Re: [Freeswitch-users] mod_opal
hhmmm :)) is there any doc following up mod_opal ? I really don't want to waste your time :) T. On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins m...@freeswitch.org wrote: On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga tculj...@gmail.comwrote: hello, i'm trying to build mod_opal and getting this error: making all mod_logfile making all mod_loopback making all mod_native_file making all mod_opal Compiling mod_opal.cpp... quiet_libtool: compile: g++ -g -ggdb -I. -I/home/tculjaga/freeswitch-trunk/src/include -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal -DHAVE_CONFIG_H -c mod_opal.cpp -fPIC -DPIC -o .libs/mod_opal.o In file included from mod_opal.cpp:25: mod_opal.h:151: error: conflicting return type specified for ‘virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall, void*)’ /usr/include/opal/opal/localep.h:267: error: overriding ‘virtual ptlib_virtual_function_changed_or_removed** OpalLocalEndPoint::CreateConnection(OpalCall, void*)’ mod_opal.cpp: In constructor ‘FSConnection::FSConnection(OpalCall, FSEndPoint, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)’: mod_opal.cpp:564: error: no matching function for call to ‘OpalLocalConnection::OpalLocalConnection(OpalCall, FSEndPoint, NULL)’ /usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall, OpalLocalEndPoint, void*, unsigned int, OpalConnection::StringOptions*, char) /usr/include/opal/opal/localep.h:276: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection) /usr/include/opal/opal/patch.h: In member function ‘switch_status_t FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)’: /usr/include/opal/opal/patch.h:272: error: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is private mod_opal.cpp:1277: error: within this context mod_opal.cpp:1277: warning: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is deprecated (declared at /usr/include/opal/opal/patch.h:272) mod_opal.cpp:1277: warning: ignoring return value of function declared with attribute warn_unused_result /usr/include/opal/opal/patch.h: In member function ‘switch_status_t FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)’: /usr/include/opal/opal/patch.h:272: error: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is private mod_opal.cpp:1399: error: within this context mod_opal.cpp:1399: warning: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is deprecated (declared at /usr/include/opal/opal/patch.h:272) mod_opal.cpp:1399: warning: ignoring return value of function declared with attribute warn_unused_result make[5]: *** [mod_opal.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_opal-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +--+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 tculj...@nemesis:~/freeswitch-trunk$ tculj...@nemesis:~/freeswitch-trunk$ tculj...@nemesis:~/freeswitch-trunk$ what ptlib/opal/fs version did you use to build it? I tried with trunk (ptlib, opal, fs)... and as you can see :) Did you run the buildopal.sh script in src/build ? -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_opal
hi, It went well obviously FS needs v3_6 opal :) thx. On Tue, Sep 1, 2009 at 8:09 AM, Tihomir Culjaga tculj...@gmail.com wrote: hhmmm :)) is there any doc following up mod_opal ? I really don't want to waste your time :) T. On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins m...@freeswitch.orgwrote: On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga tculj...@gmail.comwrote: hello, i'm trying to build mod_opal and getting this error: making all mod_logfile making all mod_loopback making all mod_native_file making all mod_opal Compiling mod_opal.cpp... quiet_libtool: compile: g++ -g -ggdb -I. -I/home/tculjaga/freeswitch-trunk/src/include -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal -DHAVE_CONFIG_H -c mod_opal.cpp -fPIC -DPIC -o .libs/mod_opal.o In file included from mod_opal.cpp:25: mod_opal.h:151: error: conflicting return type specified for ‘virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall, void*)’ /usr/include/opal/opal/localep.h:267: error: overriding ‘virtual ptlib_virtual_function_changed_or_removed** OpalLocalEndPoint::CreateConnection(OpalCall, void*)’ mod_opal.cpp: In constructor ‘FSConnection::FSConnection(OpalCall, FSEndPoint, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)’: mod_opal.cpp:564: error: no matching function for call to ‘OpalLocalConnection::OpalLocalConnection(OpalCall, FSEndPoint, NULL)’ /usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall, OpalLocalEndPoint, void*, unsigned int, OpalConnection::StringOptions*, char) /usr/include/opal/opal/localep.h:276: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection) /usr/include/opal/opal/patch.h: In member function ‘switch_status_t FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)’: /usr/include/opal/opal/patch.h:272: error: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is private mod_opal.cpp:1277: error: within this context mod_opal.cpp:1277: warning: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is deprecated (declared at /usr/include/opal/opal/patch.h:272) mod_opal.cpp:1277: warning: ignoring return value of function declared with attribute warn_unused_result /usr/include/opal/opal/patch.h: In member function ‘switch_status_t FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)’: /usr/include/opal/opal/patch.h:272: error: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is private mod_opal.cpp:1399: error: within this context mod_opal.cpp:1399: warning: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is deprecated (declared at /usr/include/opal/opal/patch.h:272) mod_opal.cpp:1399: warning: ignoring return value of function declared with attribute warn_unused_result make[5]: *** [mod_opal.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_opal-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +--+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 tculj...@nemesis:~/freeswitch-trunk$ tculj...@nemesis:~/freeswitch-trunk$ tculj...@nemesis:~/freeswitch-trunk$ what ptlib/opal/fs version did you use to build it? I tried with trunk (ptlib, opal, fs)... and as you can see :) Did you run the buildopal.sh script in src/build ? -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_opal
On Mon, Aug 31, 2009 at 11:09 PM, Tihomir Culjaga tculj...@gmail.comwrote: hhmmm :)) is there any doc following up mod_opal ? I really don't want to waste your time :) T. Last I heard this was it: http://jira.freeswitch.org/browse/MODOPAL-10 -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] remote endpoints
oh good, on remote router/dsl modem (whatever doing NAT) never use upnp, never use ALG, just do a simple NAT and it is alway gonna work! T. On Tue, Sep 1, 2009 at 2:06 AM, e schmidbauer e.schmidba...@gmail.comwrote: i cannot reach the remote endpoint. the remote endpoint can reach a locally registered endpoint. any idea why this is happening? the remote endpoint is behind a linksys with upnp enabled. 2009/8/31 João Mesquita jmesqu...@freeswitch.org: Problem is definetly on far end. If you look at the siptrace, you have the following sequence: 1. Asterisk calls in 2. FreeSWITCH replies with a Trying(100) to complete call right away and proceeds to dialplan 3. FreeSWITCH invites (calls) 7 times the final destination that never responds. 4. Asterisk sends a CANCEL message In all that, your final endpoint never responds to any message. Are you sure you can reach it? jmesquita On Mon, Aug 31, 2009 at 5:18 PM, e schmidbauer e.schmidba...@gmail.com wrote: here's the sip trace: http://pastebin.freeswitch.org/10172 2009/8/31 João Mesquita jmesqu...@freeswitch.org: 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel sofia/external/anonym...@anonymous.invalid entering state [terminated][487] The far end seems to be replying with 487 - Request Terminated... Nothing wrong on FS, seems to be a problem with your endpoints. Can you enable a sip trace? jmesquita On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer e.schmidba...@gmail.com wrote: thanks...heres the pastebin: http://pastebin.freeswitch.org/10171 2009/8/31 João Mesquita jmesqu...@freeswitch.org: Check the password dialog. It will tell you what the username/password is. post the logs for a call as well, please. Regards, jmesquita On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer e.schmidba...@gmail.com wrote: = Nameexternal Domain Name N/A DBName sofia_reg_external Pres Hosts DialplanXML Context public Challenge Realm auto_to RTP-IP 192.168.0.125 Ext-RTP-IP 98.118.151.30 SIP-IP 192.168.0.125 Ext-SIP-IP 98.118.151.30 URL sip:mod_so...@192.168.0.125:5080 BIND-URLsip:mod_so...@192.168.0.125:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS c...@48000h,c...@32000h,sp...@32000h@20i,sp...@16000h @20i,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGfalse PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDtrue STUN-AUTO-DISABLE false CALLS-IN17 FAILED-CALLS-IN 11 CALLS-OUT 9 FAILED-CALLS-OUT9 Registrations: = Call-ID:MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI. User: 1...@server1.altpressonline.com Contact:1000 sip:1...@69.204.30.67:16006;rinstance=50cbd6140e1ab991 Agent: X-Lite release 1103k stamp 53621 Status: Registered(UDP)(unknown) EXP(2009-08-31 16:24:11) Host: server1.altpressonline.com IP: 69.204.30.67 Port: 16006 Auth-User: 1000 Auth-Realm: server1.altpressonline.com = sorry, i don't know how to login to pastebin On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins m...@freeswitch.org wrote: Could you give a few more details? For example, could you pastebin the output of sofia status profile external? -MC On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer e.schmidba...@gmail.com wrote: I am unable to call a user outside of our local area network. the user is registered on the external profile but there is no way to call the phone. does anyone have any suggestions how to do this? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] problem compiling esl for use with freepbx v3
