- FreeSWITCH-users - FreeSWITCH Mailing Lists

Employment Opportunities
by Anthony Minessale 29 Oct '09

29 Oct '09

We're looking for a few individuals to work with us on the CudaTel Product powered by FreeSWITCH. AJAX Developer: Must have strong skill in HTTP/CGI/Javascript preferably some exp in jquery. We would prefer if are not overly obsessed with any standards but respect them enough to try to follow them in a well-balanced way. Best if you live near Silicon Valley, Ann Arbor Michigan, or Milwaukee WI but not mandatory. Must be self-motivated. OCD a plus. Willing to work late if necessary to get things done right. Working knowledge of Perl, DBI-X Catalyst and TT2 would make your life easier. Knowing a little about FS and Telephony would be nice. Must be able to work diligently from a ssh terminal. Back-end Developer: Good knowledge of 2 or more Perl, DBI-X Catalyst and TT2. Attention to detail, good debugging skills. Best if you live near Silicon Valley, Ann Arbor Michigan, or Milwaukee WI but not mandatory. Must be self-motivated. OCD a plus. Willing to work late if necessary to get things done right. Ability to work well with the guy in the other job description above. Knowing a little about FS and Telephony would be nice. Must be able to work diligently from a ssh terminal. Both positions involve a hands-on "extreme programming" type environment with shared screens and a permanent line of communication to your peers. Send CR to jobs(a)freeswitch.org -- 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 <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:213-799-1400

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Determining Frame Size
by Matthew Fong 29 Oct '09

29 Oct '09

I noticed that the wait_for_silence's <silence_hits> <listen_hits> use number of frames as base values (rather than seconds or milliseconds). I'd like to wait for 5 seconds of consistent silence, but I'm wondering how I can determine the sample rate (and thus frames per second) for an existing call. Is there a variable I can read? Thanks. --matt

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Re: [Freeswitch-users] FreeSWITCH on FreeBSD
by Mikhail T. 29 Oct '09

29 Oct '09

Richard E Neese написав(ла): >> * What hardware is known to work well on FreeBSD? >> * What else -- other than a computer with a phone card -- will I >> need? >> * Do end-user phones have to be regular analog phones (all >> plugged-in to the computer), or IP-phones, such as made by Avaya >> (connecting via an Ethernet switch)? >> * How is the computer connecting to the world? Do I plug-in the >> incoming analog phone line(s) somewhere, or is it going to go >> through the Internet? >> * If the connection to the world is over the Internet, don't we >> need >> an account with someone else, or is Verizon known to work with >> FreeSWITCH-based PBX-systems directly? >> * Should I not simply buy a used Avaya-system like this: >> >> http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=110450339950&_trksid=p27… >> >> What will I gain and what will I lose (other than personal >> satisfaction, of course) by going that route instead of building >> my own box? Thank you very much, Richard, for you answering some of my questions. A few remain, however: > using freeswitch on bsd is free. nd all you kneed is phones like > polycom soundpoint or aastra or snome or cisco . Sip phones. So, the phones connect -- via CAT6 cables -- to my network switch, and talk to my FreeBSD server running FreeSWITCH. Ok. But how does the server connect to the world? Somewhere between me talking into a Polycom handset in New Jersey and my friend in hearing me in Ukraine, the audio has to hit the telephone network... Where does this happen? Do I plug-in a regular phone line into the same server? If so, what hardware do I need inside it? Or do I use a VOIP-provider -- connecting my server to them via the Internet? And if so, which providers should I consider -- would Verizon themselves qualify? I ask, because the phone-service provided with Verizon's FIOS is already a VOIP of some kind -- there is no copper line any more, it is all done through optic cable... Verizon-provided device connects to that fiber on one end, and gives you a phone-line on the other... So it would be rather silly for my SIP-phones to convert the analog speech into packets, then for the FreeSWITCH server to convert them back into analog audio, and then for Verizon's equipment right in my house to digitize back... Thanks! Yours, -mi

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pass arguments into javascript
by Erwin Davis 29 Oct '09

29 Oct '09

Hi, new to javascript. I tried to pass two arguments into javascript, <action application="javascript" data="test.js $1 $2"/> In test.js, I tried to use argv[1] to retrieve $1 and argv[2] to retrieve $2, however, the javascript test.js complained about argv[] as undefined variables. How to retrieve the passing arguments in a javascript same as the case above. Thanks,

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error in loading spidermonkey
by Erwin Davis 29 Oct '09

29 Oct '09

Hi, I got an error in loading mod_spidermonkey. my fs in ver 1.04 runs on fedora 8 VM. Any clue? thanks, [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey.so **/usr/lib/libnss3.so: undefined symbol: PR_UnloadLibrary**

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mod_opal - call charged before H.225 connect
by Tihomir Culjaga 29 Oct '09

