger. Trying then to dial the extension for play back, just hangs up, with freeswitch saying: od_fsv.c:247 File version does not match! There seems to be no information on the FSV format or the mod_fsv module on the wiki. Is this at all supposed to work?. What clients and codecs were successful?. Any pointers as to what I can try?. -- Esben Stien is b0ef@e s a http://www. s t n m irc://irc. b - i . e/%23contact sip:b0ef@ e e jid:b0ef@ n n

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Siptapi and Freeswitch
by Peter P GMX 10 Sep '10

10 Sep '10

Anybody tried siptapi with freeswitch? http://sourceforge.net/projects/siptapi/ This may enable Click2Dial e.g. from Outlook to Freeswitch. So anybody has experience with that solution? Best regards Peter

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error loading module 'luasql.mysql'
by Adam Wilt 02 Sep '10

02 Sep '10

Hi, I followed the instructions in the Lua documentation for setting up luasql, but when I try to run my script I get: 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so': /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment prot after reloc: Permission denied stack traceback: [C]: ? [C]: in function 'require' /usr/local/freeswitch/scripts/l.lua:2: in main chunk I'm running FreeSWITCH version 1.4 and luasql version 2.1.1. I changed the permissions for mysql.so and for my script to 777, so I'm not sure where the permission problem could be. I'd appreciate any suggestions. Thanks, Adam

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Stream audio file/live to multiple SIP endpoints with IP multicast
by Sluschny, Thomas 30 Aug '10

30 Aug '10

Hi, i want to stream audio (from file and live) from FreeSwitch to multiple SIP endpoints (these are proprietary). Because of bandwith limitations we have to use IP multicast for audio data. Now the question: how can i do this smartly in FreeSwitch? I have read some Wiki entries: - mod_esf: seams to use multicast, but i couldn't find usage information - mod_shout: streaming, but from internet ? MP3, but i want wav-files to stream - mod_local_stream: seams to play files in a loop, i want to play a whole file once. - mod_conference: seams not to support IP multicast Im searching for a simple way to send a SIP INVITE to all my endpoints and then stream audio (live from soundcard or from file) over IP multicast to all these endpoints. Thanks in advance, Thomas

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[contrib] a new java ESL inbound/outbound client
by david varnes 30 Aug '10

30 Aug '10

Hi all, I had a basic java inbound ESL client kicking around that I have used in a couple of small projects over the last year. I needed something a little more complete for a new project so I dusted it off and made it less incomplete. It still needs more work and testing, but it certainly is usable right now. I have tested it against FS 1.0.4 and latest trunk. I have put it in my contrib area in svn in hopes that some may find it useful: http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/dvarnes/java/esl-cli… I would be interested in any feedback ... Features * Apache License (ASL) version 2 * Standalone Inbound client * Framework classes to easily create an Outbound socket client * based on Netty [1] nio library version 3.1.5.GA (previously was using Apache MINA, but this is easier) * logging via slf4j * only dependencies are slf4j-api and netty (both Apache licensed) * single jar which is a valid OSGi bundle * built using maven * eclipse projects * reasonable level of java docs Still todo * Docs * Simple example apps * .. more in TODO.txt in project root. There is no binary jar available right now since I don't know how/if I can put files up to file.freeswitch.org. In the meantime to build you need maven [2] installed. If you are unfamiliar with maven usage, I can post a simple howto. davidv [1] http://www.jboss.org/netty/downloads.html [2] http://maven.apache.org

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Lua script hangup detect
by Chia-Yen Wu 10 Jun '10

10 Jun '10

Hello, i tried to use Lua script to replace xml macro in dialplan, but I found out that Lua wont terminate if client hangup, ,so the session is still on but client is already hangup, is there a way to avoid this ?

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Direct inward dialling
by Peter P GMX 07 Jun '10

07 Jun '10

Dear all, when I get an incoming call from my SIP provider I do receive an invite on the trunk number, e.g. in Germany 0xxxxx9. However I have an extension block of 0xxxxx90 to 0xxxxx999. In the dialplan I checked the condition field="destination_number" but this compares to the number of the trunk 0xxxxx9, so I do not get the direct inward dialling extensions. The SIP to-header contains the direct inward dialling number 0xxxxx90 to 0xxxxx999 but I have no success in accessing it in the dialplan. I tried sip_to_user, sip_to_uri, sip_req_user, sip_req_uri, but no success. They are all empty. Any idea which variable to parse? Best regards Peter

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Simple IAX2 setup - help with converting from asterisk to freeswitch
by Matthew Law 26 May '10

26 May '10

Hi, I have compiled freeswitch trunk from svn on debian. I am trying to convert the following simple working asterisk config to freeswitch and I would be really grateful if someone could point me in the right direction: pbx:/etc/asterisk# cat iax.conf [general] bindport=4569 bindaddr=my.ip.address jitterbuffer=yes disallow=all allow=ulaw allow=alaw context=deadend [voiptalk] type=peer username=XXXXXX secret=XXXXXX host=iax.voiptalk.org [0843XXXXXX] type=friend username=08433XXXXXX context=incoming requirecalltoken=auto [1000] type=friend host=dynamic mailbox=1000 secret=XXXXXX context=phones requirecalltoken=auto pbx:/etc/asterisk# cat extensions.conf [general] autofallthrough=yes [outgoing] exten => _0[1-9].,1,Dial(IAX2/XXXXXX@voiptalk/44${EXTEN:1}) exten => _00.,1,Dial(IAX2/XXXXXX@voiptalk/${EXTEN:2}) [internal] exten => 901,1,VoiceMailMain() exten => 901,2,Hangup() exten => 902,1,MeetMe(1234,cdM) exten => 902,2,Hangup() [incoming] exten => XXXXXXXX,1,Dial(IAX2/1000,30) exten => XXXXXXXX,2,VoiceMail(1000@internal) [phones] include => internal include => outgoing pbx:/etc/asterisk# cat voicemail.conf [internal] 1000 => XXXX,Matt,me(a)myomain.com format=wav49 maxsilence=0 Sorry to be a bonehead, but I'm struggling with the wiki docs (especially with regard to IAX2) and also concerned about security. I installed the samples when I compiled freeswitch but wondering if that is a security risk as the PBX box is on the public internet? Many thanks, Matt