Did the simple make in the libs/esl directory run properly? Just curious. I'll have to defer to the Ubuntu gurus out there for thoughts on what else could be wrong. -MC On Mon, Aug 31, 2009 at 10:43 PM, Harondel J. Sibble h...@pdscc.com wrote: Haven't had any responses, anyone have any ideas on the problem with compiling the ESL modules as below? On 23 Aug 2009 at 11:51, Harondel J. Sibble wrote: Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server, then went to install FreePBX v3, I've gotten all the prerequisities in the wizard fixed except for ESL As per http://wiki.freeswitch.org/wiki/Event_Socket_Library http://wiki.freeswitch.org/wiki/Event_Socket I go into my FS source dir /home/sibbleh/freeswitch-1.0.4/libs/esl Run make and then sudo make phpmod-install and I get $ sudo make phpmod-install make MYLIB=../libesl.a SOLINK=-shared -Xlinker -x CFLAGS=- I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused- variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes CXXFLAGS=- I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable CXX_CFLAGS= -C php make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php' g++ -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable - I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM - I/usr/include/php5/Zend -I/usr/include/php5/ext - I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 - Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o cc1plus: warnings being treated as errors esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1047: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1073: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int, zval*, zval**, zval*, int)': esl_wrap.cpp:: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1141: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1172: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1198: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1234: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1269: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1294: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1346: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1403: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1441: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1478: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1508: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1538: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1571: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_addHeader(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1611: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void
Re: [Freeswitch-users] mod_opal
Please look into MODOPAL-10 in jira. You need to apply a patch if you're using latest opal trunk, ro else you need to use the latest stable version of opal. However, I'm not sure how automated this is in the build process in Linux. I've only done this on Windows builds lately. /Peter Från: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] För Tihomir Culjaga Skickat: den 1 september 2009 08:09 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] mod_opal hhmmm :)) is there any doc following up mod_opal ? I really don't want to waste your time :) T. On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins m...@freeswitch.orgmailto:m...@freeswitch.org wrote: On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga tculj...@gmail.commailto:tculj...@gmail.com wrote: hello, i'm trying to build mod_opal and getting this error: making all mod_logfile making all mod_loopback making all mod_native_file making all mod_opal Compiling mod_opal.cpp... quiet_libtool: compile: g++ -g -ggdb -I. -I/home/tculjaga/freeswitch-trunk/src/include -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal -DHAVE_CONFIG_H -c mod_opal.cpp -fPIC -DPIC -o .libs/mod_opal.o In file included from mod_opal.cpp:25: mod_opal.h:151: error: conflicting return type specified for 'virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall, void*)' /usr/include/opal/opal/localep.h:267: error: overriding 'virtual ptlib_virtual_function_changed_or_removed** OpalLocalEndPoint::CreateConnection(OpalCall, void*)' mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall, FSEndPoint, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)': mod_opal.cpp:564: error: no matching function for call to 'OpalLocalConnection::OpalLocalConnection(OpalCall, FSEndPoint, NULL)' /usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall, OpalLocalEndPoint, void*, unsigned int, OpalConnection::StringOptions*, char) /usr/include/opal/opal/localep.h:276: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection) /usr/include/opal/opal/patch.h: In member function 'switch_status_t FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)': /usr/include/opal/opal/patch.h:272: error: 'virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()' is private mod_opal.cpp:1277: error: within this context mod_opal.cpp:1277: warning: 'virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()' is deprecated (declared at /usr/include/opal/opal/patch.h:272) mod_opal.cpp:1277: warning: ignoring return value of function declared with attribute warn_unused_result /usr/include/opal/opal/patch.h: In member function 'switch_status_t FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)': /usr/include/opal/opal/patch.h:272: error: 'virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()' is private mod_opal.cpp:1399: error: within this context mod_opal.cpp:1399: warning: 'virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()' is deprecated (declared at /usr/include/opal/opal/patch.h:272) mod_opal.cpp:1399: warning: ignoring return value of function declared with attribute warn_unused_result make[5]: *** [mod_opal.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_opal-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +--+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 tculj...@nemesis:~/freeswitch-trunk$ tculj...@nemesis:~/freeswitch-trunk$ tculj...@nemesis:~/freeswitch-trunk$ what ptlib/opal/fs version did you use to build it? I tried with trunk (ptlib, opal, fs)... and as you can see :) Did you run the buildopal.sh script in src/build ? -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto:FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4a9cbdb632933764890742! ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
[Freeswitch-users] FS performance under windows
Hi folk, First of all, thank you for FS - really strong project. I have already asked this once in other thread but didn't got any answer. So, I'll try to re-ask. We are playing currently with FS under Windows 2008 64bit. So far there are some issues but I hope we'll solve it in nearest future. After FS will be configured correctly we plan to play with performance things on FS. The question is: Does it makes any sense to try to setup FS under Win for a same performance level possible under Linux (e.g. CentOs)? Or it's just wasting of time? An additional question is: Are there any important and well know issues during migration from Win to Lin. Or it is just like copying of all configs into Linux installation? Thank you -- Best regards, Dmitry Kadantsev ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS performance under windows
If you want to try FS on Windows only for feature testing etc. then its okay, however for production deployments (that includes load testing) i strongly recommend CentOS 5.x. As far as configuration migration is concerned, you don't need to change any configuration files, simply copy them to Linux installation. Thank you. On Tue, Sep 1, 2009 at 2:13 PM, Dmitry Kadantsev kadantse...@gmail.comwrote: Hi folk, First of all, thank you for FS - really strong project. I have already asked this once in other thread but didn't got any answer. So, I'll try to re-ask. We are playing currently with FS under Windows 2008 64bit. So far there are some issues but I hope we'll solve it in nearest future. After FS will be configured correctly we plan to play with performance things on FS. The question is: Does it makes any sense to try to setup FS under Win for a same performance level possible under Linux (e.g. CentOs)? Or it's just wasting of time? An additional question is: Are there any important and well know issues during migration from Win to Lin. Or it is just like copying of all configs into Linux installation? Thank you -- Best regards, Dmitry Kadantsev ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] SRTP Encryption
Hi, i have a problem using SRTP Encrytion. All intern calls are SRTP encrypted. Some of my Gateway don´t support SRTP encryption. In my dialplan I now set the sip_secure_media to false. action application=set data=sip_secure_media=false/ It works. But is there any chance to encrypt the call on one side and use a unencrypted call on the other side of the freeswitch? Phone --(SRTP)-- Freeswitch --(RTP)-- Gateway Thanks for help NOx -- View this message in context: http://www.nabble.com/SRTP-Encryption-tp25237296p25237296.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] sofia_reg_external in odbc?
