May 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Memory leak in mod_nibblebill or in ODBC core?
by Sergey Okhapkin 12 May '10

12 May '10

I want to utilize nibblebill features for prepaid support, but as soon as I enable nibble billing with set nibble_rate=XXX set nibble_account=NNNNNN set enable_heartbeat_events=60 in dialplan, FS process begins to eat more and more memory, RSS grows from 23M to 100M and more after processing few thousands calls and continues to grow. If I comment out those 3 lines from dialplan, then FS RSS grows from initial 23M to 50-60M (depending on the number of concurrent calls) and stays at this value, no memory leaks. Do anybody use mod_nibblebill? I'm experiencing this problem with all svn versions I tried including latest git version.

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ClueCon MMX in 3 months Register Early / Become a Sponsor
by Anthony Minessale 12 May '10

12 May '10

This is the time of year we like to push for those interested in ClueCon to register. Knowing the registration count makes it easier to plan and guarantees that everyone gets a room. Also, we have a handful of speaker slots still available if anyone wants to speak, contact marketing(a)cluecon.com Finally, if your company is benefiting from open source software such as FreeSWITCH or you have some new telephony products and services and you want to have your logo engraved into our prizes and proudly displayed on a banner at the conference, contact marketing(a)cluecon.com as well to become a ClueCon Sponsor. -- 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> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:+19193869900

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Questions on Building an application for FreeSWITCH
by Phone 12 May '10

12 May '10

I am looking for a boot in the right direction on the following project. I am looking to build an "application" with FS where it will frequently check a database and then make an outbound call when a record with certain conditions is found in the DB. It needs to determine if the call was then answered by a machine/vm, busy, or answered by a person, play a message and get a response back to be recorded in the db. It may find many calls that need to be made at the same time. It will be running on a PRI. I am confused as to what approach to use to accomplish this. Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc. Liverpie, ESL....etc. Or a totally different approach. I realize that there is not only one "correct" way to build this, but I am looking for advise from those who have done this type of thing before. I value the opinion of those who have gone down this road and would be willing to share their thoughts on a recommended path to take to accomplish this. Thanks

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echo video calls
by Mikhail Krivushin 12 May '10

12 May '10

Hello! I'm trying to build echo service with video support, but fail. I add video codecs to `global_codec_prefs` and `outbound_codec_prefs`. Add extension: <extension name="echo123"> <condition field="destination_number" expression="^echo123$"> <action application="answer"/> <action application="echo"/> </condition> </extension> And try to make video call. My phone send SDP: v=0 o=deepwalker 123456 654321 IN IP4 s=A conversation c=IN IP4 t=0 0 m=audio 7078 RTP/AVP 111 110 112 3 0 8 101 b=AS:80 a=rtpmap:111 speex/16000/1 a=rtpmap:110 speex/8000/1 a=rtpmap:112 speex/32000/1 a=rtpmap:3 GSM/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 m=video 9078 RTP/AVP 102 103 34 98 99 97 b=AS:410 a=rtpmap:102 H264/90000 a=rtpmap:103 H264/90000 a=fmtp:103 packetization-mode=1 a=rtpmap:34 H263/90000 a=rtpmap:98 H263-1998/90000 a=fmtp:98 CIF=1;QCIF=1 a=rtpmap:99 MP4V-ES/90000 a=fmtp:99 profile-level-id=3 a=rtpmap:97 theora/90000 But FreeSWITCH send back: v=0 o=FreeSWITCH 1273664404 1273664405 IN IP4 s=FreeSWITCH c=IN IP4 t=0 0 m=audio 20146 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 m=video 0 RTP/AVP 19 Can anyone suggest me, what I need to do? -- С уважением, Кривушин Михаил г. Томск сот. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME(a)jabber.ru skype: mkrivushin

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build error on latest git
by babak yakhchali 12 May '10

12 May '10

hi I got this error on latest git pulled today: Error 1 error LNK2019: unresolved external symbol _saturate16 referenced in function _sig_tone_rx@12 sig_tone.obj Error 2 fatal error LNK1120: 1 unresolved externals D:\freeswitch\Debug\libspandsp.dll thanks

