March 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

UK ring tone
by Gavin Henry 12 Mar '10

12 Mar '10

Hi, I've checked the internal and external sip profiles and think this needs to be set in autoload switch file as per: http://wiki.freeswitch.org/wiki/Variable_transfer_ringback I want by default a UK ringtone to be heard when I call a DDI that is configured on the system. At the moment it sounds like I'm calling the US. Thanks. -- http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com

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Thangappan Mohan sent you a little gift
by Thangappan Mohan 12 Mar '10

12 Mar '10

Thangappan Mohan is on Skoost and sent you a little gift. To collect your gift, follow the link below: http://www.skoost.com/?id=114961367_5284543 P.S. This is a safe and innocent gift that Thangappan Mohan sent from Skoost. Skoost

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The demo xml dialplan extension mad_boss_intercom problem
by Larry Vatland 11 Mar '10

11 Mar '10

I'm new to freeswitch and just exploring the default dialplan. I have not been able to get the mad_boss_intercom example to work correctly, it pages fine but you should be able to press *2 to start a two-way conversation. Whats happening is the * toggles the phones mute. How do I set up the conference to pass the *2 off to the bind_meta_app to get a page to intercom system working? Thanks for any help. Larry Vatland Computer World 3015 W. Wisconsin Ave. Appleton, WI 54914 (920) 733-9547 lvatland(a)gmail.com

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more than one condition default.xml
by srinivasula reddy 11 Mar '10

11 Mar '10

Hi, can i have more than one conditions in externsion tag in dialplan\default.xml. -- Srinivasula Reddy K

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No UPNP/NAT-PMP, no STUN! How can i make calls between two NAT?
by arliyardo 11 Mar '10

11 Mar '10

my scenario like this: PHONE1->FS (192.168.0.4) -> NAT1 (Public IP 1.2.3.4) -> INTERNET -> NAT2 (Public IP 5.6.7.8) -> PHONE2 (192.168.1.100) First, I have forwarded sip port and rtp port. I have try servaral configurations in http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios and without luck! PHONE2 only regists to FS, how can FS gets through NAT2 to PHONE2, include sip signaling and rtp connection ?

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can't get programming language modules to load
by Todd 11 Mar '10

11 Mar '10

Hey list- I have make mod_whatever-installed all the individual mod I want and have been able to make all but the following work. I have followed wiki directions for odbc and have odbc working. Any help in getting these modules to load would be great. Thanks **/usr/local/freeswitch/mod/mod_spidermonkey_core_db.so: undefined symbol: mod_spidermonkey_core_db_module_interface** 2010-03-10 16:55:50.990489 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_curl.so **/usr/local/freeswitch/mod/mod_spidermonkey_curl.so: undefined symbol: mod_spidermonkey_curl_module_interface** 2010-03-10 16:55:50.990632 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: undefined symbol: mod_spidermonkey_odbc_module_interface** 2010-03-10 16:55:50.990715 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_socket.so **/usr/local/freeswitch/mod/mod_spidermonkey_socket.so: undefined symbol: mod_spidermonkey_socket_module_interface** 2010-03-10 16:55:50.990796 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_spidermonkey_teletone.so **/usr/local/freeswitch/mod/mod_spidermonkey_teletone.so: undefined symbol: mod_spidermonkey_teletone_module_interface** 2010-03-10 16:55:50.990906 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_perl.so **/usr/local/freeswitch/mod/mod_perl.so: cannot open shared object file: No such file or directory** 2010-03-10 16:55:50.991020 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_python.so **/usr/local/freeswitch/mod/mod_python.so: cannot open shared object file: No such file or directory** 2010-03-10 16:55:50.991140 [CRIT] switch_loadable_module.c:872 Error Loading module /usr/local/freeswitch/mod/mod_java.so **/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No such file or directory** Thanks

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FS no frok mode
by Dome Charoenyost 11 Mar '10

11 Mar '10

Dear All, How about performance when start FS with -nf (no fork). I ask because i want to keep SQL connect (cash db) in 1 connection. but i want to know other effect like a performance , memory usage .. etc Best Regrads. Dome C.

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Kernel timer warning with CentOS 5.4
by Bruce Hopkins 11 Mar '10

11 Mar '10

Hi, I wonder if anyone would be able to advise please: When I build FreeSWITCH on yum updated CentOS 5.4, I get a warning when I start FreeSWITCH that "Abnormally large timer gap detected" "Do you have your kernel timer set to greater than 1kHz? You may experience audio problems". I get no such warning if I build on CentOS 5.3, and the test timings it measures on starting FreeSWITCH do look lower. All I was doing to upgrade to Centos5.4 was a yum update on the 5.3 build. I guess the warning comes from here: http://fisheye.freeswitch.org/browse/~raw,r=16409/FreeSWITCH/src/switch_tim… This is all on pretty low spec hardware - a couple of different Dell optiplex p4's I use for testing. Does anyone happen to know if I should just stick to Cent)S 5.3, or use 5.4 and not worry about the warnings, or if there is something I can do to fix the problem it is warning about. Perhaps it is just that I shouldn't use such crummy hardware?! Many thanks in advance Bruce

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The demo extension mad_boss_intercom problem
by Larry Vatland 11 Mar '10

11 Mar '10

I'm new to freeswitch and just exploring the default dialplan. I have not been able to get the mad_boss_intercom example to work correctly, it pages fine but you should be able to press *2 to start a two-way conversation. Whats happening is the * toggles the phones mute. How do I set up the conference to pass the *2 off to the bind_meta_app to get a page to intercom system working? Thanks for any help. Larry Vatland Computer World 3015 W. Wisconsin Ave. Appleton, WI 54914 (920) 733-9547 lvatland(a)gmail.com

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openzap Q.SIG and Q.931
by Helmut Kuper 11 Mar '10

11 Mar '10

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, maybe you know that I run a long time test with Stefan Knoblich's q931 pri-stack for openzap. In fact, I use is in my production system since a year or so. I have to say that it is very stable and I'm happy with it - ususally ... but: I wonder if anyone of you is working on a Q.SIG Basic-Call Support (ECMA-143) for openzap. I try to connect to an Avaya system using Q.SIG to be a part of the internal domain. My main targets are to send and receive Caller's and Callee's name and to remove a leading '0' (which is added by AVAYA because of q931) for internal calls. regards helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFLmKsh4tZeNddg3dwRAjaNAJ4pTdxYyQizHPDXP9gjjUyABJIuxwCcDnNj qlCG9cdIfjDLlQE3bQfxluw= =L1tJ -----END PGP SIGNATURE-----

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