Hello Brian, I've done this. FS creates the tables sccessfully, but then doesn't fill them. isql: SQL show tables; +-+ | Tables_in_fs_external | +-+ | sip_authentication | | sip_dialogs | | sip_presence | | sip_registrations | | sip_shared_appearance_dialogs | | sip_shared_appearance_subscriptions | | sip_subscriptions | +-+ SQLRowCount returns 7 7 rows fetched Is that right, that the tables have the same structure as for the internal database? sofia status shows 7 registered external gateways, but none of them is shown in the ODBC database. All tables are empty. Any idea? Best regrads Peter Brian West schrieb: param name=odbc-dsn value=dsn:user:pass/ On the profile. /b On Aug 30, 2009, at 5:25 PM, Peter P GMX wrote: Hello, is there a chance to have sofia_reg_external in odbc/mysql instead of sqlite? In a B2BUA environment we have thousand of external registrations during a migration phase, and it would be good to have easy external control over the registered gateways (like in fs_internal. sip_registrations). Best regards Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SRTP Encryption
Sure this works, you can set rtp or srtp independently to every call leg (if FS is in media path) and even mix them in a conference. Best regards Peter NOx-WHV schrieb: Hi, i have a problem using SRTP Encrytion. All intern calls are SRTP encrypted. Some of my Gateway don´t support SRTP encryption. In my dialplan I now set the sip_secure_media to false. action application=set data=sip_secure_media=false/ It works. But is there any chance to encrypt the call on one side and use a unencrypted call on the other side of the freeswitch? Phone --(SRTP)-- Freeswitch --(RTP)-- Gateway Thanks for help NOx ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] problem compiling esl for use with freepbx v3
I gave up on compiling esl, I got a bunch of errors, there were several people on the list with problems and apparently no straight solution, especially for php-esl. I am now using a ruby library, posted here by Diego Viola I believe. On Tue, Sep 1, 2009 at 2:33 AM, Michael Collins m...@freeswitch.org wrote: Did the simple make in the libs/esl directory run properly? Just curious. I'll have to defer to the Ubuntu gurus out there for thoughts on what else could be wrong. -MC On Mon, Aug 31, 2009 at 10:43 PM, Harondel J. Sibble h...@pdscc.comwrote: Haven't had any responses, anyone have any ideas on the problem with compiling the ESL modules as below? On 23 Aug 2009 at 11:51, Harondel J. Sibble wrote: Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server, then went to install FreePBX v3, I've gotten all the prerequisities in the wizard fixed except for ESL As per http://wiki.freeswitch.org/wiki/Event_Socket_Library http://wiki.freeswitch.org/wiki/Event_Socket I go into my FS source dir /home/sibbleh/freeswitch-1.0.4/libs/esl Run make and then sudo make phpmod-install and I get $ sudo make phpmod-install make MYLIB=../libesl.a SOLINK=-shared -Xlinker -x CFLAGS=- I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused- variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes CXXFLAGS=- I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable CXX_CFLAGS= -C php make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php' g++ -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable - I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM - I/usr/include/php5/Zend -I/usr/include/php5/ext - I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 - Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o cc1plus: warnings being treated as errors esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1047: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1073: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int, zval*, zval**, zval*, int)': esl_wrap.cpp:: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1141: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1172: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1198: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1234: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1269: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1294: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1346: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1403: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1441: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1478: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1508: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*, zval**, zval*, int)': esl_wrap.cpp:1538: error: format not a string literal and no format arguments esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*, zval**, zval*,
Re: [Freeswitch-users] SRTP Encryption
How can I see if the FS is in media path? Or how can i set the FS in media path? Peter P GMX wrote: Sure this works, you can set rtp or srtp independently to every call leg (if FS is in media path) and even mix them in a conference. Best regards Peter NOx-WHV schrieb: Hi, i have a problem using SRTP Encrytion. All intern calls are SRTP encrypted. Some of my Gateway don´t support SRTP encryption. In my dialplan I now set the sip_secure_media to false. action application=set data=sip_secure_media=false/ It works. But is there any chance to encrypt the call on one side and use a unencrypted call on the other side of the freeswitch? Phone --(SRTP)-- Freeswitch --(RTP)-- Gateway Thanks for help NOx ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://www.nabble.com/SRTP-Encryption-tp25237296p25238144.html Sent from the Freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS performance under windows
Thank you! -- Best regards, Dmitry Kadantsev http://www.doxwox.com - Best web meeting and online collaboration tool. On Tue, Sep 1, 2009 at 11:00 AM, Muhammad Shahzad shaherya...@googlemail.com wrote: If you want to try FS on Windows only for feature testing etc. then its okay, however for production deployments (that includes load testing) i strongly recommend CentOS 5.x. As far as configuration migration is concerned, you don't need to change any configuration files, simply copy them to Linux installation. Thank you. On Tue, Sep 1, 2009 at 2:13 PM, Dmitry Kadantsev kadantse...@gmail.comwrote: Hi folk, First of all, thank you for FS - really strong project. I have already asked this once in other thread but didn't got any answer. So, I'll try to re-ask. We are playing currently with FS under Windows 2008 64bit. So far there are some issues but I hope we'll solve it in nearest future. After FS will be configured correctly we plan to play with performance things on FS. The question is: Does it makes any sense to try to setup FS under Win for a same performance level possible under Linux (e.g. CentOs)? Or it's just wasting of time? An additional question is: Are there any important and well know issues during migration from Win to Lin. Or it is just like copying of all configs into Linux installation? Thank you -- Best regards, Dmitry Kadantsev ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.com Email: shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS performance under windows
Do you have any specific notes why production or load testing isn’t recommended on Windows? Or just lack of data? From: freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: Tuesday, September 01, 2009 3:00 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] FS performance under windows If you want to try FS on Windows only for feature testing etc. then its okay, however for production deployments (that includes load testing) i strongly recommend CentOS 5.x. As far as configuration migration is concerned, you don't need to change any configuration files, simply copy them to Linux installation. Thank you. On Tue, Sep 1, 2009 at 2:13 PM, Dmitry Kadantsev kadantse...@gmail.commailto:kadantse...@gmail.com wrote: Hi folk, First of all, thank you for FS - really strong project. I have already asked this once in other thread but didn't got any answer. So, I'll try to re-ask. We are playing currently with FS under Windows 2008 64bit. So far there are some issues but I hope we'll solve it in nearest future. After FS will be configured correctly we plan to play with performance things on FS. The question is: Does it makes any sense to try to setup FS under Win for a same performance level possible under Linux (e.g. CentOs)? Or it's just wasting of time? An additional question is: Are there any important and well know issues during migration from Win to Lin. Or it is just like copying of all configs into Linux installation? Thank you -- Best regards, Dmitry Kadantsev ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.orgmailto:FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Muhammad Shahzad --- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_78...@hotmail.commailto:shari_78...@hotmail.com Email: shaherya...@googlemail.commailto:shaherya...@googlemail.com ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SRTP Encryption
If you do not explicitely set bypass_media to true, then FS is in the media path. Best regards Peter NOx-WHV schrieb: How can I see if the FS is in media path? Or how can i set the FS in media path? Peter P GMX wrote: Sure this works, you can set rtp or srtp independently to every call leg (if FS is in media path) and even mix them in a conference. Best regards Peter NOx-WHV schrieb: Hi, i have a problem using SRTP Encrytion. All intern calls are SRTP encrypted. Some of my Gateway don´t support SRTP encryption. In my dialplan I now set the sip_secure_media to false. action application=set data=sip_secure_media=false/ It works. But is there any chance to encrypt the call on one side and use a unencrypted call on the other side of the freeswitch? Phone --(SRTP)-- Freeswitch --(RTP)-- Gateway Thanks for help NOx ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@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] Bind_meta_app and second-degree bridge (.. this is a bad title.)
