January 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS hangs up after 30 seconds by timeout on ACK (after sending 200 OK)
by Алексей Дорофеев 20 Jan '13

20 Jan '13

Hi, all! I have two sip profiles: profile_spb and profile_msk for one gateway. I assign sip port 5060 for profile_spb and port 5070 for profile_msk. FS has successfully registered two users with distinct usernames: num_spb and num_msk on the gateway. By the way, field "Contact" in message REGISTER has normal port number - 5060 and 5070 accordingly. When someone calls to num_spb, it shoot to port 5060 and all works well. When someone calls to num_msk, it shoot to port 5070. Call is received, but is rejected after 30 seconds by timeout (not received ACK on 200 OK). I think it's because FS sends 200 OK with port number = 0 in field Contact. All another headers are good (it seems). Does someone know how to say FS to use correct port number in field Contact in 200 OK? Does someone has another version? Thanks a lot!

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Re: [Freeswitch-users] Unloop extension
by Ержан Туле пов 19 Jan '13

19 Jan '13

FreeSWITCH 1.0.6 book on p. 187 says: ... When you set a variable during this phase, that variable is considered global automatically and becomes accessible throughout the application as $${variable} elsewhere in the XML. ... You can utilize global variables in your conditions, your variable and parameter declarations, pretty much anywhere, like this: <condition field="$${global_variable}"> This should had an addition, that ${g_v} can be used as well. ----- Original Message ----- From: Michael Collins [mailto:msc@freeswitch.org] To: FreeSWITCH Users Help [mailto:freeswitch-users@lists.freeswitch.org] Sent: Thu, 17 Jan 2013 07:17:31 +0600 Subject: Re: [Freeswitch-users] Unloop extension > No, it's available globally, whether used with $${} or ${} notation. You > can also see it from fs_cli: > > freeswitch@default> global_getvar unroll_loops > true > > More info about $${} vs. ${} is found > here<http://wiki.freeswitch.org/wiki/Channel_Variables#.24.7Bvariable.7D_vs._.24…> > . > > -MC > > On Tue, Jan 15, 2013 at 8:09 PM, Ержан Тулепов > <e.tylepov(a)btcom.kz> wrote: > > > Ain't what was set with <x-PRE-PROCESS cmd="set" ... gets set *only* when > > used in $${...} notation? > > > > ----- Original Message ----- > > From: Michael Collins [mailto:msc@freeswitch.org] > > To: FreeSWITCH Users Help [mailto:freeswitch-users@lists.freeswitch.org] > > Sent: Wed, 16 Jan 2013 06:54:12 +0600 > > Subject: Re: [Freeswitch-users] Unloop extension > > > > > > > Yes, it's okay. The pre-process does a set and ${unroll_loops} gets set > > as > > > needed. Then this dp entry does the work: > > > <extension name="unloop"> > > > <condition field="${unroll_loops}" expression="^true$"/> > > > <condition field="${sip_looped_call}" expression="^true$"> > > > <action application="deflect" data="${destination_number}"/> > > > </condition> > > > </extension> > > > > > > More info on sip_looped_call var: > > > http://wiki.freeswitch.org/wiki/Variable_sip_looped_call > > > > > > -MC > > > > > > On Mon, Jan 14, 2013 at 9:33 PM, Ержан Тулепов > > > <e.tylepov(a)btcom.kz> wrote: > > > > > > > What unloop extension is used for? BTW: vars.xml pre-(process)-sets > > > > unroll_loops, whereas DPs (public and default) are using channel > > variable > > > > syntax? Is this ok? > > > > > > > > Y. > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting(a)freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > > > http://www.cudatel.com > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.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 > > > > > > > > > > > > > > > > -- > > > Michael S Collins > > > Twitter: @mercutioviz > > > http://www.FreeSWITCH.org > > > http://www.ClueCon.com > > > http://www.OSTAG.org > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting(a)freeswitch.org > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > > http://www.cudatel.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.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 > > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org >

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<macro> levels of <input pattern>?
by Jakob 18 Jan '13

18 Jan '13

Hi, Can you make a construct like below? Match the second argument within a match statement. Or are there a switch statement that can be used? <macro name="voicemail_message_count"> <input pattern="^(1):(.*)$"> <match> <input pattern="(.*):(new)$"> <match> </match> </input> <input pattern="(.*):(saved)$"> <match> </match> </input> <input pattern="(.*):(urgent)$"> <match> </match> </input> </match> <nomatch> </nomatch> </input> </macro> Best Regards Jakob

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SIP Message Parsing and Auto-Action
by Nick Vines 18 Jan '13

18 Jan '13

I have a couple Grandstream HT502s that occasionally get stuck with the detected NAT type of Warning: 399 ip.address "Detected NAT type is UDP Blocked". When that happens the device is never able to complete a call until it is restarted. Where should I start looking to create a monitor that will parse those messages and send a reboot (i.e. sofia profile [profile] flush_inbound_reg [call_id] reboot) Thanks, Nick *Sample SIP Message* SIP/2.0 200 OK Via: SIP/2.0/TCP XXX.XXX.XXX.XXX:PORT;branch=z9hG4bK2N9vy7BmyF40B From: <sip:22521002@DOMAIN.COM>;tag=3yNZa6ZNv6K6g To: <sip:22521002@DOMAIN.COM>;tag=1662499825 Call-ID: 3901be5e-db58-1230-109f-bc764e103a41 CSeq: 38881081 NOTIFY Supported: replaces, path, timer, eventlist User-Agent: Grandstream HT-502 V1.2A 1.0.8.4 chip V2.2 Warning: 399 XXX.XXX.XXX.XXX *"Detected NAT type is UDP Blocked"* Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE Content-Length: 0

