May 2012 - FreeSWITCH-users - FreeSWITCH Mailing Lists

FS and Opensips
by azza miled 11 May '12

11 May '12

Hi! I know that is a stupid question but can FS do automatic call ditribution for Opensips agent??? Thanks a lot

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Re: [Freeswitch-users] Freeswitch hardware cofig
by tarik.bts.gi＠gmail.com 11 May '12

11 May '12

I have atom procressor, i want to have 30 simeltanous calls, and i dont have transcoding we use the same codec ------Original message------ From: Michael Collins <msc(a)freeswitch.org> To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Date: Thursday, May 10, 2012 3:18:13 PM GMT-0700 Subject: Re: [Freeswitch-users] Freeswitch hardware cofig On Thu, May 10, 2012 at 12:43 PM, <tarik.bts.gi(a)gmail.com> wrote: > What is the good hardware configuration > that needs freeswitch to support more then 30 calls? We have now 1.6 GHz > in the CPU and 2 GB in the mem but is not enough. > What kind of processor? What operating system? Do you want 30 simultaneous calls or 30 new calls per second? Will you be doing transcoding on these calls? All those details will help you decide what to do next. -MC _________________________________________________________________________ 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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Where can i find ESL.php?
by Szilárd Kovács 11 May '12

11 May '12

Hello I would like to try the event socket connection with php, but i don't know where can i download ESL.php. Anybody? Thanks, Szilard Kovacs

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Choopy one-way noise
by Christian Benke 11 May '12

11 May '12

Hello! On a box with a Sangoma A500 BRI card i have a serious audio quality problem. The "outside"-leg of a call hears a choppy robotic noise, it's difficult to understand the actual voice. I hope someone recognizes this issue, as i have tried several approaches and so far i neither know what causes the problem nor where it is caused exactly. Here's a sample file(The noise is not very noticeable, it's the oscillating hum you hear in the background, the short voice-sample you hear is mine, but it's inbound so it's not affected): http://poab.org/test1.wav If the call is set on-hold or if there's a playback/moh, the quality for the outside-leg is fine while there's no two-way-audio, so my assumption was that it's some kind of problem with codecs or transcoding. Capturing the RTP-Stream from the phone to the server proved though that the quality is fine before it arrives at FS(There are no audio problems on internal calls as well). Here's what else i've tried without success: - set disable-monotonic-timing to true - started FS with the -rt Flag(As well as -waste and -heavy-timer) - jitter "time_test" 100 100 responds with avg 130(Thats fine right?) - rearranged the codec-settings of the phone(Sangoma SPIP 450) to give A-law the highest priority - high-precision-timer is available in the kernel according to dmesg and /proc/timer_list (kernel is 2.6.32-5-686 on debian 6.0) Sangoma's Support appears to be clueless as well, they didn't have effective suggestions so far(Opened a ticket 2 weeks). Since i don't have this problem with identical hardware without Sangoma-Cards(SIP-Trunk instead), my first assumption was the card. Probably an interrupt issue? Any suggestions would be highly appreciated, i'm a bit at loss now and only know more drastic next steps(Replacing ISDN-Modules or the whole hardware). Best regards, Christian

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macports installed - does this create compilation problem
by Chris Mylonas 10 May '12