My basic functionality is this: A calls in, is bridged to B (). I use bind_meta_app to let B rebridge A to C (). After having been rebridged to C, C should be able to rebridge A to B *again*, and so on. This is the code I have: context name=public extension name=ff-ivr condition field=destination_number expression=^(.*)$ action application=set data=bypass_media=false/ action application=answer/ action application=bind_meta_app data=1 b a bridge::sofia/gateway/gw1// action application=bind_meta_app data=2 b a bridge::sofia/gateway/gw1// action application=bridge data=sofia/gateway/gw1/ / /condition /extension /context The first bridge is fine, and B can press *2 to bridge to C/. But if C presses *1, it seems to execute the bridge app, but nothing at all happens: 2009-09-01 11:14:59.258325 [DEBUG] switch_rtp.c: RTP RECV DTMF *:2000 2009-09-01 11:15:00.118195 [DEBUG] switch_rtp.c: RTP RECV DTMF 1:2000 2009-09-01 11:15:00.118195 [DEBUG] switch_ivr_async.c:1725 sofia/external/unkn...@129.142.224.250 Processing meta digit '2' [bridge::sofia/gateway/gw1/] 2009-09-01 11:15:00.118195 [DEBUG] switch_core_session.c:813 Send signal sofia/external/unkn...@129.142.224.250 [BREAK] Any ideas? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_voicemail email template variables
I tried doing a set right before the application is called to make a customer
variable but it doesn't get transferred to the template this way either:
---dialplan snip---
action application=set data=test_var=this is a test/
action application=voicemail data=default $${domain} $1/
---end snip---
---voicemail.tpl snip---
Created: ${voicemail_time}
${test_var}
From: ${voicemail_caller_id_name} ${sip_from_user_stripped}
Duration: ${voicemail_message_len}
---end snip---
---resultant email snip---
Created: Tuesday, September 01 2009, 08 30 AM
From: LEMBERGER,NICK
Duration: 00:00:07
---end snip---
Notice I also tried the channel variable ${sip_from_user_stripped} as it should
be available as well, at least according to the 'info' app. Any ideas?
-Nick
Anthony Minessale anthony.miness...@gmail.com 08/27/09 1:54 PM
you should be able to for instance put
action application=set data=test_var=this is a test/
right before the voicemail app is called
then put
${test_var} in your template.
making sure to issue reloadxml or restart FS
On Thu, Aug 27, 2009 at 1:06 PM, Nick Lemberger nick.lember...@lkfd.netwrote:
Thanks for the fast reply!
I just tried 10 random variables from
http://wiki.freeswitch.org/wiki/Channel_Variables and I only see the
whitespace where the variable should be. I've only been able to get the
ones that are set in mod_voicemail.c circa line 1600 to work.
-Nick
On 8/27/2009 at 12:44 PM, in message
191c3a030908271044k63973088xeec12c578d02e...@mail.gmail.com, Anthony
Minessale anthony.miness...@gmail.com wrote:
all variables referenced in the template should expand when sending the
email.
On Thu, Aug 27, 2009 at 12:41 PM, Nick Lemberger
nick.lember...@lkfd.netwrote:
Is there a way to use dialplan variables in the email that gets sent
with
the voicemail attachement. I tried using some but nothing seems to show
up,
I'm guessing it's a different channel or something...
Any ideas?
Thanks,
-Nick
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Re: [Freeswitch-users] sofia_reg_external in odbc?
gateways do not go into the table... ONLY inbound registrations to the profile do. /b On Sep 1, 2009, at 5:16 AM, Peter P GMX wrote: Hello Brian, I've done this. FS creates the tables sccessfully, but then doesn't fill them. isql: SQL show tables; +-+ | Tables_in_fs_external | +-+ | sip_authentication | | sip_dialogs | | sip_presence | | sip_registrations | | sip_shared_appearance_dialogs | | sip_shared_appearance_subscriptions | | sip_subscriptions | +-+ SQLRowCount returns 7 7 rows fetched Is that right, that the tables have the same structure as for the internal database? sofia status shows 7 registered external gateways, but none of them is shown in the ODBC database. All tables are empty. Any idea? Best regrads Peter ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] SRTP Encryption
Try this one. Outbound action application=export data=nolocal:sip_secure_media=false/ Inbound action application=export data=nolocal:sip_secure_media=true/ /b On Sep 1, 2009, at 4:40 AM, NOx-WHV wrote: action application=set data=sip_secure_media=false/ ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] Incorrect method of PHP call control?
Thanks for the input. You'll have to decide on static vs. dynamic based on your needs. In either case, once the call is connected to your socket you've got all sorts of control options. PHP has an ESL abstraction just like the other languages so there shouldn't be any issue about PHP lacking the ability to control calls. But I'm having a hard time seeing how the ESL would duplicate this JS functionality: session.collectInput(onInputsml, emptyobject, 7000); How do I set the PHP callback routine, etc.? -- Greg Thoen On Aug 31, 2009, at 12:55 PM, Michael Collins wrote: On Mon, Aug 31, 2009 at 8:22 AM, Greg Thoen gr...@cgicommunications.com wrote: Hi. Before I go to far down this path, I wonder if what I intend to do is not a good practice. I started using mod_xml_curl to use PHP on localhost to generate a dialplan dynamically, based on the Caller-Destination-Number variable that is posted. It prints out the XML that calls the javascript that then controls the call. For example, $response = XML ?xml version=1.0 encoding=UTF-8 standalone=no? document type=freeswitch/xml section name=dialplan description=example_curl_dialplan context name=public extension name=curl_test condition field=destination_number expression=^(\+1|1?) (5844111)$ action application=javascript data=demos/stest-examp- st.js / /condition /extension /context /section /document XML; Then I thought, that's silly to go back out to javascript to handle the actions, playing files, using pocketsphinx, etc. I should just stay in PHP, using esl.php to answer and handle the call. Then I rethought, is that a good practice to take over the call control from freeswitch at that point, while it is in the xml-curl dialplan hunt? Then I also thought, is it even possible to do some of the things I need to do from the php esl, like the equivalent of this javascript: session.collectInput(onInputsml, emptyobject, 7000); -- Greg Thoen Just remember that you're dealing with two somewhat related but still distinctly separate entities: generating a dialplan and executing some sort of call control from the dialplan. You need some sort of dialplan no matter what, so the issue there is whether you need a dynamic one or not. If you're just going to drop calls to an extension that opens an outbound socket to your call control program then you may not need the dynamic dp generation that mod_xml_curl gives you. You'll have to decide on static vs. dynamic based on your needs. In either case, once the call is connected to your socket you've got all sorts of control options. PHP has an ESL abstraction just like the other languages so there shouldn't be any issue about PHP lacking the ability to control calls. I say start hacking away at it and see what happens. :) Definitely join us in #freeswitch on irc.freenode.net if you want to discuss this more in realtime. -MC ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] FS performance under windows
Have you gotten past the problems with pthread-win32 on 64 bit? you will need the trunk version of that library if not because the released version has problems with 64bit. There are some other simple compilation problems I assume you may have already got past? If not see http://jira.freeswitch.org/browse/FSBUILD-147 for a reference. That bug is basically waiting for pthread-win32 to release their next version. What other kinds of problems are you having? Dmitry Kadantsev wrote: Hi folk, First of all, thank you for FS - really strong project. I have already asked this once in other thread but didn't got any answer. So, I'll try to re-ask. We are playing currently with FS under Windows 2008 64bit. So far there are some issues but I hope we'll solve it in nearest future. After FS will be configured correctly we plan to play with performance things on FS. The question is: Does it makes any sense to try to setup FS under Win for a same performance level possible under Linux (e.g. CentOs)? Or it's just wasting of time? An additional question is: Are there any important and well know issues during migration from Win to Lin. Or it is just like copying of all configs into Linux installation? Thank you -- Best regards, Dmitry Kadantsev ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- View this message in context: http://n2.nabble.com/FS-performance-under-windows-tp3559027p3560840.html Sent from the freeswitch-users mailing list archive at Nabble.com. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] Mod_fifo posision in queue
Dear sir, I want to say posision in queue to caller but fifo_chime_list can't say more than one sound file. i try fifo_chime_list = queue/say1.wav,queue/say2.wav. Best Regards. Dome C. ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] remote endpoints