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mod_sofia help
by Jor3l Boa 12 May '10

12 May '10

Hello there, I need some help with Sofia and FreeSwitch (new here :)) when the client attempts to connect I'm getting this: 2010-05-12 13:51:16.888628 [WARNING] sofia_reg.c:1031 SIP auth failure (REGISTER) on sofia profile 'internal' for [xbd6Y0PU7RI2mIOBsPW83Gw==@ 192.168.0.171] from ip 192.168.0.158 2010-05-12 13:51:39.707814 [WARNING] sofia_reg.c:1879 Can't find user [xbd6Y0PU7RI2mIOBsPW83Gw==(a)192.168.0.171] You must define a domain called '192.168.0.171' in your directory and add a user with the id="xbd6Y0PU7RI2mIOBsPW83Gw==" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2010-05-12 13:51:39.710879 [WARNING] sofia_reg.c:1031 SIP auth failure (REGISTER) on sofia profile 'internal' for [xbd6Y0PU7RI2mIOBsPW83Gw==@ 192.168.0.171] from ip 192.168.0.158 I guess there is a command to allow any user using this port but I'm not sure.. tried google with no luck. So, is there a way to allow any user to use the service or be registered automatically? since each user name, is different and I can't add each one manually (it's a lot of info), or, is possible to shutdown Sofia and use the service without it? Thanks :-) Jorel

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FreeSWITCH Conference Call Starting Shortly!
by Michael Collins 12 May '10

12 May '10

C'mon down! http://wiki.freeswitch.org/wiki/FS_weekly_2010_05_12 DRK is discussing mod_managed and we have a few other things to talk about. Please bring your questions and your handy tips & tricks! -MC

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files.freeswitch.org is down !
by Shrouk Khan 12 May '10

12 May '10

hi , was trying to compile and install freeswitch 1.0.6 and it keeps getting stuck at make stage: making install mod_curl --2010-05-12 23:13:27-- http://files.freeswitch.org/downloads/libs/json-c-0.9.tar.gz Resolving files.freeswitch.org... 69.174.57.101 Connecting to files.freeswitch.org|69.174.57.101|:80... failed: Connection refused. cannot find json-c-0.9.tar.gz make[5]: *** [/usr/src/freeswitch-1.0.6/libs/json-c-0.9] Error 1 make[4]: *** [install] Error 1 make[3]: *** [mod_curl-install] Error 1 make[2]: *** [install-recursive] Error 1 i tried browsing to that site using the browser and that didnt work either. is the server down ? when will it be up ? -- Regards Shrouk Khan (Khan) System Administrator / Telecommunication System Developer Office: +354 4400807 (Reykjavik) +44 2031370800 (London) Mobile: +66 875049439 (Bangkok) Web: www.softverk.is Reykjavik, Iceland // London, UK // Bangkok, Thailand

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SIPfoundry ACD w/FreeSWITCH
by Jan Berger 12 May '10

12 May '10

Hi, Maybe old news, but found this design page with reference to FreeSWITCH on SIPfoundry. http://sipx-wiki.calivia.com/index.php/ACD_Proposed_Architecture They seem to be increasingly found of FreeSWITCH. Jan

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Freeswitch Scripting strangeness
by paul bilke 12 May '10

12 May '10

I am working on a system that invokes a LUA script of the dial plan to handle interaction on the call. All is fine when running a single instance of the script (one call). The issue is when there is more than one call things become erratic. Right now the script opens a log file who's name is based on the date of the call and its DNIS using io.open with append. The script then writes lines to file as it runs detailing its actions. The same log function also uses posix.syslog to log the same information via syslog. All is fine when there is one call up. If there is more than one concurrent on two different DNIS's both log files a present but have output that was created by the other call. The syslog is sane but not the file of logging information. The time stamp will jump backward and pickup with information from the other call (can tell via the UUID) It is my assumption that each call should spin up a thread and they should share no data/variables/handles with other calls in the system. Looking at the logs indicates there is some issue but logging is not the only issue since it seems that call events updating state information is affecting the other call also. If my assumption on a isolated thread per call is incorrect let me know. I inherited this code and suspect that running an application out of the dialplan script is unconventional. Paul

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