August 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Bug in spidermonkey-odbc-numRows
by Ashutosh 16 Aug '08

16 Aug '08

Hi, i observed that mod_spidermonkey_odbc function numRows has been returning me number of columns rather than rows. Looking deep revealed following: file : freeswitch-1.0.1/src/mod/languages/mod_spidermonkey_odbc/mod_spidermonkey_odbc.c * static JSBool odbc_num_rows(JSContext * cx, JSObject * obj, uintN argc, jsval * argv, jsval * rval) { .......... .......... if (odbc_obj->stmt) { SQLNumResultCols(odbc_obj->stmt, &rows); } ..... .........* I suppose the fn call in bold above might be problematic. Has anyone else countered it before ?I will try to resolve and put the patch if resolved. Thanks, ashutosh

2 1

mod_perl buid errors on Solaris
by Bruce McAlister 16 Aug '08

16 Aug '08

Hi All, I am trying to build mod_perl for use in FS on Solaris. The build completes successfully if I don't include mod_perl. However, when I do include mod_perl I receive the following compiler error: making all mod_perl Compiling freeswitch_perl.cpp... /opt/SUNWspro/bin/CC -w -DMULTIPLICITY -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -D_TS_ERRNO -I/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE -DEMBED_PERL -I/export/home/vzhsxn/packages/BUILD/freeswitch-1.0.1/src/include -I/export/home/vzhsxn/packages/BUILD/freeswitch-1.0.1/libs/libteletone/src -KPIC -DPIC -features=extensions -g -D_GNU_SOURCE -DHAVE_CONFIG_H -c -o freeswitch_perl.o freeswitch_perl.cpp "freeswitch_perl.cpp", line 53: Warning (Anachronism): Formal argument state_change of type extern "C" switch_status_t(*)(switch_core_session*) in call to switch_core_event_hook_remove_state_change(switch_core_session*, extern "C" switch_status_t(*)(switch_core_session*)) is being passed switch_status_t(*)(switch_core_session*). "freeswitch_perl.cpp", line 162: Warning (Anachronism): Formal argument state_change of type extern "C" switch_status_t(*)(switch_core_session*) in call to switch_core_event_hook_add_state_change(switch_core_session*, extern "C" switch_status_t(*)(switch_core_session*)) is being passed switch_status_t(*)(switch_core_session*). 2 Warning(s) detected. Compiling mod_perl_wrap.cpp... /opt/SUNWspro/bin/CC -w -DMULTIPLICITY -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -D_TS_ERRNO -I/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE -DEMBED_PERL -I/export/home/vzhsxn/packages/BUILD/freeswitch-1.0.1/src/include -I/export/home/vzhsxn/packages/BUILD/freeswitch-1.0.1/libs/libteletone/src -KPIC -DPIC -features=extensions -g -D_GNU_SOURCE -DHAVE_CONFIG_H -c -o mod_perl_wrap.o mod_perl_wrap.cpp Compiling perlxsi.c... /opt/SUNWspro/bin/cc -w -DMULTIPLICITY -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -D_TS_ERRNO -I/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE -DEMBED_PERL -I/export/home/vzhsxn/packages/BUILD/freeswitch-1.0.1/src/include -I/export/home/vzhsxn/packages/BUILD/freeswitch-1.0.1/libs/libteletone/src -KPIC -DPIC -DPATH_MAX=2048 -g -v -Xc -xc99=all -D_GNU_SOURCE -DHAVE_CONFIG_H -c -o perlxsi.o perlxsi.c "/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE/scope.h", line 221: syntax error before or at: sigjmp_buf "/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE/proto.h", line 199: syntax error before or at: off64_t "/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE/proto.h", line 201: syntax error before or at: Perl_do_sysseek "/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE/proto.h", line 201: syntax error before or at: off64_t "/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE/proto.h", line 202: syntax error before or at: Perl_do_tell "/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE/proto.h", line 1308: syntax error before or at: Perl_PerlIO_tell "/usr/perl5/5.8.4/lib/i86pc-solaris-64int/CORE/proto.h", line 1309: syntax error before or at: off64_t cc: acomp failed for perlxsi.c make[5]: *** [perlxsi.o] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_perl-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Steps to reproduce: [1] Install Solaris 10 Update 5 [2] Install Sun Studio 12 [3] Patched system with recommended patches as of 15-08-2008 [4] Install JDC CBE 1.7.0 RC1 [5] Download/Install SFE [6] Setup user with "Installation" permissions [7] Build and install SFEcurl/SFEunixodbc packages (including development packages) [8] Create SFEfreeswitch spec file (attached) [9] Run pkgtool -interactive -download -pkgformat=ds build-only SFEfreeswitch.spec Any comments suggestions would be greatly appreciated. Thanks Bruce