10 May '12

Hi List, In the distant past I had FS running on my laptop (OSX 10.6.x) It hasn't compiled for months. I just found this JIRA http://jira.freeswitch.org/browse/FS-3473 There is a mention that the build environment is stuffed possibly because of macports. My build stuffs up on gsm..... and spandsp I have commented out, sic, applications/spandsp in modules.conf and now this is the error making all mod_spidermonkey cd config; make -j1 export cd pr; make -j1 export cd include; make export cd md; make export cd private; make export cd obsolete; make export cd src; make export cd io; make export cd malloc; make export cd md; make export cd unix; make export os_Darwin_ppc.s:49:no such instruction: `lwarx r4,0,r3' os_Darwin_ppc.s:50:no such instruction: `addi r0,r4,1' os_Darwin_ppc.s:51:invalid character '.' in mnemonic os_Darwin_ppc.s:52:invalid character '-' in mnemonic os_Darwin_ppc.s:53:no such instruction: `mr r3,r0' os_Darwin_ppc.s:54:no such instruction: `blr' os_Darwin_ppc.s:62:no such instruction: `lwarx r4,0,r3' os_Darwin_ppc.s:63:no such instruction: `addi r0,r4,-1' os_Darwin_ppc.s:64:invalid character '.' in mnemonic os_Darwin_ppc.s:65:invalid character '-' in mnemonic os_Darwin_ppc.s:66:no such instruction: `mr r3,r0' os_Darwin_ppc.s:67:no such instruction: `blr' os_Darwin_ppc.s:75:no such instruction: `lwarx r5,0,r3' os_Darwin_ppc.s:76:invalid character '.' in mnemonic os_Darwin_ppc.s:77:invalid character '-' in mnemonic os_Darwin_ppc.s:78:no such instruction: `mr r3,r5' os_Darwin_ppc.s:79:no such instruction: `blr' os_Darwin_ppc.s:87:no such instruction: `lwarx r5,0,r3' os_Darwin_ppc.s:88:too many memory references for `add' os_Darwin_ppc.s:89:invalid character '.' in mnemonic os_Darwin_ppc.s:90:invalid character '-' in mnemonic os_Darwin_ppc.s:91:no such instruction: `mr r3,r0' os_Darwin_ppc.s:92:no such instruction: `blr' make[10]: *** [os_Darwin_ppc.o] Error 1 make[9]: *** [export] Error 2 make[8]: *** [export] Error 2 make[7]: *** [export] Error 2 make[6]: *** [export] Error 2 make[5]: *** [/Users/chrismylonas/freeswitch-git/freeswitch/libs/js/libjs.la] Error 2 make[4]: *** [mod_spidermonkey-all] Error 1 make[3]: *** [all-recursive] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all] Error 2 make: *** [current] Error 2 Can this be confirmed to be a macports problem? Thanks

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Can't compile with ODBC support under FreeBSD for current git
by Nikolay Rogoshchenkov 10 May '12

10 May '12

Hello, I already have FreeSWITCH Version 1.1.beta1 (git-d730df7 2012-04-02 18-00-38 -0400) with ODBC support and my LUA scripts working properly with MySQL. But when I tried to *configure *the latest git (after git pull) I have got the following error: ================================================================================ [....] *** your path, or set the LIBGNUTLS_CONFIG environment variable to the *** full path to libgnutls-config. checking libtool major version... 2 using libtool library extension... la adding "-fPIC" to SWITCH_AM_CFLAGS adding "-fPIC" to SWITCH_AM_CXXFLAGS adding "-Werror" to SWITCH_AM_CFLAGS checking whether the compiler supports -fvisibility=hidden... yes adding "-fvisibility=hidden" to SWITCH_AM_CFLAGS adding "-DSWITCH_API_VISIBILITY=1" to SWITCH_AM_CFLAGS adding "-DHAVE_VISIBILITY=1" to SWITCH_AM_CFLAGS adding "-fvisibility=hidden" to SWITCH_AM_CXXFLAGS adding "-DSWITCH_API_VISIBILITY=1" to SWITCH_AM_CXXFLAGS adding "-DHAVE_VISIBILITY=1" to SWITCH_AM_CXXFLAGS checking CFLAGS for maximum ansi warnings... -Wall -std=c99 -pedantic adding "-g" to SWITCH_AM_CFLAGS adding "-ggdb" to SWITCH_AM_CFLAGS checking for jack... gnome-config: not found gnome-config: not found checking for snd_pcm_open in -lasound... no checking size of long... 8 checking what directory libraries are found in... lib checking for odbc header in /usr/include... no found checking for odbc header in /usr/local/include... found checking for SQLDisconnect in -lodbc... no checking for odbc library in /usr/lib... no found checking for odbc library in /usr/local/lib... found checking whether to include odbc... yes checking for SQLDisconnect in -lodbc... (cached) no configure: error: no usable libodbc; please install unixodbc devel package or equivalent # ================================================================================ but why? Thank you. -- Nikolay