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trouble with record_session
by Ashok M 18 Jan '13

18 Jan '13

I am trying to test recording feature using record_session app. Here is my dial plan for the internal extension that I am trying dial from another extension which gets registered with FreeSWITCH after starting x-lite. The record_session seems to creating the wav file, but if I try to play it back and I don't hear anything. am I doing this right? or am I missing something here? <extension name="test_recording"> <condition field="destination_number" expression="^1300$"> <action application="set" data="continue_on_fail=true"/> <action application="answer"/> <action application="record_session" data="d:/record/Site/recordtest.wav"/> <action application="sleep" data="20000"/> <action application="playback" data="d:/record/Site/recordtest.wav"/> <action application="hangup"/> </condition> </extension> Thanks, -Ashok

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SONUS - dtg inside INVITE
by Andrea Mazzeo 18 Jan '13

18 Jan '13

Hi all, I need to configure my FS to send call to a Sonus GSX via SIP. Is there a way to send SIP parameter "dtg" (destination trunk group) inside the INVITE? Example: Request-Line: INVITE sip: 39340xxxxxx(a)38.xxx.xxx.xxx SIP/2.0 dtg=ATTMDT7807IPO Thanks, Andrea

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Re: [Freeswitch-users] segfault in libfreeswitch.so.1.0.0
by Peter Olsson 18 Jan '13

18 Jan '13

First of all - try latest from git head. If it still exists, report to Jira. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Mimiko Skickat: den 18 januari 2013 10:23 Till: FreeSWITCH Users Help Ämne: [Freeswitch-users] segfault in libfreeswitch.so.1.0.0 New one: kernel: freeswitch[3559]: segfault at 30 ip 00007fbb945862a4 sp 00007fbb8ef59bc0 error 4 in libfreeswitch.so.1.0.0[7fbb944fb000+277000] This happened when channels where destroyed after a bridge and a hangup. -- Mimiko desu. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 !DSPAM:50f911b532761664217464!

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Re: [Freeswitch-users] limit memory usage of freeswitch
by Raimund Sacherer 18 Jan '13

18 Jan '13

Hello Anthony, I also see Freeswitch eating away on Memory. Right now we are in the process of installing a new FS Cluster and I will upgrade all our PBX's with the new stable branch and I will test with that again. Our version of Freeswitch in production is: FreeSWITCH Version 1.0.head (git-4ceca18 2011-10-21 20-38-41 -0500) This version slowly grows and grows, Over month the biggest grow I saw was about 4-5 Gigabyte of RES memory. Before that we used a Freeswitch version from 2011-03 and that version grow to 273 Megabyte of memory usage and did not grow further. In the future I will report my findings with the new stable branch, hopefully whatever caused the memory consumption is allready fixed. Best regards, Ray ----- Original Message ----- From: "Anthony Minessale" <anthony.minessale(a)gmail.com> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Sent: Jueves, 17 de Enero 2013 2:12:58 Subject: Re: [Freeswitch-users] limit memory usage of freeswitch It does not eat memory.... if you attack it with sipp you may cause the internal pools to swell up but normal usage only consumes memory based on usage. On Wed, Jan 16, 2013 at 4:09 AM, jay binks < jaybinks(a)gmail.com > wrote: OpenVZ / Proxmox will allow you use set quota's and containerize your freeswitch... thats ONE way to do what your after. On 16 January 2013 19:48, rentmycoder rentmycoder < rentmycoder(a)gmail.com > wrote: <blockquote> Hi All, According to my experineces and some info found on FS lists it is the normal behaviour that freeswitch eats up all of the available memory... It's not a memory leak, but the standard behaviour of FS by design... Correct me if I'm wrong... Is there any way to limit FS memory usage??? I need another services to run on the FS box... Thanks, John _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 -- Sincerely Jay _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 </blockquote> -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/ PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server http://www.cudatel.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.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 --

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Linux PREEMPT RT kernel and freeswitch -rp
by Dmitry Lysenko 17 Jan '13

17 Jan '13

Hello, freeswitchers. Does anybody here use freeswitch -rp with PREEMPT RT Linux kernel? WBR.

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Mod_Conference Improvement (Denoise)
by Usama Zaidi 17 Jan '13

17 Jan '13

Hi, I talked about this a while back on the channel, the way how energy-level is implemented in mod_conference isn't the right way to handle noise in my opinion because the participants that get bridged would have a different noise floor depending on where they are located, so if I set the value of energy-level too high to accommodate one participant, any other participant in a quite sitting would never get bridged, I proposed we should tackle this the way Asterisk does using a denoise dialplan function ( https://wiki.asterisk.org/wiki/display/AST/Asterisk+10+Function_DENOISE) we can use libav for noise filtering on the muxed audio. I'm willing to contribute if one of the devs would help me out regarding implementation. Any comments? -Regards -- I'd love to change the world, but they wont gimme the source code to it

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