I put tomato on the router and still no success. upnp is enabled, should i disable it? what do you mean by simple NAT? On Tue, Sep 1, 2009 at 2:32 AM, Tihomir Culjagatculj...@gmail.com wrote: oh good, on remote router/dsl modem (whatever doing NAT) never use upnp, never use ALG, just do a simple NAT and it is alway gonna work! T. On Tue, Sep 1, 2009 at 2:06 AM, e schmidbauer e.schmidba...@gmail.com wrote: i cannot reach the remote endpoint. the remote endpoint can reach a locally registered endpoint. any idea why this is happening? the remote endpoint is behind a linksys with upnp enabled. 2009/8/31 João Mesquita jmesqu...@freeswitch.org: Problem is definetly on far end. If you look at the siptrace, you have the following sequence: 1. Asterisk calls in 2. FreeSWITCH replies with a Trying(100) to complete call right away and proceeds to dialplan 3. FreeSWITCH invites (calls) 7 times the final destination that never responds. 4. Asterisk sends a CANCEL message In all that, your final endpoint never responds to any message. Are you sure you can reach it? jmesquita On Mon, Aug 31, 2009 at 5:18 PM, e schmidbauer e.schmidba...@gmail.com wrote: here's the sip trace: http://pastebin.freeswitch.org/10172 2009/8/31 João Mesquita jmesqu...@freeswitch.org: 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel sofia/external/anonym...@anonymous.invalid entering state [terminated][487] The far end seems to be replying with 487 - Request Terminated... Nothing wrong on FS, seems to be a problem with your endpoints. Can you enable a sip trace? jmesquita On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer e.schmidba...@gmail.com wrote: thanks...heres the pastebin: http://pastebin.freeswitch.org/10171 2009/8/31 João Mesquita jmesqu...@freeswitch.org: Check the password dialog. It will tell you what the username/password is. post the logs for a call as well, please. Regards, jmesquita On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer e.schmidba...@gmail.com wrote: = Name external Domain Name N/A DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 192.168.0.125 Ext-RTP-IP 98.118.151.30 SIP-IP 192.168.0.125 Ext-SIP-IP 98.118.151.30 URL sip:mod_so...@192.168.0.125:5080 BIND-URL sip:mod_so...@192.168.0.125:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS c...@48000h,c...@32000h,sp...@32000h@20i,sp...@16000h@20i,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 17 FAILED-CALLS-IN 11 CALLS-OUT 9 FAILED-CALLS-OUT 9 Registrations: = Call-ID: MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI. User: 1...@server1.altpressonline.com Contact: 1000 sip:1...@69.204.30.67:16006;rinstance=50cbd6140e1ab991 Agent: X-Lite release 1103k stamp 53621 Status: Registered(UDP)(unknown) EXP(2009-08-31 16:24:11) Host: server1.altpressonline.com IP: 69.204.30.67 Port: 16006 Auth-User: 1000 Auth-Realm: server1.altpressonline.com = sorry, i don't know how to login to pastebin On Mon, Aug 31, 2009 at 1:42 PM, Michael Collinsm...@freeswitch.org wrote: Could you give a few more details? For example, could you pastebin the output of sofia status profile external? -MC On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer e.schmidba...@gmail.com wrote: I am unable to call a user outside of our local area network. the user is registered on the external profile but there is no way to call the phone. does anyone have any suggestions how to do this? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org
Re: [Freeswitch-users] remote endpoints
ok, please can you provide a tcpdump/wireshark sniff on before and after that linksys. this is something trivial. T. On Tue, Sep 1, 2009 at 6:22 PM, e schmidbauer e.schmidba...@gmail.comwrote: I put tomato on the router and still no success. upnp is enabled, should i disable it? what do you mean by simple NAT? On Tue, Sep 1, 2009 at 2:32 AM, Tihomir Culjagatculj...@gmail.com wrote: oh good, on remote router/dsl modem (whatever doing NAT) never use upnp, never use ALG, just do a simple NAT and it is alway gonna work! T. On Tue, Sep 1, 2009 at 2:06 AM, e schmidbauer e.schmidba...@gmail.com wrote: i cannot reach the remote endpoint. the remote endpoint can reach a locally registered endpoint. any idea why this is happening? the remote endpoint is behind a linksys with upnp enabled. 2009/8/31 João Mesquita jmesqu...@freeswitch.org: Problem is definetly on far end. If you look at the siptrace, you have the following sequence: 1. Asterisk calls in 2. FreeSWITCH replies with a Trying(100) to complete call right away and proceeds to dialplan 3. FreeSWITCH invites (calls) 7 times the final destination that never responds. 4. Asterisk sends a CANCEL message In all that, your final endpoint never responds to any message. Are you sure you can reach it? jmesquita On Mon, Aug 31, 2009 at 5:18 PM, e schmidbauer e.schmidba...@gmail.com wrote: here's the sip trace: http://pastebin.freeswitch.org/10172 2009/8/31 João Mesquita jmesqu...@freeswitch.org: 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel sofia/external/anonym...@anonymous.invalid entering state [terminated][487] The far end seems to be replying with 487 - Request Terminated... Nothing wrong on FS, seems to be a problem with your endpoints. Can you enable a sip trace? jmesquita On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer e.schmidba...@gmail.com wrote: thanks...heres the pastebin: http://pastebin.freeswitch.org/10171 2009/8/31 João Mesquita jmesqu...@freeswitch.org: Check the password dialog. It will tell you what the username/password is. post the logs for a call as well, please. Regards, jmesquita On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer e.schmidba...@gmail.com wrote: = Nameexternal Domain Name N/A DBName sofia_reg_external Pres Hosts DialplanXML Context public Challenge Realm auto_to RTP-IP 192.168.0.125 Ext-RTP-IP 98.118.151.30 SIP-IP 192.168.0.125 Ext-SIP-IP 98.118.151.30 URL sip:mod_so...@192.168.0.125:5080 BIND-URLsip:mod_so...@192.168.0.125:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS c...@48000h,c...@32000h,sp...@32000h@20i,sp...@16000h @20i,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEGfalse PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLEDtrue STUN-AUTO-DISABLE false CALLS-IN17 FAILED-CALLS-IN 11 CALLS-OUT 9 FAILED-CALLS-OUT9 Registrations: = Call-ID:MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI. User: 1...@server1.altpressonline.com Contact:1000 sip:1...@69.204.30.67:16006;rinstance=50cbd6140e1ab991 Agent: X-Lite release 1103k stamp 53621 Status: Registered(UDP)(unknown) EXP(2009-08-31 16:24:11) Host: server1.altpressonline.com IP: 69.204.30.67 Port: 16006 Auth-User: 1000 Auth-Realm: server1.altpressonline.com = sorry, i don't know how to login to pastebin On Mon, Aug 31, 2009 at 1:42 PM, Michael Collinsm...@freeswitch.org wrote: Could you give a few more details? For example, could you pastebin the output of sofia status profile external? -MC On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer e.schmidba...@gmail.com wrote: I am unable to
Re: [Freeswitch-users] mod_opal
Hi Peter, i did it on linux... it was enough to use svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/trunkptlib svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/branches/v3_6opal this is something that works well :) BTW: do you get a correct callingPartyNumber when you place calls through opal/h323? I'm always getting 000 even if i set *effective_caller_id_number to some value*... T. On Tue, Sep 1, 2009 at 8:37 AM, Peter Olsson peter.ols...@visionutveckling.se wrote: Please look into MODOPAL-10 in jira. You need to apply a patch if you’re using latest opal trunk, ro else you need to use the latest stable version of opal. However, I’m not sure how automated this is in the build process in Linux. I’ve only done this on Windows builds lately. /Peter *Från:* freeswitch-users-boun...@lists.freeswitch.org [mailto: freeswitch-users-boun...@lists.freeswitch.org] *För *Tihomir Culjaga *Skickat:* den 1 september 2009 08:09 *Till:* freeswitch-users@lists.freeswitch.org *Ämne:* Re: [Freeswitch-users] mod_opal hhmmm :)) is there any doc following up mod_opal ? I really don't want to waste your time :) T. On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins m...@freeswitch.org wrote: On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga tculj...@gmail.com wrote: hello, i'm trying to build mod_opal and getting this error: making all mod_logfile making all mod_loopback making all mod_native_file making all mod_opal Compiling mod_opal.cpp... quiet_libtool: compile: g++ -g -ggdb -I. -I/home/tculjaga/freeswitch-trunk/src/include -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal -DHAVE_CONFIG_H -c mod_opal.cpp -fPIC -DPIC -o .libs/mod_opal.o In file included from mod_opal.cpp:25: mod_opal.h:151: error: conflicting return type specified for ‘virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall, void*)’ /usr/include/opal/opal/localep.h:267: error: overriding ‘virtual ptlib_virtual_function_changed_or_removed** OpalLocalEndPoint::CreateConnection(OpalCall, void*)’ mod_opal.cpp: In constructor ‘FSConnection::FSConnection(OpalCall, FSEndPoint, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)’: mod_opal.cpp:564: error: no matching function for call to ‘OpalLocalConnection::OpalLocalConnection(OpalCall, FSEndPoint, NULL)’ /usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall, OpalLocalEndPoint, void*, unsigned int, OpalConnection::StringOptions*, char) /usr/include/opal/opal/localep.h:276: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection) /usr/include/opal/opal/patch.h: In member function ‘switch_status_t FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)’: /usr/include/opal/opal/patch.h:272: error: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is private mod_opal.cpp:1277: error: within this context mod_opal.cpp:1277: warning: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is deprecated (declared at /usr/include/opal/opal/patch.h:272) mod_opal.cpp:1277: warning: ignoring return value of function declared with attribute warn_unused_result /usr/include/opal/opal/patch.h: In member function ‘switch_status_t FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)’: /usr/include/opal/opal/patch.h:272: error: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is private mod_opal.cpp:1399: error: within this context mod_opal.cpp:1399: warning: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is deprecated (declared at /usr/include/opal/opal/patch.h:272) mod_opal.cpp:1399: warning: ignoring return value of function declared with attribute warn_unused_result make[5]: *** [mod_opal.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_opal-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +--+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 tculj...@nemesis:~/freeswitch-trunk$ tculj...@nemesis:~/freeswitch-trunk$ tculj...@nemesis:~/freeswitch-trunk$ what ptlib/opal/fs version did you use to build it? I tried with trunk (ptlib, opal, fs)... and as you can see :) Did you run the buildopal.sh script in src/build ? -MC ___ FreeSWITCH-users mailing list
[Freeswitch-users] ANNOUNCEMENT: Friday Public Meetings Are Coming Back!