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Re: [Freeswitch-users] Grandstream gateways
by Samuel Mukoti 23 Apr '10

23 Apr '10

Thank you for those tips, I do have some small setups using gxw4108 they work or, except CID doesn't seem to work. I will try the channel bank route - just don't know too much about the setup options or how you'd purchase the correct config, eg. For 150 FXS channel bank, can I get a single PCI card for that? I may end up using the grandstream fxs gateways then use the T1 channel bank from sangoma, Thank you all.. Lastly, I know asterisk now has an offical skype_ module, Is there anything similar I could use? On 25 Nov,2009, at 9:52 PM, Cory Andrews <cory(a)voipsupply.com> wrote: > Samuel - you could go with FXS gateways or channel banks. If you go > the gateway route Grandstream or Audiocodes would work fine. > Audiocodes are a bit more telco grade. If you have 25 POTS incoming > you could use a 24FXO channel bank cross connected with Rhino T1 > cards, or individual FXO gateways but you may have a hard time > finding 24 ports of FXO in a single GW. Best performing T1 cards in > my experience (thousands of deployments) are Sangoma. Your server > configuration looks fine. > > Cory J. Andrews > Director New Market Initiatives > > Sayers Media Group > VoIP Supply, LLC > 454 Sonwil Drive > Buffalo, NY 14225 > 716-250-3402 OFFICE > 716-630-1548 FAX > 716-601-4474 MOBILE > candrews(a)sayersmedia.com > > > Have I exceeded your expectations? Please share your experience > with my boss, Benjamin P. Sayers, CEO > > NOTICE: The information contained in this email and any document > attached hereto is intended only for the named recipient(s). It is > the property of the VoIP Supply, LLC and shall not be used, > disclosed or reproduced without the express written consent of VoIP > Supply, LLC. If you are not the intended recipient, nor the employee > or agent responsible for delivering this message in confidence to > the intended recipient(s), you are hereby notified that you have > received this transmittal in error, and any review, dissemination, > distribution or copying of this transmittal or its attachments is > strictly prohibited. If you have received this transmittal and/or > attachments in error, please notify me immediately by reply e-mail > or telephone and then delete this message, including any > attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY > 14225 USA. > > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of > Samuel Mukoti > Sent: Wednesday, November 25, 2009 2:40 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: [Freeswitch-users] Grandstream gateways > > Hi all, > > I'm wanting to try out a my first large scale setup at the office, 200 > extensions and 24 POTS incoming, also a T1 line once the telco guys > are ready. I wanted assistance with choosing the most appropriate > hardware. We already have about 150 analogue phones, and I was > wondering what's best? A couple of grandstream FXS GXW4024? Also for > my POTS lines, gxw4108 FXO gateway or is it better to buy a sangoma > or digium card? The best voice quality is paramount. Lastly for T1 > what cards are recommeded, > > I was also proposing to use a Dell T116 Quad core intel i7 8G DRAM, > would that perform? Or do I need hardware transcoding? > > Thank you, > > Sam > > Twitter: twitter.com/samuelmukoti > > > On 25 Nov,2009, at 8:05 PM, freeswitch-users-request(a)lists.freeswitch.org > wrote: > >> 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. Re: mod_conference kick to abort invitations (Michael Jerris) >> 2. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Michael Jerris) >> 3. Re: No NOTIFY MWI when registering via proxy. (Brian West) >> 4. Re: remote_media_ip variable not set (Michael Jerris) >> 5. Re: How to find whether the destination extension supports >> encryption (Michael Jerris) >> 6. Re: Bypass_media and re_invite (srinivasula reddy) >> 7. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Stephen Crosby) >> 8. Re: Handling the 302 Moved Temporarily response from >> JavaScript (Tihomir Culjaga) >> >> >> --- >> ------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 25 Nov 2009 12:44:46 -0500 >> From: Michael Jerris <mike(a)jerris.com> >> Subject: Re: [Freeswitch-users] mod_conference kick to abort >> invitations >> To: freeswitch-users(a)lists.freeswitch.org >> Message-ID: <1CCC981C-9F4A-4D97-ACEA-A6DFB906C32B(a)jerris.com> >> Content-Type: text/plain; charset="windows-1252" >> >> Its a feature we don't have, patches welcome. >> >> Mike >> >> On Nov 24, 2009, at 5:35 PM, Jan Thiemo Fricke wrote: >> >>> Hi members, >>> I?m controlling freeswitch with the conference module via xmlrpc. >>> >>> Is it desired that the kick command can only kick users that are >>> connected to the conference? >>> Is there no chance abort an invitation? >>> The kick command has no effect until the person I invited with the >>> dial command is connected. >> >>

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