1 0

GSM gateway MV370
by Henk Oegema 15 Aug '08

15 Aug '08

I have a MV370 GSM gateway from Portech using in *. The extension number of the gateway is 2001 To dial via the gateway I use a prefix 9. ;GSM VIA MV-370 GATEWAY) exten => _9X.,1,Dial(SIP/2001/${EXTEN:1}) exten => _9X.,n,Congestion() This works well. Now I'm in the process of switching that gateway to a FS server. The gateway is also registered as number 2001 in FS. How do I translate the * dialplan to FS? (the two lines of code above) Which xml file ? (default.xml ?) Thanks Henk

2 1

Illegal character?
by Ben Holtsclaw 15 Aug '08

15 Aug '08

Earlier today, I was in the chat room trying to get help with a problem where one of my extensions didn't ring back when calling it. Any other extension would dial that extension, 305, and just hear dead air. 305 would actually ring, but when picked up, no sound was on the line. This is what the log looked like when trying to call 305: http://pastebin.freeswitch.org/5119 Here is the pastebin that shows a few of my extensions: http://pastebin.freeswitch.org/5126 You can see that extension 305 is registered with the caller id name Testing & Accountability. That is what I set in the caller id field on the Aastra 9143i phone. I finally figured out that if I remove the ampersand, the problem goes away. My question is... Is this a limitation (bug?) of the Aastra firmware, sofia, sip in general, or something in freeswitch? Ben Holtsclaw Network Engineer Avery County Schools Ph: 828.733.3567 x301

3 9

ANNOUNCEMENT: Mod of the week - mod_enum
by Michael Collins 15 Aug '08

15 Aug '08

Greetings! The documentation group met on Wednesday and decided that the next mod to work on is mod_enum. We'd like to ask all those willing and capable to assist. We want to have this page be the central point for all things mod_enum: http://wiki.freeswitch.org/wiki/Mod_enum There are other pages that have enum info, but we'd like to get this organized. If you are familiar with how enum works and can clean up the wiki a bit that would be well-received by the community. Please note these pages: http://wiki.freeswitch.org/wiki/ENUM_support <== has some enum info here http://wiki.freeswitch.org/wiki/ISN <== ISN is related to enum If you feel confident then jump right in and start fixing the enum pages. If you have questions then please join #freeswitch-docs and ask away. Thanks for your assistance! -MC P.S. - Thanks to the community for jumping in and adding content to the mod_cdr_csv page over the past two weeks!

1 0

Dialplan translation to FS.
by Henk Oegema 15 Aug '08

15 Aug '08

I still haven't been able to do this in FS: :-( exten => _9X.,1,Dial(SIP/2001/${EXTEN:1}) exten => _9X.,n,Congestion() Who can help please. Rgds Henk (p.s. I asked this earlier in: http://www.nabble.com/GSM-gateway-MV370-to18932091.html but no response)

3 5

TLS but OPTIONS requests in clear text from port 5060
by Peter P GMX 15 Aug '08

15 Aug '08

Hello, I have successfully set up Snom phones for TLS and SRTP so the whole communication should be encrypted. However I see OPTIONS SIP requests in clear text coming from Freeswitch. Is there a chance to avoid this? Best regards Peter

3 3

FreeSWITCH EC2 AMI
by Brian West 15 Aug '08

15 Aug '08

I'm currently putting together an EC2 image with FreeSWITCH on it. I'll be hosting a 1.0.1 image and try to load a weekly image of SVN trunk. I'll be incurring some costs associated with this. If anyone wants to donate to the cost of me hosting these images in my AWS account please contact me off list. Brian West sip:brian@freeswitch.org

3 4

context public , dialplan public.xml relation ?
by Lee JJ 14 Aug '08

14 Aug '08

Hello : Under sip_profiles directory these files "internal.xml , external.xml , nat.xml" all have a setting ==> <param name="context" value="public"/> Does the "public context" relate to the dialplan directory "public.xml" file ? if do , I can route to another dialplan file under some condition, do I ? " <extension name="public_extensions"> <condition field="destination_number" expression="^(10[01][0-9])$"> <action application="transfer" data="$1 XML *my-default*"/> " ; ^^^^^^^^^^^^^^^ if not , what should I do ? ps: Brian, thanks your effort here.

2 1

help getting FXO inbound to bridge to extension
by Ian Henderson 14 Aug '08

14 Aug '08

Hi, I'm tearing my hair out trying to get an inbound FXO call to bridge to a SIP extension. My setup is: Zapmicro ZMA400P (TDM400P equivalent). Cisco 7940 with SIP firmware FS 1.0.1 + zaptel 1.4.11 I'm able to make calls out from various SIP extensions via the ZMA400P FXO, however I can't get inbound calls to cause the SIP extension to ring and acknowledge the call. In variations of the dialplan I can get a delay_echo to work for the inbound call; The sample config for Local Extension will appear to transfer the call to the extension, but the extension doesn't ring; however after 30 seconds, the call diverts to voicemail successfully. I've tried transfers, bridging; default and public profiles with no success. I've been through the wiki. Does anyone have a sample dialplan, or the tricks to get FXO-to-extensions to work? thanks in advance, Ian. -- Be Yourself @ mail.com! Choose From 200+ Email Addresses Get a Free Account at www.mail.com

2 1