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Re: [Freeswitch-users] 1.2-rc2 Tagged and Tarball available on files.freeswitch.org
by Tom Parrott 10 May '12

10 May '12

Excellent, will try and build that tonight. The RPM SPEC file has seen a lot of improvements over the last few months, it's a lot more modular now. Cheers Tom > Congratulations!! I know it's been a long time coming - I've been lurking > around the FS site/irc for months! (my FS plans keep getting interfered > with though!) > > :) > > > On 09/05/2012, at 6:06 AM, Ken Rice wrote: > >> Hey Guys weve reached another milestone in getting FreeSWITCH 1.2 out >> the door. >> >> http://files.freeswitch.org/freeswitch-1.2.rc2.tar.bz2 >> >> Yep thats what you want to test... >> >> Look for debian packages and updated RPMs for these soon >> >> K >> _________________________________________________________________________ >> 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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Howto? Migrating from Siemens HiPath4000 to FreeSWITCH
by NeCoSec - Ulrich Schinz 10 May '12

10 May '12

Hi FS-List, maybe there are some experts in here, that can help me to get some things clear. I'm working for an institution with about 500 extensions. At the moment there is a Siemens HiPath4000 used. It is an old version of the Software, we are using (3.x). We are paying a lot of money for that solution(rental fee/leasing), so I'm planning to move to another option. The building is an old monestary and the infrastructure (e.g. cables) is very old. So we are planning to get the Infrastructure up to cat7 cables. On the one hand, because we need to get most parts of the building online (internet for guests and conferences) and on the other hand it makes no sense to keep a 70-80 years old cabeling up. So with this in mind VoIP would be a good option.... To bring the cabeling up to date, we're planning to do it part by part. It's not a realistic option to get this done in a short time. The building is a very large one, so this has to be done in maybe 3-5 years... We are using a Telekom PRI/E1 (german Telekom). So my plan is: Setting up a FreeSWITCH server with 2 PRI cards (e.g. Sangoma A101D), one card connecting to the PRI of Telekom and the other card connecting to the HiPath4000. In this way I hope to get the calls connected through FreeSWITCH to the HiPath4000 and I can start getting ip-phones connected, where cat7-cables are available. In this way I have some advantages: - I can handle a smooth change to VoIP (don't have to have a hard cut) - I first can integrate open-minded people to the VoIP-technologie, so they can test and give feedback, what they like, and what they wanna have better... Older or not so open-minded people (we have some really old monks and padres) would then benfit from a tested environment. The switches are planed to have QoS and PoE. As phones I'm planning to use Snom-phones. (Snom300 and Snom320) Further I'd like to build a failover system, with a second server. Another option would have been to upgrade the HiPath to VoIP, but the costs for that are in no relation to what a FreeSWITCH System would cost... My experience with FreeSWITCH is limited to quite small enviroments and I never tried it with PRI (always using "normal" ISDN base, Sangoma a500...) My experience with HiPath4000 is limited to the available webfrontend. The "internal configuration" is done by the provider of that system. My experience with Linux and networksystems is about 12 years in small an large systems... In two years the HiPath is "payed" and it's ours. So we can use this system with no costs and have full access. My questions: - Do you think this is an good way to get it running? - Maybe someone with some experience in HiPath knows, whether it's complicated, to get the HiPath configured (getting extension-blocks handled by FreeSWITCH and on the other side keep other extensions on the HiPath)? - Is the routing with this 2 PRI-cards construct possible, or better easy done? - Is there a way to simulate a PRI-Provider. I.e. if I'm buying two PRI-cards I maybe could configure two test-boxes, each having one card. Connecting these two I maybe can play around with these PRI connections (just to get a feeling for it...)? - Do you have some hints, what to consider, if I'm doing this? I'm planning this for quite a while, and im thinking a lot about this, but I always have the feeling of forgetting some essential things and the whole construct blows up, if im starting it. This would be not very optimal ;) - Maybe someone has some hints, in which order to plan this setup. The plans become more and more concrete and I'm not sure, like said before, whether I'm doing and thinking things in right order. - Just a technical detail: the HiPath is in a room, that is not optimal for this kind of it-systems (aircondition, accessible for everybody). But the patchpanels are in that room (the old ones, old cables, 2 poles for HiPath-Protocol, parts of it with 4 poles for ISDN). And the PRI-connection to the Telekom is in this room. Would it be possible to place the servers in another room and get the PRI there? Would the cable-length be a problem (I think about 50 meters)? And for the failover system it would be optimal to have the system in another part of the building (in case of fire another place is better). There it would be nice to "patch" the PRI to another place... - As the building is very large, we have some fibrechanel connections to get the long distances bridged. To keep QoS up: Do I have to have special converters? (I don't think so, cause the switches should do that, but maybe I'm wrong). - Cascading switches (Phone -> Switch -> Fibre -> Switch -> Server): Does every switch have to do QoS, or would the first switch be enough. Or the other way around, would cascaded switches make problems, if both have QoS configuration...? Sorry for this very long post and so many questions.... Kind regards Uli