We are happy to announce http://www.freeswitch.org/node/201 that the Friday public FreeSWITCH meetings are returning, starting this Friday, September 4. Meetings will run from 11am to 5pm CST. The meetings will be held in the FreeSWITCH public conference, also known as the 888 conference. Connection options include: * SIP: 8...@conference.freeswitch.org * IAX: 8...@conference.freeswitch.org * H.323: 8...@conference.freeswitch.org * GoogleTalk: 8...@conference.freeswitch.org * PSTN: 1-213-799-1400 Please join us and be a part of the conversation! We will be discussing agenda items that include programming, documentation, and janitorial projects. We welcome your input. Please bring your questions, suggestions, and ideas. If you have specific ideas for an agenda item that you feel should be discussed then please email myself and Brian West off list. We will post the agenda for each meeting on the FreeSWITCH wiki page. Thanks for helping make FreeSWITCH such a great community! -Michael ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] mod_voicemail email template variables
get latest trunk and try again.
there was a single character out of place that caused the variables to not
be expanded.
On Tue, Sep 1, 2009 at 8:38 AM, Nick Lemberger nick.lember...@lkfd.netwrote:
I tried doing a set right before the application is called to make a
customer variable but it doesn't get transferred to the template this way
either:
---dialplan snip---
action application=set data=test_var=this is a test/
action application=voicemail data=default $${domain} $1/
---end snip---
---voicemail.tpl snip---
Created: ${voicemail_time}
${test_var}
From: ${voicemail_caller_id_name} ${sip_from_user_stripped}
Duration: ${voicemail_message_len}
---end snip---
---resultant email snip---
Created: Tuesday, September 01 2009, 08 30 AM
From: LEMBERGER,NICK
Duration: 00:00:07
---end snip---
Notice I also tried the channel variable ${sip_from_user_stripped} as it
should be available as well, at least according to the 'info' app. Any
ideas?
-Nick
Anthony Minessale anthony.miness...@gmail.com 08/27/09 1:54 PM
you should be able to for instance put
action application=set data=test_var=this is a test/
right before the voicemail app is called
then put
${test_var} in your template.
making sure to issue reloadxml or restart FS
On Thu, Aug 27, 2009 at 1:06 PM, Nick Lemberger nick.lember...@lkfd.net
wrote:
Thanks for the fast reply!
I just tried 10 random variables from
http://wiki.freeswitch.org/wiki/Channel_Variables and I only see the
whitespace where the variable should be. I've only been able to get the
ones that are set in mod_voicemail.c circa line 1600 to work.
-Nick
On 8/27/2009 at 12:44 PM, in message
191c3a030908271044k63973088xeec12c578d02e...@mail.gmail.com, Anthony
Minessale anthony.miness...@gmail.com wrote:
all variables referenced in the template should expand when sending the
email.
On Thu, Aug 27, 2009 at 12:41 PM, Nick Lemberger
nick.lember...@lkfd.netwrote:
Is there a way to use dialplan variables in the email that gets sent
with
the voicemail attachement. I tried using some but nothing seems to
show
up,
I'm guessing it's a different channel or something...
Any ideas?
Thanks,
-Nick
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Re: [Freeswitch-users] Bind_meta_app and second-degree bridge (.. this is a bad title.)
you probably don't want to call bridge from bind meta app, try using the att_xfer app instead it works like bridge but when you call C you can press # to hangup and bridge a to c or press 0 to conference call all 3. On Tue, Sep 1, 2009 at 6:17 AM, Harry Vangberg ha...@vangberg.name wrote: My basic functionality is this: A calls in, is bridged to B (). I use bind_meta_app to let B rebridge A to C (). After having been rebridged to C, C should be able to rebridge A to B *again*, and so on. This is the code I have: context name=public extension name=ff-ivr condition field=destination_number expression=^(.*)$ action application=set data=bypass_media=false/ action application=answer/ action application=bind_meta_app data=1 b a bridge::sofia/gateway/gw1// action application=bind_meta_app data=2 b a bridge::sofia/gateway/gw1// action application=bridge data=sofia/gateway/gw1/ / /condition /extension /context The first bridge is fine, and B can press *2 to bridge to C/. But if C presses *1, it seems to execute the bridge app, but nothing at all happens: 2009-09-01 11:14:59.258325 [DEBUG] switch_rtp.c: RTP RECV DTMF *:2000 2009-09-01 11:15:00.118195 [DEBUG] switch_rtp.c: RTP RECV DTMF 1:2000 2009-09-01 11:15:00.118195 [DEBUG] switch_ivr_async.c:1725 sofia/external/unkn...@129.142.224.250 Processing meta digit '2' [bridge::sofia/gateway/gw1/] 2009-09-01 11:15:00.118195 [DEBUG] switch_core_session.c:813 Send signal sofia/external/unkn...@129.142.224.250 [BREAK] Any ideas? ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.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_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@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] conference question
hello
we have got a little problem with the conference application
in our setup we have da system for customers where speakers can dial in
with phonenumber+1 and the listeners dial in with phonenumber
the speakers conference is started with 323963...@conf+flags{waste}
the listeners conference is started with 323963...@conf+flags{mute,waste}
waste is needed to get the whole audio stream
it now happens that listeners sometimes hear each other, that shouldn't be
what can i do to resolve this problem?
we are using version 1.0.4
br
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Technische Leitung, Forschung Entwicklung VoIP
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Re: [Freeswitch-users] conference question
I haven't really used waste much myself, but my understanding is that
waste and mute would conflict, since waste says send audio always
and mute says send audio never. I didn't understand why you're using
waste on the listeners... you should be able to get by with waste just
on the speaker (again, that's how I understand it).