29 Oct '09

hello guys, i was playing with mod_opal to see if i can make it working ... well it seems SIP-H323 interworking is not tuned at all. I have a call from a registered sip user (1001) to PSTN via mod_opal <include> <extension name="EMERGENCY"> <condition field="destination_number" expression="^0(112|9[23456])$"> <action application="set" data="effective_caller_id_number=1001282122"/> <action application="set" data="NCX_IP=10.4.4.254"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/> </condition> </extension> <extension name="SPECIAL_SERVICES"> <condition field="destination_number" expression="^0(9[01789]\d{3,4})$"> <action application="set" data="effective_caller_id_number=1001282122"/> <action application="set" data="NCX_IP=10.4.4.254"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/> </condition> </extension> <extension name="ENYTHING_ELSE"> <condition field="destination_number" expression="^0([2-8]\d{6,7}|0[1-9]\d{7,8}|00[1-9]\d{8,16})$"> <action application="set" data="effective_caller_id_number=1001282122"/> <action application="set" data="NCX_IP=10.4.4.254"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="opal/h323:0$1@${NCX_IP}"/> </condition> </extension> </include> One of the many issues i sow is that FS connects the call on SIP leg before it actually receives H.225 connect from H323 leg... as it is configured to send 200 OK on the 1st H.225 message containing a FastStart element/OLC. Attached is the tcpdump i took on FS machine... just use this filter: "h225 or h245 or q931 or sip" Also, you can check the attac CDR,,,, this is an unanswered call i placed to PSTN and FS billed it 23 seconds. Can anyone tell where i can do correct SIP - H323 message mappings to avoid this? T.

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Multi-tenancy context manipulation
by Martin Hickman 28 Oct '09

28 Oct '09

Hello, it's my first post, and I'm quite new to Freeswitch... I'm looking for a bit of direction / advice on Multi-tenant set-ups. I have configured Freeswitch as per the fine Multi-tenant guide on the Wiki, but this leaves some aspects such as transfers and IVR still in the default context. For example, transfering a call that is passed to features.xml, or connecting to an extension thorough demo_ivr will fail if the call is left in the default context. Would you recommend a separate features.xml for each tenants context? If the features provided by features.xml and the ivr are the same for all tenants, could you advise on the best way to implement one features.xml for all tenants? Kind regards, Marty Hickman

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Re: [Freeswitch-users] Estimating Call Capacity
by Eliot Gable 28 Oct '09

28 Oct '09

Although, FYI, I just benchmarked mod_xml_curl on a separate web app server from FS with FS on a Dell R710 with their current best processor option (Intel Xeon X5570 @2.93GHz with 8-cores total) and 32 GB memory. The web app server is less than half the power of the R710. I maxed the web app server at 300 calls per second (both setting up and tearing down) and the R710 running FS was 65% idle. No audio was being proxied through FS, though. If I were running the web app server on an equivalent R710, they probably would have been on-par with each other in performance. Extrapolating, I expect that in such a case I should be able to get at least 650 CPS out of FS, though for production I would probably limit it to 400 CPS or less so I leave room for miscellaneous tasks. I maxed out the R710 at over 16,000 simultaneous calls (again, no audio proxying) but the only reason I couldn't do more was because I hit some sort of thread creation limit in Linux. There was about 17 GB of memory used for this many calls. This should give you some ballpark idea of what you can accomplish with FS. At some point, I will track down and resolve the thread creation issue, at which time I believe call limits will be limited either by a complex combination of available memory, the speed of the processor, the cost of thread context switching, calls per second setup rate, and call duration. -- Eliot Gable > -----Original Message----- > > From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli > > Sent: Monday, October 26, 2009 4:56 PM > > To: freeswitch-users(a)lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Estimating Call Capacity > > > > On Mon, Oct 26, 2009 at 9:28 PM, Vinuth Madinur > > <vinuth.madinur(a)gmail.com> wrote: > > > Here are a few benchmarks that I had stumbled upon. > > > http://wiki.voiceworks.pl/display/~pawel/FreeSwitch+performance+on+SUN+x220… > > > > Please remember NO benchmarks are endorsed by the FS community or > > developers, because there are just too many variables, and a simple > > figure is just useful for marketing hype, not for real dimensioning. > > > > You MUST do your own benchmarking, so you get an idea about how to > > dimension for your own use case and hardware. > > > > > > > Thanks, > > > Vinuth. > > > > > > On Tue, Oct 27, 2009 at 1:43 AM, Brian West <brian(a)freeswitch.org> wrote: > > >> > > >> I highly doubt it... You can wait for someone to post their results > > >> but in the end you'll have to do your own load testing because not > > >> everyone's numbers will jive with your use case. Which is the reason > > >> the project never posts or endorses a set call count. > > >> > > >> /b > > >> > > >> On Oct 26, 2009, at 2:50 PM, Ujjval Karihaloo wrote: > > >> > > >> > Are there any benchmarking test results available publicly? > > >> > ________________________________________ > > >> > > >> > > >> _______________________________________________ > > >> 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 > > > > > > > > > _______________________________________________ > > > 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, > > > > Giovanni Maruzzelli > > Cell : +39-347-2665618 > > > > _______________________________________________ > > 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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IVR Intro Clipped
by Jerry Richards 28 Oct '09

28 Oct '09

I notice that when I call IVR from the PSTN, the "Welcome to Freeswitch..." introduction is clipped at the beginning, so it sounds like "come to Freeswitch". If I call 5000 internally, then I always hear the full introduction. What can I do to resolve this? My XML config looks like: <extension name="ivr_demo"> <condition field="destination_number" expression="5000"> <action application="answer"/> <action application="start_dtmf"/> <action application="ivr" data="demo_ivr"/> </condition> </extension> Best Regards, Jerry

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meaning of created_time channel variable.
by velusamy velu 28 Oct '09

28 Oct '09

Dear All, What is the value of created_time channel variable? Is this epoch seconds? Thanks & Regards, Velusamy.

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