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Bypass media succeeds from extension to gateway but fails from extension to extension
by Phil Quesinberry 10 May '12

10 May '12

Thanks Ken. I'm still learning here, so if that's the case then help me to understand why FS properly sends the external address when that same extension makes a call out through a gateway instead of to another extension. Same settings, same originating extension. I did try enabling ndlb-connectile-dysfunction to rewrite the contact IP on those calls but to no avail. Thanks! - Phil ---------- Ken Rice Tue May 8 11:38:48 MSD 2012 This is not a bug in FreeSWITCH... Bypass Media is a special mode to allow us to act as a psuedo proxy. Other then possibly filtering some codecs the rest of the SDPs are simply copied across the bridge. That means if your sip devices are sending RFC1918 IPs in their SDPs then FreeSWITCH will forward an RFC1918 SDP. Why is that? Because FreeSWITCH has no way to know where the media is actually coming from since FreeSWITCH is not in the media path. K _____________________________________________ From: Phil Quesinberry Sent: Monday, May 07, 2012 11:54 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: Bypass media succeeds from extension to gateway but fails from extension to extension I'm not sure if this is a bug or just a NAT-related configurational problem. If it's truly a bug, let me know and I'll be happy to file a Jira. I'm trying to get bypass media to work with extension to extension calls. Both endpoint extensions are behind NAT in two different locations. This looks like a possible bug because FS sends the internal IP address of one of the endpoints to the other for media, but when making a call from the same extension to an external gateway for PSTN termination, it sends the phone's external IP address as it should, and the call succeeds. Everything works fine, of course, when "proxy media" is used. The SIP traffic for the failed call is here, look for the word "WRONG!" to see the incorrect address being passed in the SDP: http://pastebin.freeswitch.org/19003 Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com

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productivity of FS
by Yuriy Nasida 10 May '12

10 May '12

Hello guys, I have FS box with 200-250 concurrent calls. FS git 05/03/2012. Intel(R) Xeon(R) CPU E5620 @ 2.40GHzMemTotal: 12187540 kB I use xml_mod_curl + php for dynamic directory\dialplan. Also I use lua scripts + mysql for some cases. I use sip options ping for some checking of productivity.Sometimes (when loading is high) I have big delay with sip ping (>2 sec). The box which sends sip ping in same network with main box. My question:1) Is it many concurrent calls for setup I have ?2) If not, what the best way for understanding where I lose productivity ? Please advise.Thanks.

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