2009/9/1 Christian Löschenkohl christian.loeschenk...@xpirio.com:
hello
we have got a little problem with the conference application
in our setup we have da system for customers where speakers can dial in
with phonenumber+1 and the listeners dial in with phonenumber
the speakers conference is started with 323963...@conf+flags{waste}
the listeners conference is started with 323963...@conf+flags{mute,waste}
waste is needed to get the whole audio stream
it now happens that listeners sometimes hear each other, that shouldn't be
what can i do to resolve this problem?
we are using version 1.0.4
br
--
Ing. Christian Löschenkohl
Technische Leitung, Forschung Entwicklung VoIP
xpirio
Telekommunikation Service GmbH
Lakeside B04
9020 Klagenfurt
Austria
T +43 (0) 5 77 11 - 1000
F +43 (0) 5 77 11 - 1002
E christian.loeschenk...@xpirio.com
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Re: [Freeswitch-users] conference question
thank you for your response
as a listener waste influences what you hear and mute say's you cannot speak
this is what our customer wanted because the speaker is the only one who is
heard in this conference or meeting room - this rooms are for lectures
we tried to disable waste for the listeners (and let it on for the speaker) but
this
resulted in choppy sound for the listeners (silence periods between words and
sentences)
i hope i could explain my problem a little bit better
br
On 2009-09-01 23:32, Bradley Brashier wrote:
I haven't really used waste much myself, but my understanding is that
waste and mute would conflict, since waste says send audio always
and mute says send audio never. I didn't understand why you're using
waste on the listeners... you should be able to get by with waste just
on the speaker (again, that's how I understand it).
2009/9/1 Christian Löschenkohlchristian.loeschenk...@xpirio.com:
hello
we have got a little problem with the conference application
in our setup we have da system for customers where speakers can dial in
with phonenumber+1 and the listeners dial in with phonenumber
the speakers conference is started with 323963...@conf+flags{waste}
the listeners conference is started with 323963...@conf+flags{mute,waste}
waste is needed to get the whole audio stream
it now happens that listeners sometimes hear each other, that shouldn't be
what can i do to resolve this problem?
we are using version 1.0.4
br
--
Ing. Christian Löschenkohl
Technische Leitung, Forschung Entwicklung VoIP
xpirio
Telekommunikation Service GmbH
Lakeside B04
9020 Klagenfurt
Austria
T +43 (0) 5 77 11 - 1000
F +43 (0) 5 77 11 - 1002
E christian.loeschenk...@xpirio.com
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Technische Leitung, Forschung Entwicklung VoIP
xpirio
Telekommunikation Service GmbH
Lakeside B04
9020 Klagenfurt
Austria
T +43 (0) 5 77 11 - 1000
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E christian.loeschenk...@xpirio.com
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Re: [Freeswitch-users] conference question
waste + mute would result in sending audio that was all zeros or generated
silence.
On Tue, Sep 1, 2009 at 4:32 PM, Bradley Brashier bjbrash...@gmail.comwrote:
I haven't really used waste much myself, but my understanding is that
waste and mute would conflict, since waste says send audio always
and mute says send audio never. I didn't understand why you're using
waste on the listeners... you should be able to get by with waste just
on the speaker (again, that's how I understand it).
2009/9/1 Christian Löschenkohl christian.loeschenk...@xpirio.com:
hello
we have got a little problem with the conference application
in our setup we have da system for customers where speakers can dial in
with phonenumber+1 and the listeners dial in with phonenumber
the speakers conference is started with 323963...@conf+flags{waste}
the listeners conference is started with 323963...@conf
+flags{mute,waste}
waste is needed to get the whole audio stream
it now happens that listeners sometimes hear each other, that shouldn't
be
what can i do to resolve this problem?
we are using version 1.0.4
br
--
Ing. Christian Löschenkohl
Technische Leitung, Forschung Entwicklung VoIP
xpirio
Telekommunikation Service GmbH
Lakeside B04
9020 Klagenfurt
Austria
T +43 (0) 5 77 11 - 1000
F +43 (0) 5 77 11 - 1002
E christian.loeschenk...@xpirio.com
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Re: [Freeswitch-users] problem compiling esl for use with freepbx v3
Michael Yes, from memory the intial make completes successfully, when you go to make the modules themselves is when it starts barfing. On 31 Aug 2009 at 23:33, Michael Collins wrote: Did the simple make in the libs/esl directory run properly? Just curious. I'll have to defer to the Ubuntu gurus out there for thoughts on what else could be wrong. -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. h...@pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] problem compiling esl for use with freepbx v3
All of the language modules in ESL require the runtime and the devel packages for that language for the compile to work. On Tue, Sep 1, 2009 at 5:33 PM, Harondel J. Sibble h...@pdscc.com wrote: Michael Yes, from memory the intial make completes successfully, when you go to make the modules themselves is when it starts barfing. On 31 Aug 2009 at 23:33, Michael Collins wrote: Did the simple make in the libs/esl directory run properly? Just curious. I'll have to defer to the Ubuntu gurus out there for thoughts on what else could be wrong. -- Harondel J. Sibble Sibble Computer Consulting Creating Solutions for the small and medium business computer user. h...@pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.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_miness...@hotmail.com msn%3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.compaypal%3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip%3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3aconf%2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Re: [Freeswitch-users] conference question
that means something in your path does not support CNG/VAD.
it's perfectly ok to use waste and mute together.
there is no chance that you would not enter the conf muted the way you
describe unless you are
using an older revision of FS that had a bug in the parsing of the
conference flags.
2009/9/1 Christian Löschenkohl christian.loeschenk...@xpirio.com
thank you for your response
as a listener waste influences what you hear and mute say's you cannot
speak
this is what our customer wanted because the speaker is the only one who is
heard in this conference or meeting room - this rooms are for lectures
we tried to disable waste for the listeners (and let it on for the speaker)
but this
resulted in choppy sound for the listeners (silence periods between words
and sentences)
i hope i could explain my problem a little bit better
br
On 2009-09-01 23:32, Bradley Brashier wrote:
I haven't really used waste much myself, but my understanding is that
waste and mute would conflict, since waste says send audio always
and mute says send audio never. I didn't understand why you're using
waste on the listeners... you should be able to get by with waste just
on the speaker (again, that's how I understand it).
2009/9/1 Christian Löschenkohlchristian.loeschenk...@xpirio.com:
hello
we have got a little problem with the conference application
in our setup we have da system for customers where speakers can dial in
with phonenumber+1 and the listeners dial in with phonenumber
the speakers conference is started with 323963...@conf+flags{waste}
the listeners conference is started with 323963...@conf
+flags{mute,waste}
waste is needed to get the whole audio stream
it now happens that listeners sometimes hear each other, that shouldn't
be
what can i do to resolve this problem?
we are using version 1.0.4
br
--
Ing. Christian Löschenkohl
Technische Leitung, Forschung Entwicklung VoIP
xpirio
Telekommunikation Service GmbH
Lakeside B04
9020 Klagenfurt
Austria
T +43 (0) 5 77 11 - 1000
F +43 (0) 5 77 11 - 1002
E christian.loeschenk...@xpirio.com
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xpirio
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Austria
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Re: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
I run into this problem before. Don't remember the exact error but might be segfault of lame runing in freeswitch-lua. If you use Linux you would like to try iwatch. It's a perl program watching your file system and can execute the lame command as soon as it got the CLOSE_WRITE(or other) filesystem event. On Aug 26, 2009, at 4:44 PM, Michael Jerris wrote: Running out of stack space? The stack space we run freeswitch in is fairly small. Programs launched from the freeswitch process inherit this. Mike On Aug 26, 2009, at 4:28 AM, Alberto Escudero-Pascual (lists) wrote: I ran strace from freeswitch and from the command line. lame segfaults when run from system FS. The only obvious different i see is in the execve() /* XX vars */ apart from the final Segfault From execve(/usr/local/freeswitch/bin/lame, [/usr/local/freeswitch/bin/lame, /tmp/foo.wav, /tmp/foo.mp3, - S], [/* 16 vars */]) = 0 From FS execve(/usr/local/freeswitch/bin/lame, [/usr/local/freeswitch/bin/lame, /tmp/foo.wav, /tmp/fooo.mp3, -S], [/* 14 vars */]) = 0 I am attaching the full straces in case they are of any help. Not sure if this deserves a jira /aep -- Stopping junk mailers is good for the environment maybe it's writing some err to stderr that is being suppressed somehow On Tue, Aug 25, 2009 at 3:46 PM, Alberto Escudero-Pascual (lists) aep.li...@it46.se wrote: Hi Brian, From the CLI freeswi...@open46 system /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S 2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing command: /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S)] output: +OK open46:/tmp# ls foo.wav and running the command from the command line: open46:/tmp# /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -Sopen46:/tmp# ls foo.mp3 foo.wav If I do the same with lame397 freeswi...@open46 system /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S 2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing command: /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S)] output: +OK open46:/tmp# ls foo.mp3 foo.wav Highly paranormal! Sorry for hijacking the previous thread. /aep -- Stopping junk mailers is good for the environment Try running it at the CLI and see if you see any errors. Also please do not hijack threads. The original thread [Freeswitch-users] XML- RPC on different ip than 0.0.0.0 which was hijacked by clicking reply, changing the subject and clicking send. Please in the future do not do that as it clutters up the threading and could get your query lost in the noise. Thanks, Brian On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote: Here it comes the mystery. I am use lame 3.98.2 the mp3 file never appears, if I use version 3.97 (older version), it does!. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.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_miness...@hotmail.com MSN %3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.comPAYPAL %3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip %3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3Aconf %2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org lame_strace.txt___ FreeSWITCH-users mailing list FreeSWITCH-users@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
Re: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
Hi Steven, Sounds like a very good tip. Do you have any example available to share? I will be happy to upload it to the wiki when i put it up and running. /aep -- Stopping junk mailers is good for the environment I run into this problem before. Don't remember the exact error but might be segfault of lame runing in freeswitch-lua. If you use Linux you would like to try iwatch. It's a perl program watching your file system and can execute the lame command as soon as it got the CLOSE_WRITE(or other) filesystem event. On Aug 26, 2009, at 4:44 PM, Michael Jerris wrote: Running out of stack space? The stack space we run freeswitch in is fairly small. Programs launched from the freeswitch process inherit this. Mike On Aug 26, 2009, at 4:28 AM, Alberto Escudero-Pascual (lists) wrote: I ran strace from freeswitch and from the command line. lame segfaults when run from system FS. The only obvious different i see is in the execve() /* XX vars */ apart from the final Segfault From execve(/usr/local/freeswitch/bin/lame, [/usr/local/freeswitch/bin/lame, /tmp/foo.wav, /tmp/foo.mp3, - S], [/* 16 vars */]) = 0 From FS execve(/usr/local/freeswitch/bin/lame, [/usr/local/freeswitch/bin/lame, /tmp/foo.wav, /tmp/fooo.mp3, -S], [/* 14 vars */]) = 0 I am attaching the full straces in case they are of any help. Not sure if this deserves a jira /aep -- Stopping junk mailers is good for the environment maybe it's writing some err to stderr that is being suppressed somehow On Tue, Aug 25, 2009 at 3:46 PM, Alberto Escudero-Pascual (lists) aep.li...@it46.se wrote: Hi Brian, From the CLI freeswi...@open46 system /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S 2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing command: /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S)] output: +OK open46:/tmp# ls foo.wav and running the command from the command line: open46:/tmp# /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -Sopen46:/tmp# ls foo.mp3 foo.wav If I do the same with lame397 freeswi...@open46 system /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S 2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing command: /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S)] output: +OK open46:/tmp# ls foo.mp3 foo.wav Highly paranormal! Sorry for hijacking the previous thread. /aep -- Stopping junk mailers is good for the environment Try running it at the CLI and see if you see any errors. Also please do not hijack threads. The original thread [Freeswitch-users] XML- RPC on different ip than 0.0.0.0 which was hijacked by clicking reply, changing the subject and clicking send. Please in the future do not do that as it clutters up the threading and could get your query lost in the noise. Thanks, Brian On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote: Here it comes the mystery. I am use lame 3.98.2 the mp3 file never appears, if I use version 3.97 (older version), it does!. ___ FreeSWITCH-users mailing list FreeSWITCH-users@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@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.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_miness...@hotmail.com MSN %3aanthony_miness...@hotmail.com GTALK/JABBER/PAYPAL:anthony.miness...@gmail.comPAYPAL %3aanthony.miness...@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:8...@conference.freeswitch.org sip %3a...@conference.freeswitch.org iax:gu...@conference.freeswitch.org/888 googletalk:conf+...@conference.freeswitch.orggoogletalk%3Aconf %2b...@conference.freeswitch.org pstn:213-799-1400 ___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- users http://www.freeswitch.org lame_strace.txt___ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Re: [Freeswitch-users] mod_opal
Tihomir, Yes as I remember it I did get the correct caller id number. I think you need to set variable origination_caller_id_number when you originate a call. /Peter Från: Tihomir Culjaga tculj...@gmail.com Skickat: den 1 september 2009 19:20 Till: freeswitch-users@lists.freeswitch.org freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] mod_opal Hi Peter, i did it on linux... it was enough to use svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/ptlib/trunk ptlib svn co https://opalvoip.svn.sourceforge.net/svnroot/opalvoip/opal/branches/v3_6 opal this is something that works well :) BTW: do you get a correct callingPartyNumber when you place calls through opal/h323? I'm always getting 000 even if i set effective_caller_id_number to some value... T. On Tue, Sep 1, 2009 at 8:37 AM, Peter Olsson peter.ols...@visionutveckling.semailto:peter.ols...@visionutveckling.se wrote: Please look into MODOPAL-10 in jira. You need to apply a patch if you’re using latest opal trunk, ro else you need to use the latest stable version of opal. However, I’m not sure how automated this is in the build process in Linux. I’ve only done this on Windows builds lately. /Peter Från: freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org [mailto:freeswitch-users-boun...@lists.freeswitch.orgmailto:freeswitch-users-boun...@lists.freeswitch.org] För Tihomir Culjaga Skickat: den 1 september 2009 08:09 Till: freeswitch-users@lists.freeswitch.orgmailto:freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] mod_opal hhmmm :)) is there any doc following up mod_opal ? I really don't want to waste your time :) T. On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins m...@freeswitch.orgmailto:m...@freeswitch.org wrote: On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga tculj...@gmail.commailto:tculj...@gmail.com wrote: hello, i'm trying to build mod_opal and getting this error: making all mod_logfile making all mod_loopback making all mod_native_file making all mod_opal Compiling mod_opal.cpp... quiet_libtool: compile: g++ -g -ggdb -I. -I/home/tculjaga/freeswitch-trunk/src/include -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal -DHAVE_CONFIG_H -c mod_opal.cpp -fPIC -DPIC -o .libs/mod_opal.o In file included from mod_opal.cpp:25: mod_opal.h:151: error: conflicting return type specified for ‘virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall, void*)’ /usr/include/opal/opal/localep.h:267: error: overriding ‘virtual ptlib_virtual_function_changed_or_removed** OpalLocalEndPoint::CreateConnection(OpalCall, void*)’ mod_opal.cpp: In constructor ‘FSConnection::FSConnection(OpalCall, FSEndPoint, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)’: mod_opal.cpp:564: error: no matching function for call to ‘OpalLocalConnection::OpalLocalConnection(OpalCall, FSEndPoint, NULL)’ /usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall, OpalLocalEndPoint, void*, unsigned int, OpalConnection::StringOptions*, char) /usr/include/opal/opal/localep.h:276: note: OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection) /usr/include/opal/opal/patch.h: In member function ‘switch_status_t FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)’: /usr/include/opal/opal/patch.h:272: error: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is private mod_opal.cpp:1277: error: within this context mod_opal.cpp:1277: warning: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is deprecated (declared at /usr/include/opal/opal/patch.h:272) mod_opal.cpp:1277: warning: ignoring return value of function declared with attribute warn_unused_result /usr/include/opal/opal/patch.h: In member function ‘switch_status_t FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)’: /usr/include/opal/opal/patch.h:272: error: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is private mod_opal.cpp:1399: error: within this context mod_opal.cpp:1399: warning: ‘virtual ptlib_virtual_function_changed_or_removed** OpalMediaPatch::OnPatchStart()’ is deprecated (declared at /usr/include/opal/opal/patch.h:272) mod_opal.cpp:1399: warning: ignoring return value of function declared with attribute warn_unused_result make[5]: *** [mod_opal.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_opal-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build + FreeSWITCH Build Complete ---+ + FreeSWITCH has been successfully built. + + Install by running:
