From kadantsev.d at gmail.com Tue Sep 1 01:13:30 2009 From: kadantsev.d at gmail.com (Dmitry Kadantsev) Date: Tue, 1 Sep 2009 10:13:30 +0200 Subject: [Freeswitch-users] FS performance under windows Message-ID: <681a20520909010113o3556535er7415dbb1961f6ea4@mail.gmail.com> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/1d5d8b40/attachment.html From shaheryarkh at googlemail.com Tue Sep 1 02:00:19 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Tue, 1 Sep 2009 15:00:19 +0600 Subject: [Freeswitch-users] FS performance under windows In-Reply-To: <681a20520909010113o3556535er7415dbb1961f6ea4@mail.gmail.com> References: <681a20520909010113o3556535er7415dbb1961f6ea4@mail.gmail.com> Message-ID: 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 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 at 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_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/bdcb4e41/attachment.html From enno.egbert at googlemail.com Tue Sep 1 02:40:51 2009 From: enno.egbert at googlemail.com (NOx-WHV) Date: Tue, 1 Sep 2009 02:40:51 -0700 (PDT) Subject: [Freeswitch-users] SRTP Encryption Message-ID: <25237296.post@talk.nabble.com> 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. 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. From Prometheus001 at gmx.net Tue Sep 1 03:16:01 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 01 Sep 2009 12:16:01 +0200 Subject: [Freeswitch-users] sofia_reg_external in odbc? In-Reply-To: References: <4A9AFC3D.9060208@gmx.net> Message-ID: <4A9CF461.6070407@gmx.net> 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: > > > 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From Prometheus001 at gmx.net Tue Sep 1 03:19:31 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 01 Sep 2009 12:19:31 +0200 Subject: [Freeswitch-users] SRTP Encryption In-Reply-To: <25237296.post@talk.nabble.com> References: <25237296.post@talk.nabble.com> Message-ID: <4A9CF533.10601@gmx.net> 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. > > > > 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 > From nicolas at medularis.com Tue Sep 1 03:45:44 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 1 Sep 2009 06:45:44 -0400 Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3 In-Reply-To: <87f2f3b90908312333s5bf47d6et94ec24fbc51d6d24@mail.gmail.com> References: <20090823185152.D17845FE@sinclaire.sibble.net> <20090901054337.7317F65B@sinclaire.sibble.net> <87f2f3b90908312333s5bf47d6et94ec24fbc51d6d24@mail.gmail.com> Message-ID: <1b46b4e80909010345r7c3928b1l93f1411dc2853533@mail.gmail.com> 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 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 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:1111: 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 _wrap_ESLevent_delHeader(int, zval*, >> zval**, >> > zval*, int)': >> > esl_wrap.cpp:1644: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLevent_firstHeader(int, zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:1674: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLevent_nextHeader(int, zval*, >> > zval**, >> > zval*, int)': >> > esl_wrap.cpp:1704: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_0(int, >> zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:1744: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_1(int, >> zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:1770: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection(int, zval*, >> zval**, >> > zval*, int)': >> > esl_wrap.cpp:1803: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void >> _wrap_ESLconnection_socketDescriptor(int, >> > zval*, zval**, zval*, int)': >> > esl_wrap.cpp:1846: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_connected(int, >> zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:1872: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_getInfo(int, zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:1898: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_send(int, zval*, >> zval**, >> > zval*, int)': >> > esl_wrap.cpp:1931: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendRecv(int, zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:1964: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_api(int, zval*, >> zval**, >> > zval*, int)': >> > esl_wrap.cpp:2007: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_bgapi(int, zval*, >> > zval**, >> > zval*, int)': >> > esl_wrap.cpp:2050: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendEvent(int, >> zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:2082: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEvent(int, >> zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:2108: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEventTimed(int, >> > zval*, zval**, zval*, int)': >> > esl_wrap.cpp:2141: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_filter(int, zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:2181: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_events(int, zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:2221: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_execute(int, zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:2272: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_executeAsync(int, >> zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:2323: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_setAsyncExecute(int, >> > zval*, zval**, zval*, int)': >> > esl_wrap.cpp:2356: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_setEventLock(int, >> zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:2389: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_disconnect(int, >> zval*, >> > zval**, zval*, int)': >> > esl_wrap.cpp:2415: error: format not a string literal and no format >> > arguments >> > esl_wrap.cpp: In function 'void _wrap_eslSetLogLevel(int, zval*, zval**, >> > zval*, int)': >> > esl_wrap.cpp:2438: error: format not a string literal and no format >> > arguments >> > make[1]: *** [esl_wrap.o] Error 1 >> > make[1]: Leaving directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php' >> > make: *** [phpmod] Error 2 >> > >> > Same thing happens if I try sudo make everymod >> > >> > Checking the list archives I found this thread >> > >> > http://www.nabble.com/ESL-Wrapper-td22209991.html#a22222338 >> > >> > I've made sure that the php-dev packages are installed. Any suggestions >> on >> > what to do next? >> > -- >> >> -- >> Harondel J. Sibble >> Sibble Computer Consulting >> Creating Solutions for the small and medium business computer user. >> help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com >> (604) 739-3709 (voice) >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/7a2170a0/attachment-0001.html From enno.egbert at googlemail.com Tue Sep 1 03:48:17 2009 From: enno.egbert at googlemail.com (NOx-WHV) Date: Tue, 1 Sep 2009 03:48:17 -0700 (PDT) Subject: [Freeswitch-users] SRTP Encryption In-Reply-To: <4A9CF533.10601@gmx.net> References: <25237296.post@talk.nabble.com> <4A9CF533.10601@gmx.net> Message-ID: <25238144.post@talk.nabble.com> 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. >> >> >> >> 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 at 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. From harry at vangberg.name Tue Sep 1 04:17:32 2009 From: harry at vangberg.name (Harry Vangberg) Date: Tue, 1 Sep 2009 13:17:32 +0200 Subject: [Freeswitch-users] Bind_meta_app and second-degree bridge (.. this is a bad title.) Message-ID: <74d41a3d0909010417m5654adebia4be53281574864@mail.gmail.com> My basic functionality is this: A calls in, is bridged to B (1111). I use bind_meta_app to let B rebridge A to C (2222). 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: The first bridge is fine, and B can press *2 to bridge to C/2222. 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:2222 RTP RECV DTMF *:2000 2009-09-01 11:15:00.118195 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF 1:2000 2009-09-01 11:15:00.118195 [DEBUG] switch_ivr_async.c:1725 sofia/external/unknown at 129.142.224.250 Processing meta digit '2' [bridge::sofia/gateway/gw1/1111] 2009-09-01 11:15:00.118195 [DEBUG] switch_core_session.c:813 Send signal sofia/external/unknown at 129.142.224.250 [BREAK] Any ideas? From kadantsev.d at gmail.com Tue Sep 1 04:20:31 2009 From: kadantsev.d at gmail.com (Dmitry Kadantsev) Date: Tue, 1 Sep 2009 13:20:31 +0200 Subject: [Freeswitch-users] FS performance under windows In-Reply-To: References: <681a20520909010113o3556535er7415dbb1961f6ea4@mail.gmail.com> Message-ID: <681a20520909010420k7f5e39ach67dcd08682113cb2@mail.gmail.com> 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 < shaheryarkh at 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 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 at 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_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/1edd3ca6/attachment.html From mgg at giagnocavo.net Tue Sep 1 04:20:31 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 1 Sep 2009 07:20:31 -0400 Subject: [Freeswitch-users] FS performance under windows In-Reply-To: References: <681a20520909010113o3556535er7415dbb1961f6ea4@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4109@mse17be1.mse17.exchange.ms> Do you have any specific notes why production or load testing isn?t recommended on Windows? Or just lack of data? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad Shahzad Sent: Tuesday, September 01, 2009 3:00 AM To: freeswitch-users at 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 > 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 at 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_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/291a2eb9/attachment-0001.html From Prometheus001 at gmx.net Tue Sep 1 04:21:45 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Tue, 01 Sep 2009 13:21:45 +0200 Subject: [Freeswitch-users] SRTP Encryption In-Reply-To: <25238144.post@talk.nabble.com> References: <25237296.post@talk.nabble.com> <4A9CF533.10601@gmx.net> <25238144.post@talk.nabble.com> Message-ID: <4A9D03C9.60509@gmx.net> 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. >>> >>> >>> >>> 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > From Nick.Lemberger at lkfd.net Tue Sep 1 06:38:44 2009 From: Nick.Lemberger at lkfd.net (Nick Lemberger) Date: Tue, 01 Sep 2009 08:38:44 -0500 Subject: [Freeswitch-users] mod_voicemail email template variables Message-ID: <4A9CDD94020000FE0000A15D@application-tr-fa-1.lakefield.telco> 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--- ---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 08/27/09 1:54 PM >>> you should be able to for instance put 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 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 > <191c3a030908271044k63973088xeec12c578d02ef14 at mail.gmail.com>, Anthony > Minessale wrote: > > all variables referenced in the template should expand when sending the > > email. > > > > > > On Thu, Aug 27, 2009 at 12:41 PM, Nick Lemberger > > wrote: > > > >> 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 > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 From brian at freeswitch.org Tue Sep 1 06:46:13 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 1 Sep 2009 08:46:13 -0500 Subject: [Freeswitch-users] SRTP Encryption In-Reply-To: <25237296.post@talk.nabble.com> References: <25237296.post@talk.nabble.com> Message-ID: <78E6640D-64BB-4453-994B-43928065C01C@freeswitch.org> Try this one. Outbound Inbound /b On Sep 1, 2009, at 4:40 AM, NOx-WHV wrote: > From brian at freeswitch.org Tue Sep 1 06:46:52 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 1 Sep 2009 08:46:52 -0500 Subject: [Freeswitch-users] sofia_reg_external in odbc? In-Reply-To: <4A9CF461.6070407@gmx.net> References: <4A9AFC3D.9060208@gmx.net> <4A9CF461.6070407@gmx.net> Message-ID: 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 From gregt at cgicommunications.com Tue Sep 1 06:54:22 2009 From: gregt at cgicommunications.com (Greg Thoen) Date: Tue, 1 Sep 2009 09:54:22 -0400 Subject: [Freeswitch-users] Incorrect method of PHP call control? In-Reply-To: <87f2f3b90908310955u399f5c5v1214d58e6004e392@mail.gmail.com> References: <4E10B925-CEFD-4AF1-8527-282DCDEC0603@cgicommunications.com> <87f2f3b90908310955u399f5c5v1214d58e6004e392@mail.gmail.com> Message-ID: <40C6E7F6-D49A-4902-B3F8-224567325875@cgicommunications.com> 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 > 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; > > 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/ba280bc4/attachment.html From math.parent at gmail.com Tue Sep 1 04:06:46 2009 From: math.parent at gmail.com (Mathieu Parent) Date: Tue, 1 Sep 2009 13:06:46 +0200 Subject: [Freeswitch-users] mod_fax not working In-Reply-To: <4A980D58.5020808@coppice.org> References: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com> <4A980D58.5020808@coppice.org> Message-ID: <960738410909010406x4d938d0cx26c79e0fc83b8c1f@mail.gmail.com> Hi, On Fri, Aug 28, 2009 at 7:01 PM, Steve Underwood wrote: (snip) >> > The log shows the same thing happening every time. A bad CRC from the > far end, followed by a good DCS frame followed by what seems to be > rubbish. I think I'd need an audio log from one of these calls to figure > out any more. > I have attached a pcap file only with SIP and RTP. Thanks Mathieu -------------- next part -------------- A non-text attachment was scrubbed... Name: dump-filtered.pcap Type: application/octet-stream Size: 1501716 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/94133deb/attachment-0001.obj From jlenk at frontiernet.net Tue Sep 1 07:24:51 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Tue, 1 Sep 2009 09:24:51 -0500 (CDT) Subject: [Freeswitch-users] FS performance under windows In-Reply-To: <681a20520909010113o3556535er7415dbb1961f6ea4@mail.gmail.com> References: <681a20520909010113o3556535er7415dbb1961f6ea4@mail.gmail.com> Message-ID: <1251815091853-3560840.post@n2.nabble.com> 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 at 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. From dome at tel.co.th Tue Sep 1 07:37:33 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Tue, 1 Sep 2009 21:37:33 +0700 Subject: [Freeswitch-users] Mod_fifo posision in queue Message-ID: <8ccbff060909010737v4a8f17ep4d6175c2a2a2f866@mail.gmail.com> 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. From e.schmidbauer at gmail.com Tue Sep 1 09:22:01 2009 From: e.schmidbauer at gmail.com (e schmidbauer) Date: Tue, 1 Sep 2009 12:22:01 -0400 Subject: [Freeswitch-users] remote endpoints In-Reply-To: <65d96fc80908312332h48c92eden4cdd28ae2f66e4f7@mail.gmail.com> References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com> <87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com> <2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com> <2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com> <2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com> <2cef777b0908311706o4173312bw9af8b217aca47392@mail.gmail.com> <65d96fc80908312332h48c92eden4cdd28ae2f66e4f7@mail.gmail.com> Message-ID: <2cef777b0909010922h36119c87u58273cc60ba3d815@mail.gmail.com> 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 Culjaga 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 > 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 Jo?o Mesquita : >> > 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 >> > wrote: >> >> >> >> here's the sip trace: >> >> http://pastebin.freeswitch.org/10172 >> >> >> >> >> >> >> >> 2009/8/31 Jo?o Mesquita : >> >> > 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel >> >> > sofia/external/Anonymous at 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 >> >> > >> >> > wrote: >> >> >> >> >> >> thanks...heres the pastebin: >> >> >> http://pastebin.freeswitch.org/10171 >> >> >> >> >> >> 2009/8/31 Jo?o Mesquita : >> >> >> > 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 >> >> >> > >> >> >> > 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_sofia at 192.168.0.125:5080 >> >> >> >> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.0.125:5080 >> >> >> >> HOLD-MUSIC ? ? ? ? ? ? ?local_stream://moh >> >> >> >> OUTBOUND-PROXY ? ? ? ? ?N/A >> >> >> >> CODECS >> >> >> >> >> >> >> >> >> >> >> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 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: ? ? ? ? ? 1000 at server1.altpressonline.com >> >> >> >> Contact: ? ? ? ?"1000" >> >> >> >> >> >> >> >> 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 >> >> >> >> 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 >> >> >> >> > >> >> >> >> > 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From tculjaga at gmail.com Tue Sep 1 09:56:49 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 1 Sep 2009 18:56:49 +0200 Subject: [Freeswitch-users] remote endpoints In-Reply-To: <2cef777b0909010922h36119c87u58273cc60ba3d815@mail.gmail.com> References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com> <2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com> <2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com> <2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com> <2cef777b0908311706o4173312bw9af8b217aca47392@mail.gmail.com> <65d96fc80908312332h48c92eden4cdd28ae2f66e4f7@mail.gmail.com> <2cef777b0909010922h36119c87u58273cc60ba3d815@mail.gmail.com> Message-ID: <65d96fc80909010956v2557819fs73d8ff714f5521b7@mail.gmail.com> 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 wrote: > 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 Culjaga 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 > > 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 Jo?o Mesquita : > >> > 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.schmidbauer at gmail.com> > >> > wrote: > >> >> > >> >> here's the sip trace: > >> >> http://pastebin.freeswitch.org/10172 > >> >> > >> >> > >> >> > >> >> 2009/8/31 Jo?o Mesquita : > >> >> > 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel > >> >> > sofia/external/Anonymous at 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 > >> >> > > >> >> > wrote: > >> >> >> > >> >> >> thanks...heres the pastebin: > >> >> >> http://pastebin.freeswitch.org/10171 > >> >> >> > >> >> >> 2009/8/31 Jo?o Mesquita : > >> >> >> > 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 > >> >> >> > > >> >> >> > 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_sofia at 192.168.0.125:5080 > >> >> >> >> BIND-URL sip:mod_sofia at 192.168.0.125:5080 > >> >> >> >> HOLD-MUSIC local_stream://moh > >> >> >> >> OUTBOUND-PROXY N/A > >> >> >> >> CODECS > >> >> >> >> > >> >> >> >> > >> >> >> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 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: 1000 at server1.altpressonline.com > >> >> >> >> Contact: "1000" > >> >> >> >> > >> >> >> >> 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 > >> >> >> >> 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 > >> >> >> >> > > >> >> >> >> > 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/69c3b1c3/attachment-0001.html From tculjaga at gmail.com Tue Sep 1 10:12:45 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 1 Sep 2009 19:12:45 +0200 Subject: [Freeswitch-users] mod_opal In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C52ED0F4743@cooper> References: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com> <87f2f3b90908311600h43468c1ao70719aa64479d226@mail.gmail.com> <65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com> <549CFEF87AEDE841A38E9D15EAB4C04C52ED0F4743@cooper> Message-ID: <65d96fc80909011012w4308046djd7fc027ac8027724@mail.gmail.com> 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 0000000 even if i set *effective_caller_id_number to some value*... T. On Tue, Sep 1, 2009 at 8:37 AM, Peter Olsson < peter.olsson at 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 > > > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *Tihomir Culjaga > *Skickat:* den 1 september 2009 08:09 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* 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 > wrote: > > > > On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga > 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 > tculjaga at nemesis:~/freeswitch-trunk$ > tculjaga at nemesis:~/freeswitch-trunk$ > tculjaga at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/0b8f935b/attachment.html From msc at freeswitch.org Tue Sep 1 10:23:07 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 1 Sep 2009 10:23:07 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: Friday Public Meetings Are Coming Back! Message-ID: <87f2f3b90909011023pfaac01dw472ec8df11311de@mail.gmail.com> We are happy to announce 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: 888 at conference.freeswitch.org * IAX: 888 at conference.freeswitch.org * H.323: 888 at conference.freeswitch.org * GoogleTalk: 888 at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/2ecac354/attachment.html From anthony.minessale at gmail.com Tue Sep 1 10:41:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Sep 2009 12:41:48 -0500 Subject: [Freeswitch-users] mod_voicemail email template variables In-Reply-To: <4A9CDD94020000FE0000A15D@application-tr-fa-1.lakefield.telco> References: <4A9CDD94020000FE0000A15D@application-tr-fa-1.lakefield.telco> Message-ID: <191c3a030909011041o51432e7bmec28c01db4aa74b5@mail.gmail.com> 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 wrote: > 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--- > > > ---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 08/27/09 1:54 PM >>> > you should be able to for instance put > > > > 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 >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 > > <191c3a030908271044k63973088xeec12c578d02ef14 at mail.gmail.com>, Anthony > > Minessale wrote: > > > all variables referenced in the template should expand when sending the > > > email. > > > > > > > > > On Thu, Aug 27, 2009 at 12:41 PM, Nick Lemberger > > > wrote: > > > > > >> 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 > > >> > > >> > > >> _______________________________________________ > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at 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 at 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_minessale at hotmail.com < > MSN%3Aanthony_minessale at hotmail.com > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org < > sip%3A888 at conference.freeswitch.org > > > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > > > > pstn:213-799-1400 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/48be755c/attachment-0001.html From anthony.minessale at gmail.com Tue Sep 1 11:02:11 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Sep 2009 13:02:11 -0500 Subject: [Freeswitch-users] Bind_meta_app and second-degree bridge (.. this is a bad title.) In-Reply-To: <74d41a3d0909010417m5654adebia4be53281574864@mail.gmail.com> References: <74d41a3d0909010417m5654adebia4be53281574864@mail.gmail.com> Message-ID: <191c3a030909011102k38ba525dkb1c89fd8ea84f23@mail.gmail.com> 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 wrote: > My basic functionality is this: A calls in, is bridged to B (1111). I > use bind_meta_app to let B rebridge A to C (2222). 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: > > > > > > > > > > > > > > The first bridge is fine, and B can press *2 to bridge to C/2222. 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:2222 RTP RECV DTMF *:2000 > 2009-09-01 11:15:00.118195 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF 1:2000 > 2009-09-01 11:15:00.118195 [DEBUG] switch_ivr_async.c:1725 > sofia/external/unknown at 129.142.224.250 Processing meta digit '2' > [bridge::sofia/gateway/gw1/1111] > 2009-09-01 11:15:00.118195 [DEBUG] switch_core_session.c:813 Send > signal sofia/external/unknown at 129.142.224.250 [BREAK] > > Any ideas? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/d1b3c7d4/attachment.html From christian.loeschenkohl at xpirio.com Tue Sep 1 12:44:33 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Tue, 01 Sep 2009 21:44:33 +0200 Subject: [Freeswitch-users] conference question Message-ID: <4A9D79A1.5080107@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 323963096 at conf+flags{waste} the listeners conference is started with 323963096 at 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 L?schenkohl 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.loeschenkohl at xpirio.com From bjbrashier at gmail.com Tue Sep 1 14:32:41 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 1 Sep 2009 14:32:41 -0700 Subject: [Freeswitch-users] conference question In-Reply-To: <4A9D79A1.5080107@xpirio.com> References: <4A9D79A1.5080107@xpirio.com> Message-ID: <7bcfdd290909011432k2e806cc5if57dd612c6825cf@mail.gmail.com> 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 L?schenkohl : > 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 323963096 at conf+flags{waste} > the listeners conference is started with 323963096 at 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 L?schenkohl > 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.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Tue Sep 1 14:55:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Sep 2009 16:55:17 -0500 Subject: [Freeswitch-users] conference question In-Reply-To: <7bcfdd290909011432k2e806cc5if57dd612c6825cf@mail.gmail.com> References: <4A9D79A1.5080107@xpirio.com> <7bcfdd290909011432k2e806cc5if57dd612c6825cf@mail.gmail.com> Message-ID: <191c3a030909011455m53de9253he6f3da9b8df9b724@mail.gmail.com> 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 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 L?schenkohl : > > 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 323963096 at conf+flags{waste} > > the listeners conference is started with 323963096 at 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 L?schenkohl > > 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.loeschenkohl at xpirio.com > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/c3a59d38/attachment.html From christian.loeschenkohl at xpirio.com Tue Sep 1 14:56:05 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Tue, 01 Sep 2009 23:56:05 +0200 Subject: [Freeswitch-users] conference question In-Reply-To: <7bcfdd290909011432k2e806cc5if57dd612c6825cf@mail.gmail.com> References: <4A9D79A1.5080107@xpirio.com> <7bcfdd290909011432k2e806cc5if57dd612c6825cf@mail.gmail.com> Message-ID: <4A9D9875.1090707@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 L?schenkohl: >> 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 323963096 at conf+flags{waste} >> the listeners conference is started with 323963096 at 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 L?schenkohl >> 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.loeschenkohl at xpirio.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com From help at pdscc.com Tue Sep 1 15:33:22 2009 From: help at pdscc.com (Harondel J. Sibble) Date: Tue, 01 Sep 2009 15:33:22 -0700 Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3 In-Reply-To: <87f2f3b90908312333s5bf47d6et94ec24fbc51d6d24@mail.gmail.com> References: <20090823185152.D17845FE@sinclaire.sibble.net>, <20090901054337.7317F65B@sinclaire.sibble.net>, <87f2f3b90908312333s5bf47d6et94ec24fbc51d6d24@mail.gmail.com> Message-ID: <20090901223315.2E937BEE@sinclaire.sibble.net> 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. help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com (604) 739-3709 (voice) From anthony.minessale at gmail.com Tue Sep 1 15:50:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Sep 2009 17:50:05 -0500 Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3 In-Reply-To: <20090901223315.2E937BEE@sinclaire.sibble.net> References: <20090823185152.D17845FE@sinclaire.sibble.net> <20090901054337.7317F65B@sinclaire.sibble.net> <87f2f3b90908312333s5bf47d6et94ec24fbc51d6d24@mail.gmail.com> <20090901223315.2E937BEE@sinclaire.sibble.net> Message-ID: <191c3a030909011550o15e7a4c4sf6c5fc0472c7e925@mail.gmail.com> 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 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. > help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com > (604) 739-3709 (voice) > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/97cee85f/attachment.html From anthony.minessale at gmail.com Tue Sep 1 15:52:01 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 1 Sep 2009 17:52:01 -0500 Subject: [Freeswitch-users] conference question In-Reply-To: <4A9D9875.1090707@xpirio.com> References: <4A9D79A1.5080107@xpirio.com> <7bcfdd290909011432k2e806cc5if57dd612c6825cf@mail.gmail.com> <4A9D9875.1090707@xpirio.com> Message-ID: <191c3a030909011552h441981cla53dae8a439f748c@mail.gmail.com> 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 L?schenkohl > 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 L?schenkohl: > >> 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 323963096 at conf+flags{waste} > >> the listeners conference is started with 323963096 at 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 L?schenkohl > >> 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.loeschenkohl at xpirio.com > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > 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.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090901/bfa31b33/attachment.html From dujinfang at gmail.com Tue Sep 1 18:37:20 2009 From: dujinfang at gmail.com (Seven Du) Date: Wed, 2 Sep 2009 09:37:20 +0800 Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey In-Reply-To: <5896508D-3BA9-4B9A-A0DF-E8BE1EA69CB4@jerris.com> References: <4A92789C.5040900@gmx.net> <191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com> <4A93035B.5010907@gmx.net> <191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com> <6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org> <5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org> <45b2e91d7aa77afcf6430a14dde4b886.squirrel@correo.nodo50.org> <191c3a030908251354l25206d0ch19836c8996f43e84@mail.gmail.com> <5896508D-3BA9-4B9A-A0DF-E8BE1EA69CB4@jerris.com> Message-ID: 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/fooooooooooooooo.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.lists at it46.se> wrote: >>> >>>> Hi Brian, >>>> >>>>> From the CLI> >>>> >>>> freeswitch at 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 >>>> >>>> freeswitch at 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 at 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 at 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_minessale at hotmail.com >> %3Aanthony_minessale at hotmail.com> >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>> %3Aanthony.minessale at gmail.com> >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >> %3A888 at conference.freeswitch.org> >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org>> %2B888 at conference.freeswitch.org> >>> pstn:213-799-1400 >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From peter.olsson at visionutveckling.se Tue Sep 1 22:32:23 2009 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 2 Sep 2009 07:32:23 +0200 Subject: [Freeswitch-users] mod_opal Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED94DC@cooper> 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 ________________________________ Fr?n: Tihomir Culjaga Skickat: den 1 september 2009 19:20 Till: freeswitch-users at lists.freeswitch.org ?mne: 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 0000000 even if i set effective_caller_id_number to some value... T. On Tue, Sep 1, 2009 at 8:37 AM, Peter Olsson > 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 Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Tihomir Culjaga Skickat: den 1 september 2009 08:09 Till: freeswitch-users at lists.freeswitch.org ?mne: 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 > wrote: On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga > 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 tculjaga at nemesis:~/freeswitch-trunk$ tculjaga at nemesis:~/freeswitch-trunk$ tculjaga at 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 at 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 at 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:4a9d57f132931900217636! From aep.lists at it46.se Tue Sep 1 22:38:23 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Wed, 2 Sep 2009 07:38:23 +0200 Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey In-Reply-To: References: <4A92789C.5040900@gmx.net> <191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com> <4A93035B.5010907@gmx.net> <191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com> <6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org> <5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org> <45b2e91d7aa77afcf6430a14dde4b886.squirrel@correo.nodo50.org> <191c3a030908251354l25206d0ch19836c8996f43e84@mail.gmail.com> <5896508D-3BA9-4B9A-A0DF-E8BE1EA69CB4@jerris.com> Message-ID: <870ff3cf8ef1d369fa759614613eda57.squirrel@correo.nodo50.org> 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/fooooooooooooooo.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.lists at it46.se> wrote: >>>> >>>>> Hi Brian, >>>>> >>>>>> From the CLI> >>>>> >>>>> freeswitch at 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 >>>>> >>>>> freeswitch at 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 at 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 at 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_minessale at hotmail.com >>> %3Aanthony_minessale at hotmail.com> >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com>>> %3Aanthony.minessale at gmail.com> >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>> %3A888 at conference.freeswitch.org> >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org>>> %2B888 at conference.freeswitch.org> >>>> pstn:213-799-1400 >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From juanbackson at gmail.com Tue Sep 1 23:31:26 2009 From: juanbackson at gmail.com (Juan Backson) Date: Wed, 2 Sep 2009 14:31:26 +0800 Subject: [Freeswitch-users] Fwd: freeswitch odbc problem In-Reply-To: <87f2f3b90908280849w34bda66dt4d6ac3227b1fddf3@mail.gmail.com> References: <27c25bc40908272351g345ad1e2w31dc945e8f9a0331@mail.gmail.com> <87f2f3b90908280849w34bda66dt4d6ac3227b1fddf3@mail.gmail.com> Message-ID: <27c25bc40909012331x7483c426ka5cd0269028317d8@mail.gmail.com> hi Michael, Thank you for your help. It works now. JB ---------- Forwarded message ---------- From: Michael Collins Date: Fri, Aug 28, 2009 at 11:49 PM Subject: Re: [Freeswitch-users] freeswitch odbc problem To: freeswitch-users at lists.freeswitch.org Juan, Is this a new OS install? Make sure that you have all of the prerequisites installed. Hopefully you're on CentOS and can just do this: yum -y install unixODBC-devel gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf automake db4-devel gdbm-devel ncurses-devel wget Let us know how it goes. -MC On Thu, Aug 27, 2009 at 11:51 PM, Juan Backson wrote: > Hi, > In a new environment, I am getting the following error when building the > latest freeswitch from svn. Does anyone know how to resolve it? > > gcc -I/usr/src/freeswitch-snapshot/src/include > -I/usr/src/freeswitch-snapshot/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -g -O2 -Wall -std=c99 -pedantic -o .libs/freeswitch freeswitch-switch.o -lm > ./.libs/libfreeswitch.so > -L/usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib > /usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib/.libs/libexpat.a > /usr/src/freeswitch-snapshot/libs/apr/.libs/libapr-1.a > -L/usr/src/freeswitch-snapshot/libs/srtp -L/usr/kerberos/lib64 > libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt -lpthread > libs/libedit/src/.libs/libedit.a -lssl -lcrypto -ldl -lz -lncurses > -Wl,--rpath -Wl,/usr/local/freeswitch/lib > ./.libs/libfreeswitch.so: undefined reference to `SQLGetData' > ./.libs/libfreeswitch.so: undefined reference to `SQLDisconnect' > ./.libs/libfreeswitch.so: undefined reference to `SQLSetConnectAttr' > ./.libs/libfreeswitch.so: undefined reference to `SQLGetDiagRec' > ./.libs/libfreeswitch.so: undefined reference to `SQLConnect' > ./.libs/libfreeswitch.so: undefined reference to `SQLExecute' > ./.libs/libfreeswitch.so: undefined reference to `SQLNumResultCols' > ./.libs/libfreeswitch.so: undefined reference to `SQLFreeHandle' > ./.libs/libfreeswitch.so: undefined reference to `SQLGetInfo' > ./.libs/libfreeswitch.so: undefined reference to `SQLAllocHandle' > ./.libs/libfreeswitch.so: undefined reference to `SQLDriverConnect' > ./.libs/libfreeswitch.so: undefined reference to `SQLPrepare' > ./.libs/libfreeswitch.so: undefined reference to `SQLError' > ./.libs/libfreeswitch.so: undefined reference to `SQLRowCount' > ./.libs/libfreeswitch.so: undefined reference to `SQLDescribeCol' > ./.libs/libfreeswitch.so: undefined reference to `SQLFetch' > ./.libs/libfreeswitch.so: undefined reference to `SQLSetEnvAttr' > collect2: ld returned 1 exit status > make[2]: *** [freeswitch] Error 1 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/292f1453/attachment.html From lakindia89 at gmail.com Wed Sep 2 02:42:40 2009 From: lakindia89 at gmail.com (lakshmanan) Date: Wed, 2 Sep 2009 02:42:40 -0700 (PDT) Subject: [Freeswitch-users] How to find which leg got terminated Message-ID: <25254511.post@talk.nabble.com> I want to find which leg has cut the call? Is it possible to find that in freeswitch? For example: A and B are speaking. If B cuts the call, then I need to play appropriate message to A, and if A cuts the call, I need to play some messages to B. What is the way to do this? -- View this message in context: http://www.nabble.com/How-to-find-which-leg-got-terminated-tp25254511p25254511.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From harry at vangberg.name Wed Sep 2 03:00:46 2009 From: harry at vangberg.name (Harry Vangberg) Date: Wed, 2 Sep 2009 12:00:46 +0200 Subject: [Freeswitch-users] Bind_meta_app and second-degree bridge (.. this is a bad title.) In-Reply-To: <191c3a030909011102k38ba525dkb1c89fd8ea84f23@mail.gmail.com> References: <74d41a3d0909010417m5654adebia4be53281574864@mail.gmail.com> <191c3a030909011102k38ba525dkb1c89fd8ea84f23@mail.gmail.com> Message-ID: <74d41a3d0909020300h4bf3c41bjbf973c536849a693@mail.gmail.com> Ah. att_xfer seems nice. But, it still doesn't allow C to eventually rebridge A to B (or possibly D, E etc) at some point in the conversation, where the caller needs to talk with somebody else. 2009/9/1 Anthony Minessale : > 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 wrote: >> >> My basic functionality is this: A calls in, is bridged to B (1111). I >> use bind_meta_app to let B rebridge A to C (2222). 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: >> >> ? >> ? ? >> ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? >> ? ? >> ? >> >> The first bridge is fine, and B can press *2 to bridge to C/2222. 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:2222 RTP RECV DTMF *:2000 >> 2009-09-01 11:15:00.118195 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF 1:2000 >> 2009-09-01 11:15:00.118195 [DEBUG] switch_ivr_async.c:1725 >> sofia/external/unknown at 129.142.224.250 Processing meta digit '2' >> [bridge::sofia/gateway/gw1/1111] >> 2009-09-01 11:15:00.118195 [DEBUG] switch_core_session.c:813 Send >> signal sofia/external/unknown at 129.142.224.250 [BREAK] >> >> Any ideas? >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mayamatakeshi at gmail.com Wed Sep 2 03:59:12 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Wed, 2 Sep 2009 19:59:12 +0900 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. Message-ID: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> Hello, I'm testing FS support for the header Path (FS is behind opensips). It pretty much works: I tested calling from one user to the other and calls work perfectly. However, I've noticed that when I register my terminal directly with FS without going thru the proxy, I receive an unsolicited NOTIFY containing Message-Waiting information. But when I register via proxy, FS doesn't send this NOTIFY. What could be causing this difference of behavior? (enabling debug (F8) doesn't show anything for registration handling). I have just updated to trunk Revision 14729 and this behavior persists. Also, is it possible to disable this unsolicited MWI notification? (I want to send it only to subscribed terminals). regards, takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/af5f758e/attachment.html From dome at tel.co.th Wed Sep 2 05:39:43 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 2 Sep 2009 19:39:43 +0700 Subject: [Freeswitch-users] outbould PHP ESL Message-ID: <8ccbff060909020539n7931ca6av6f7d9bab5ee9cdf4@mail.gmail.com> Dear sir, How to get digit from outbound php esl ? example in my php echo "sendmsg\n"; echo "call-command: execute\n"; echo "execute-app-name: read\n"; echo "execute-app-arg: 0 20 /opt/freeswitch/sounds/th/tuxza/welcome.wav res 5000 #\n\n"; How to get res ? best regards. Dome C. From juanbackson at gmail.com Wed Sep 2 05:50:30 2009 From: juanbackson at gmail.com (Juan Backson) Date: Wed, 2 Sep 2009 20:50:30 +0800 Subject: [Freeswitch-users] register problem while using external profile Message-ID: <27c25bc40909020550n2427ba17y2818ecd4c4298727@mail.gmail.com> Hi, Things are working find before I tried using public IP ( behind NAT ) to register IP phones. I am getting: 2009-09-02 20:46:50.575837 [WARNING] sofia_reg.c:1713 Can't find user [180001 at public-ip] You must define a domain called 'public-ip' in your directory and add a user with the id="180001" attribute and you must configure your device to use the proper domain in it's authentication credentials. I am using the external sip profile in this case. Does anyone know why and how to fix this problem. jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/b69af1ad/attachment.html From fraunhofer.lists.freeswitch-001 at traced.net Wed Sep 2 05:59:39 2009 From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer) Date: Wed, 2 Sep 2009 14:59:39 +0200 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com> <3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com> <20090828001050.GA8108@jdc.jasonjgw.net> <86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com> <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org> Message-ID: Hello *, 2009/8/31 Rupa Schomaker : > Isn't there a known issue with lua+sql leaking memory on some platforms? just lua, no sql in use :) > On Mon, Aug 31, 2009 at 8:32 AM, Brian West wrote: >> Use valgrind. i tried that... in the beginning valgrind segfaultet several times with some error message that it's unable to monitor so many threads. vg_alloc_ThreadState: no free slots available Increase VG_N_THREADS, rebuild and try again. valgrind: the 'impossible' happened: VG_N_THREADS is too low it happeneded with 400 concurrent calls pretty soon, with 100 it took a while, but at least it consumed half a gig of ram. Can someone interpret the file it generated so far? i put it at http://ns42.ath.cx/B0GdWh/vg.log.bz2 (5k compressed, 452k uncompressed) or is there just not enough information contained? I'll give it another shot with 50 concurrent calls but i guess this will take ages... Thx in advance Beni. From dftoro at yahoo.com Wed Sep 2 06:01:13 2009 From: dftoro at yahoo.com (Diego Toro) Date: Wed, 2 Sep 2009 06:01:13 -0700 (PDT) Subject: [Freeswitch-users] FS performance under windows In-Reply-To: Message-ID: <892489.91947.qm@web33505.mail.mud.yahoo.com> What is the reason for saying this? Perhaps the effort of the development group of FS has been wasted trying to support Windows as a platform for production systems? Diego http://lacarretade.blogspot.com/ --- On Tue, 9/1/09, Muhammad Shahzad wrote: From: Muhammad Shahzad Subject: Re: [Freeswitch-users] FS performance under windows To: freeswitch-users at lists.freeswitch.org Date: Tuesday, September 1, 2009, 4:00 AM 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 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 at 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_786pk at hotmail.com Email: shaheryarkh at googlemail.com -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/7b52404e/attachment.html From t.mahe at telemaque.fr Wed Sep 2 06:16:18 2009 From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=) Date: Wed, 02 Sep 2009 15:16:18 +0200 Subject: [Freeswitch-users] outbould PHP ESL In-Reply-To: <8ccbff060909020539n7931ca6av6f7d9bab5ee9cdf4@mail.gmail.com> References: <8ccbff060909020539n7931ca6av6f7d9bab5ee9cdf4@mail.gmail.com> Message-ID: <4A9E7022.4000800@telemaque.fr> Hi, just a fast 2cent: get var via channel status ? ( variable_res ) Dome Charoenyost a ?crit : > Dear sir, > > How to get digit from outbound php esl ? > example in my php > > echo "sendmsg\n"; > echo "call-command: execute\n"; > echo "execute-app-name: read\n"; > echo "execute-app-arg: 0 20 > /opt/freeswitch/sounds/th/tuxza/welcome.wav res 5000 #\n\n"; > > How to get res ? > > best regards. > > Dome C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From helmut.kuper at ewetel.de Wed Sep 2 06:44:05 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Wed, 02 Sep 2009 15:44:05 +0200 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A658FDA.8080908@ewetel.de> References: <4A3266E5.2000702@ewetel.de> <4A328E8E.6030607@freeswitch.org> <4A35F0F0.50406@ewetel.de> <4A643C7D.7010209@ewetel.de> <4A658FDA.8080908@ewetel.de> Message-ID: <4A9E76A5.7030200@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, just to close this thread, I would like to say that after 21000 pstn-calls and 45 days up time (old stack: max 10 days) the ISDN call resource management based on Q931 timers delivered by the *new* Q931 stack is still clean. Each done call was torn down to the DOWN state cleanly. This doesn't mean that the new stack is done - but for simple calls to and from PSTN it works very good compared to the old Q931 stack. regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKnnak4tZeNddg3dwRAgPEAJsH+wg4CKV2MKQPAUbO69iijR/r4ACgug+a zLeZX1bmYNF4UlirU0XJHfc= =B/LZ -----END PGP SIGNATURE----- From brian at freeswitch.org Wed Sep 2 06:47:12 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Sep 2009 08:47:12 -0500 Subject: [Freeswitch-users] register problem while using external profile In-Reply-To: <27c25bc40909020550n2427ba17y2818ecd4c4298727@mail.gmail.com> References: <27c25bc40909020550n2427ba17y2818ecd4c4298727@mail.gmail.com> Message-ID: <7AE1E7A6-D97A-4A5C-9968-EFDF5428FF4C@freeswitch.org> You don't have to do that anymore... the default profile on 5060 will work when the users are on the public internet also. No need to have two profiles anymore. /b On Sep 2, 2009, at 7:50 AM, Juan Backson wrote: > Hi, > > Things are working find before I tried using public IP ( behind > NAT ) to register IP phones. I am getting: > > 2009-09-02 20:46:50.575837 [WARNING] sofia_reg.c:1713 Can't find > user [180001 at public-ip] > You must define a domain called 'public-ip' in your directory and > add a user with the id="180001" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > > > I am using the external sip profile in this case. Does anyone know > why and how to fix this problem. > > jb > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Wed Sep 2 06:48:31 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Sep 2009 08:48:31 -0500 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com> <3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com> <20090828001050.GA8108@jdc.jasonjgw.net> <86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com> <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org> Message-ID: What are you doing in these lua scripts? Because there are a few things you can do in the lua script itself that will cause you to leak like crazy due to improper use. /b On Sep 2, 2009, at 7:59 AM, Benedikt Fraunhofer wrote: > Hello *, > > 2009/8/31 Rupa Schomaker : >> Isn't there a known issue with lua+sql leaking memory on some >> platforms? > > just lua, no sql in use :) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/7a8a9269/attachment.html From brian at freeswitch.org Wed Sep 2 06:50:29 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Sep 2009 08:50:29 -0500 Subject: [Freeswitch-users] outbould PHP ESL In-Reply-To: <4A9E7022.4000800@telemaque.fr> References: <8ccbff060909020539n7931ca6av6f7d9bab5ee9cdf4@mail.gmail.com> <4A9E7022.4000800@telemaque.fr> Message-ID: uuid_getvar /b On Sep 2, 2009, at 8:16 AM, Tristan Mah? wrote: > Hi, > > just a fast 2cent: > > get var via channel status ? ( variable_res ) From brian at freeswitch.org Wed Sep 2 06:50:01 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Sep 2009 08:50:01 -0500 Subject: [Freeswitch-users] FS performance under windows In-Reply-To: <892489.91947.qm@web33505.mail.mud.yahoo.com> References: <892489.91947.qm@web33505.mail.mud.yahoo.com> Message-ID: <91609A0C-AA80-4707-A62B-6FF129204DC7@freeswitch.org> I know people that have deployed on windows... not a huge problem just hasn't been load tested like linux... we don't have the resources or time to load test every single platform, tune and tweak it. The community can help out with this area a lot. /b On Sep 2, 2009, at 8:01 AM, Diego Toro wrote: > What is the reason for saying this? Perhaps the effort of the > development group of FS has been wasted trying to support Windows as > a platform for production systems? > > Diego > http://lacarretade.blogspot.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/2cfe26f0/attachment.html From fraunhofer.lists.freeswitch-001 at traced.net Wed Sep 2 07:19:14 2009 From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer) Date: Wed, 2 Sep 2009 16:19:14 +0200 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com> <20090828001050.GA8108@jdc.jasonjgw.net> <86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com> <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org> Message-ID: Hello Brian, 2009/9/2 Brian West : > What are you doing in these lua scripts? ?Because there are a few things you > can do in the lua script itself that will cause you to leak like crazy due > to improper use. > /b the setup is the same as in http://jira.freeswitch.org/browse/MODSOFIA-22 one is ----- local reason = session:getVariable("originate_disposition"); session:setAutoHangup(false); if(reason) then if(reason == "NO_ANSWER") then -- nothing end if(reason == "USER_BUSY") then -- nothing end end freeswitch.consoleLog(... ------ anotherone is ---- local sess = "nil"; if(argv[1]) then sess=argv[1]; end freeswitch.consoleLog(... api = freeswitch.API(); local res = api:execute("sched_api" ... freeswitch.consoleLog(... ---- and the scheduled script does --- function log(msg) freeswitch.consoleLog("notice", "c2c-hangup-timeout.lua: " .. msg .. "\n"); end local sess = argv[1]; if(sess) then freeswitch.consoleLog("INFO", "hangup-timeout.lua for uuid " .. sess .. "\n"); api = freeswitch.API(); local stillValid = api:execute("uuid_getvar", sess .. " Dummy-DoesChannelExists"); if(stillValid:sub(1,4) == "-ERR") then log("session uuid " .. sess .. " disappeared (nothing bad)"); else -- this is important!!! Otherwise the aleg get's just hung up! api:execute("uuid_media", sess); api:execute("uuid_transfer", sess .. " -both timeout"); end else -- /if(sess) log("called with nil session?"); end -- /if(sess) --- i don't have to do some kind of "free()" on all local variables, do i? Thx. Beni. From math.parent at gmail.com Wed Sep 2 07:27:56 2009 From: math.parent at gmail.com (Mathieu Parent) Date: Wed, 2 Sep 2009 16:27:56 +0200 Subject: [Freeswitch-users] Run a command on event Message-ID: <960738410909020727q3286bce9q2ff7d4edfcf6e025@mail.gmail.com> Hi, I wanted to run an "originate" command when a MESSAGE_WAITING event is fired. Is there a simpler way than creating a daemon listening to the event socket ? Thanks Mathieu Parent From brian at freeswitch.org Wed Sep 2 07:37:26 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Sep 2009 09:37:26 -0500 Subject: [Freeswitch-users] Run a command on event In-Reply-To: <960738410909020727q3286bce9q2ff7d4edfcf6e025@mail.gmail.com> References: <960738410909020727q3286bce9q2ff7d4edfcf6e025@mail.gmail.com> Message-ID: <1F7FA356-12F2-4CCA-A82D-708D70210D3B@freeswitch.org> Using esl + perl you could do it. /b On Sep 2, 2009, at 9:27 AM, Mathieu Parent wrote: > Hi, > > I wanted to run an "originate" command when a MESSAGE_WAITING event > is fired. > > Is there a simpler way than creating a daemon listening to the event > socket ? > > Thanks > > Mathieu Parent From kadantsev.d at gmail.com Wed Sep 2 07:42:34 2009 From: kadantsev.d at gmail.com (Dmitry Kadantsev) Date: Wed, 2 Sep 2009 16:42:34 +0200 Subject: [Freeswitch-users] FS performance under windows In-Reply-To: <91609A0C-AA80-4707-A62B-6FF129204DC7@freeswitch.org> References: <892489.91947.qm@web33505.mail.mud.yahoo.com> <91609A0C-AA80-4707-A62B-6FF129204DC7@freeswitch.org> Message-ID: <681a20520909020742q59470e78pf73f0abfb8337c9d@mail.gmail.com> As far as we'll solve all FS configuration issues I think we'll make some tests on Windows platform and will share results with community. Please if someone has an opportunity to test FS on modern windows server editions (32/64 bit) - submit your test results too. It is interesting to put together some statistic. -- Best regards, Dmitry Kadantsev http://www.doxwox.com - Best web meeting and online collaboration tool. On Wed, Sep 2, 2009 at 3:50 PM, Brian West wrote: > I know people that have deployed on windows... not a huge problem just > hasn't been load tested like linux... we don't have the resources or time to > load test every single platform, tune and tweak it. The community can help > out with this area a lot. > /b > > On Sep 2, 2009, at 8:01 AM, Diego Toro wrote: > > What is the reason for saying this? Perhaps the effort of the development > group of FS has been wasted trying to support Windows as a platform for > production systems? > > Diego > http://lacarretade.blogspot.com/ > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/9ab05257/attachment.html From diego.viola at gmail.com Wed Sep 2 07:46:32 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 2 Sep 2009 14:46:32 +0000 Subject: [Freeswitch-users] Run a command on event In-Reply-To: <960738410909020727q3286bce9q2ff7d4edfcf6e025@mail.gmail.com> References: <960738410909020727q3286bce9q2ff7d4edfcf6e025@mail.gmail.com> Message-ID: <86a32abc0909020746n1250604ds3ebaf3a3d7abdd21@mail.gmail.com> Or you can do something like this with Ruby + FSR: http://pastebin.freeswitch.org/10184 On Wed, Sep 2, 2009 at 2:27 PM, Mathieu Parent wrote: > Hi, > > I wanted to run an "originate" command when a MESSAGE_WAITING event is > fired. > > Is there a simpler way than creating a daemon listening to the event socket > ? > > Thanks > > Mathieu Parent > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/ed1c552d/attachment.html From anthony.minessale at gmail.com Wed Sep 2 07:54:49 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Sep 2009 09:54:49 -0500 Subject: [Freeswitch-users] How to find which leg got terminated In-Reply-To: <25254511.post@talk.nabble.com> References: <25254511.post@talk.nabble.com> Message-ID: <191c3a030909020754g869951fr5e4dc39bcf9ad452@mail.gmail.com> sip_hangup_disposition variable On Wed, Sep 2, 2009 at 4:42 AM, lakshmanan wrote: > > > I want to find which leg has cut the call? Is it possible to find that in > freeswitch? > For example: > A and B are speaking. If B cuts the call, then I need to play appropriate > message to A, and if A cuts the call, I need to play some messages to B. > What is the way to do this? > > -- > View this message in context: > http://www.nabble.com/How-to-find-which-leg-got-terminated-tp25254511p25254511.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/f7f1108e/attachment.html From dome at tel.co.th Wed Sep 2 07:56:24 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 2 Sep 2009 21:56:24 +0700 Subject: [Freeswitch-users] outbould PHP ESL In-Reply-To: References: <8ccbff060909020539n7931ca6av6f7d9bab5ee9cdf4@mail.gmail.com> <4A9E7022.4000800@telemaque.fr> Message-ID: <8ccbff060909020756t167cd564qde77e849227a4f7@mail.gmail.com> I follow http://wiki.freeswitch.org/wiki/PHP_ESL#ivrd how to get from php ? Dome C. ------------------------------------------------------ #!/usr/bin/php -q 2009/9/2 Brian West : > uuid_getvar > > /b > > On Sep 2, 2009, at 8:16 AM, Tristan Mah? wrote: > >> Hi, >> >> just a fast 2cent: >> >> get var via channel status ? ( variable_res ) > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From woof at iwoof.org Wed Sep 2 08:02:04 2009 From: woof at iwoof.org (Andy Spitzer) Date: Wed, 02 Sep 2009 11:02:04 -0400 Subject: [Freeswitch-users] conference question In-Reply-To: <191c3a030909011552h441981cla53dae8a439f748c@mail.gmail.com> References: <4A9D79A1.5080107@xpirio.com> <7bcfdd290909011432k2e806cc5if57dd612c6825cf@mail.gmail.com> <4A9D9875.1090707@xpirio.com> <191c3a030909011552h441981cla53dae8a439f748c@mail.gmail.com> Message-ID: Woof! On Tue, 01 Sep 2009 18:52:01 -0400, Anthony Minessale wrote: > 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. Perhaps some listeners are hitting the "unmute" DTMF key? --Woof! From anthony.minessale at gmail.com Wed Sep 2 08:06:47 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Sep 2009 10:06:47 -0500 Subject: [Freeswitch-users] Bind_meta_app and second-degree bridge (.. this is a bad title.) In-Reply-To: <74d41a3d0909020300h4bf3c41bjbf973c536849a693@mail.gmail.com> References: <74d41a3d0909010417m5654adebia4be53281574864@mail.gmail.com> <191c3a030909011102k38ba525dkb1c89fd8ea84f23@mail.gmail.com> <74d41a3d0909020300h4bf3c41bjbf973c536849a693@mail.gmail.com> Message-ID: <191c3a030909020806t60e034d4n7f62ece63639856a@mail.gmail.com> instead of bridge or att_xfer then use transfer to transfer to an extension that does the bridge. or transfer to the inline dialplan. On Wed, Sep 2, 2009 at 5:00 AM, Harry Vangberg wrote: > Ah. att_xfer seems nice. But, it still doesn't allow C to eventually > rebridge A to B (or possibly D, E etc) at some point in the > conversation, where the caller needs to talk with somebody else. > > 2009/9/1 Anthony Minessale : > > 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 > wrote: > >> > >> My basic functionality is this: A calls in, is bridged to B (1111). I > >> use bind_meta_app to let B rebridge A to C (2222). 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: > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> The first bridge is fine, and B can press *2 to bridge to C/2222. 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:2222 RTP RECV DTMF > *:2000 > >> 2009-09-01 11:15:00.118195 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF > 1:2000 > >> 2009-09-01 11:15:00.118195 [DEBUG] switch_ivr_async.c:1725 > >> sofia/external/unknown at 129.142.224.250 Processing meta digit '2' > >> [bridge::sofia/gateway/gw1/1111] > >> 2009-09-01 11:15:00.118195 [DEBUG] switch_core_session.c:813 Send > >> signal sofia/external/unknown at 129.142.224.250 [BREAK] > >> > >> Any ideas? > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/681488f4/attachment.html From anthony.minessale at gmail.com Wed Sep 2 08:11:41 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Sep 2009 10:11:41 -0500 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com> <3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com> <20090828001050.GA8108@jdc.jasonjgw.net> <86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com> <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org> Message-ID: <191c3a030909020811q6c04518au514fe8e9d6c8734e@mail.gmail.com> You have to reduce the load when running valgrind. On Wed, Sep 2, 2009 at 7:59 AM, Benedikt Fraunhofer < fraunhofer.lists.freeswitch-001 at traced.net> wrote: > Hello *, > > 2009/8/31 Rupa Schomaker : > > Isn't there a known issue with lua+sql leaking memory on some platforms? > > just lua, no sql in use :) > > > On Mon, Aug 31, 2009 at 8:32 AM, Brian West wrote: > >> Use valgrind. > > i tried that... in the beginning valgrind segfaultet several times > with some error message that it's unable to monitor so many threads. > > vg_alloc_ThreadState: no free slots available > Increase VG_N_THREADS, rebuild and try again. > > valgrind: the 'impossible' happened: > VG_N_THREADS is too low > > it happeneded with 400 concurrent calls pretty soon, with 100 it took > a while, but at least it consumed half a gig of ram. > > Can someone interpret the file it generated so far? i put it at > > http://ns42.ath.cx/B0GdWh/vg.log.bz2 > > (5k compressed, 452k uncompressed) or is there just not enough > information contained? > > I'll give it another shot with 50 concurrent calls but i guess this > will take ages... > > Thx in advance > > Beni. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/10207ac0/attachment.html From raffaele.p.guidi at gmail.com Wed Sep 2 08:13:29 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Wed, 2 Sep 2009 17:13:29 +0200 Subject: [Freeswitch-users] FS performance under windows In-Reply-To: <91609A0C-AA80-4707-A62B-6FF129204DC7@freeswitch.org> References: <892489.91947.qm@web33505.mail.mud.yahoo.com> <91609A0C-AA80-4707-A62B-6FF129204DC7@freeswitch.org> Message-ID: I'm planning to deploy on windows a small call center (around 50 people) and willing to help anyhow. I will be able to test on machines mounting windows 2003 server. Is there a standard test that could be employed to correctly benchmark the results? On Wed, Sep 2, 2009 at 15:50, Brian West wrote: > I know people that have deployed on windows... not a huge problem just > hasn't been load tested like linux... we don't have the resources or time to > load test every single platform, tune and tweak it. The community can help > out with this area a lot. > /b > > On Sep 2, 2009, at 8:01 AM, Diego Toro wrote: > > What is the reason for saying this? Perhaps the effort of the development > group of FS has been wasted trying to support Windows as a platform for > production systems? > > Diego > http://lacarretade.blogspot.com/ > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/7c65a0b4/attachment-0001.html From anthony.minessale at gmail.com Wed Sep 2 08:15:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Sep 2009 10:15:33 -0500 Subject: [Freeswitch-users] memory leak In-Reply-To: <191c3a030909020811q6c04518au514fe8e9d6c8734e@mail.gmail.com> References: <3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com> <20090828001050.GA8108@jdc.jasonjgw.net> <86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com> <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org> <191c3a030909020811q6c04518au514fe8e9d6c8734e@mail.gmail.com> Message-ID: <191c3a030909020815k6006f96fh3df8512018f9b28d@mail.gmail.com> run it slower and make sure it shuts down clean. valgrind --tool=memcheck --log-file-exactly=vg.log --leak-check=full --leak-resolution=high --show-reachable=yes /path/to/freeswitch -vg On Wed, Sep 2, 2009 at 10:11 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You have to reduce the load when running valgrind. > > > > On Wed, Sep 2, 2009 at 7:59 AM, Benedikt Fraunhofer < > fraunhofer.lists.freeswitch-001 at traced.net> wrote: > >> Hello *, >> >> 2009/8/31 Rupa Schomaker : >> > Isn't there a known issue with lua+sql leaking memory on some platforms? >> >> just lua, no sql in use :) >> >> > On Mon, Aug 31, 2009 at 8:32 AM, Brian West >> wrote: >> >> Use valgrind. >> >> i tried that... in the beginning valgrind segfaultet several times >> with some error message that it's unable to monitor so many threads. >> >> vg_alloc_ThreadState: no free slots available >> Increase VG_N_THREADS, rebuild and try again. >> >> valgrind: the 'impossible' happened: >> VG_N_THREADS is too low >> >> it happeneded with 400 concurrent calls pretty soon, with 100 it took >> a while, but at least it consumed half a gig of ram. >> >> Can someone interpret the file it generated so far? i put it at >> >> http://ns42.ath.cx/B0GdWh/vg.log.bz2 >> >> (5k compressed, 452k uncompressed) or is there just not enough >> information contained? >> >> I'll give it another shot with 50 concurrent calls but i guess this >> will take ages... >> >> Thx in advance >> >> Beni. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/d3155ad5/attachment.html From msc at freeswitch.org Wed Sep 2 08:24:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Sep 2009 08:24:09 -0700 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A9E76A5.7030200@ewetel.de> References: <4A3266E5.2000702@ewetel.de> <4A328E8E.6030607@freeswitch.org> <4A35F0F0.50406@ewetel.de> <4A643C7D.7010209@ewetel.de> <4A658FDA.8080908@ewetel.de> <4A9E76A5.7030200@ewetel.de> Message-ID: <87f2f3b90909020824i2f17ca8cwd43f04fe229f0a3f@mail.gmail.com> w00t! On Wed, Sep 2, 2009 at 6:44 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > just to close this thread, I would like to say that after 21000 > pstn-calls and 45 days up time (old stack: max 10 days) the ISDN call > resource management based on Q931 timers delivered by the *new* Q931 > stack is still clean. Each done call was torn down to the DOWN state > cleanly. > > This doesn't mean that the new stack is done - but for simple calls to > and from PSTN it works very good compared to the old Q931 stack. > > regards > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFKnnak4tZeNddg3dwRAgPEAJsH+wg4CKV2MKQPAUbO69iijR/r4ACgug+a > zLeZX1bmYNF4UlirU0XJHfc= > =B/LZ > -----END PGP SIGNATURE----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/f695ded3/attachment.html From msc at freeswitch.org Wed Sep 2 08:25:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Sep 2009 08:25:10 -0700 Subject: [Freeswitch-users] outbould PHP ESL In-Reply-To: <8ccbff060909020756t167cd564qde77e849227a4f7@mail.gmail.com> References: <8ccbff060909020539n7931ca6av6f7d9bab5ee9cdf4@mail.gmail.com> <4A9E7022.4000800@telemaque.fr> <8ccbff060909020756t167cd564qde77e849227a4f7@mail.gmail.com> Message-ID: <87f2f3b90909020825x516c2054l7a54dc6a1b61691c@mail.gmail.com> Are you trying to get a channel variable or capture DTMF input from the caller? -MC On Wed, Sep 2, 2009 at 7:56 AM, Dome Charoenyost wrote: > I follow > http://wiki.freeswitch.org/wiki/PHP_ESL#ivrd > > how to get from php ? > > > Dome C. > ------------------------------------------------------ > #!/usr/bin/php -q > > > // set a couple of things so we dont kill the system > ob_implicit_flush(true); > set_time_limit(30); > > // Open stdin so we can read the AGI data in > $in = fopen("php://stdin", "r"); > > // Connect > echo "connect\n\n"; > > // Answer > echo "sendmsg\n"; > echo "call-command: execute\n"; > echo "execute-app-name: answer\n\n"; > > echo "sendmsg\n"; > echo "call-command: execute\n"; > echo "execute-app-name: read\n"; > echo "execute-app-arg: 0 20 > /opt/freeswitch/sounds/th/tuxza/welcome.wav res 5000 #\n\n"; > > // Wait > sleep(5); > > // Hangup > echo "sendmsg\n"; > echo "call-command: hangup\n\n"; > > fclose($in); > > ?> > > > 2009/9/2 Brian West : > > uuid_getvar > > > > /b > > > > On Sep 2, 2009, at 8:16 AM, Tristan Mah? wrote: > > > >> Hi, > >> > >> just a fast 2cent: > >> > >> get var via channel status ? ( variable_res ) > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/65226237/attachment.html From fraunhofer.lists.freeswitch-001 at traced.net Wed Sep 2 08:26:51 2009 From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer) Date: Wed, 2 Sep 2009 17:26:51 +0200 Subject: [Freeswitch-users] memory leak In-Reply-To: <191c3a030909020815k6006f96fh3df8512018f9b28d@mail.gmail.com> References: <3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com> <20090828001050.GA8108@jdc.jasonjgw.net> <86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com> <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org> <191c3a030909020811q6c04518au514fe8e9d6c8734e@mail.gmail.com> <191c3a030909020815k6006f96fh3df8512018f9b28d@mail.gmail.com> Message-ID: Hello Anthony, 2009/9/2 Anthony Minessale : > run it slower and make sure it shuts down clean. i already reduced load 37% but that didnt help, now i'm down to 25% and it's running. > valgrind --tool=memcheck --log-file-exactly=vg.log --leak-check=full > --leak-resolution=high --show-reachable=yes /path/to/freeswitch -vg that's the line in the wiki? I used that but "log-file-exactly" is not available in my valgrind-3.3.0-Debian, just "log-file". Does this make any difference? Also it wont shut down cleanly but oom-coredump (some assertion failed) and if i stop the loadgen after some time and try to 'fsctl shutdown' it wont cleanly bring it down, too. (see first post, last line printed is switch_core_memory.c:567 Stopping memory pool queue. ) Thx. Beni. From diego.viola at gmail.com Wed Sep 2 08:28:06 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 2 Sep 2009 15:28:06 +0000 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <4A9E76A5.7030200@ewetel.de> References: <4A3266E5.2000702@ewetel.de> <4A328E8E.6030607@freeswitch.org> <4A35F0F0.50406@ewetel.de> <4A643C7D.7010209@ewetel.de> <4A658FDA.8080908@ewetel.de> <4A9E76A5.7030200@ewetel.de> Message-ID: <86a32abc0909020828h429a98a8uabc892c42427469f@mail.gmail.com> Great stuff, thanks for your hard work :). Keep it up. Regards, Diego On Wed, Sep 2, 2009 at 1:44 PM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > just to close this thread, I would like to say that after 21000 > pstn-calls and 45 days up time (old stack: max 10 days) the ISDN call > resource management based on Q931 timers delivered by the *new* Q931 > stack is still clean. Each done call was torn down to the DOWN state > cleanly. > > This doesn't mean that the new stack is done - but for simple calls to > and from PSTN it works very good compared to the old Q931 stack. > > regards > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFKnnak4tZeNddg3dwRAgPEAJsH+wg4CKV2MKQPAUbO69iijR/r4ACgug+a > zLeZX1bmYNF4UlirU0XJHfc= > =B/LZ > -----END PGP SIGNATURE----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/649c287f/attachment-0001.html From harry at vangberg.name Wed Sep 2 08:32:50 2009 From: harry at vangberg.name (Harry Vangberg) Date: Wed, 2 Sep 2009 17:32:50 +0200 Subject: [Freeswitch-users] Bind_meta_app and second-degree bridge (.. this is a bad title.) In-Reply-To: <191c3a030909020806t60e034d4n7f62ece63639856a@mail.gmail.com> References: <74d41a3d0909010417m5654adebia4be53281574864@mail.gmail.com> <191c3a030909011102k38ba525dkb1c89fd8ea84f23@mail.gmail.com> <74d41a3d0909020300h4bf3c41bjbf973c536849a693@mail.gmail.com> <191c3a030909020806t60e034d4n7f62ece63639856a@mail.gmail.com> Message-ID: <74d41a3d0909020832u6fd3784ex6d8aaf20e605f45c@mail.gmail.com> That's what I ended up with earlier today. Transfering to another extension that does the att_xfer/bridge and rebinds meta app. I think it works. Unfortunately my third phone is out of power, so haven't had much chance to test it. 2009/9/2 Anthony Minessale : > instead of bridge or att_xfer then use transfer to transfer to an extension > that does the bridge. > or transfer to the inline dialplan. > > > On Wed, Sep 2, 2009 at 5:00 AM, Harry Vangberg wrote: >> >> Ah. att_xfer seems nice. But, it still doesn't allow C to eventually >> rebridge A to B (or possibly D, E etc) at some point in the >> conversation, where the caller needs to talk with somebody else. >> >> 2009/9/1 Anthony Minessale : >> > 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 >> > wrote: >> >> >> >> My basic functionality is this: A calls in, is bridged to B (1111). I >> >> use bind_meta_app to let B rebridge A to C (2222). 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: >> >> >> >> ? >> >> ? ? >> >> ? ? ? >> >> ? ? ? ? >> >> ? ? ? ? >> >> ? ? ? ? >> >> ? ? ? ? >> >> ? ? ? ? >> >> ? ? ? >> >> ? ? >> >> ? >> >> >> >> The first bridge is fine, and B can press *2 to bridge to C/2222. 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:2222 RTP RECV DTMF >> >> *:2000 >> >> 2009-09-01 11:15:00.118195 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF >> >> 1:2000 >> >> 2009-09-01 11:15:00.118195 [DEBUG] switch_ivr_async.c:1725 >> >> sofia/external/unknown at 129.142.224.250 Processing meta digit '2' >> >> [bridge::sofia/gateway/gw1/1111] >> >> 2009-09-01 11:15:00.118195 [DEBUG] switch_core_session.c:813 Send >> >> signal sofia/external/unknown at 129.142.224.250 [BREAK] >> >> >> >> Any ideas? >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at 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_minessale at hotmail.com >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > iax:guest at conference.freeswitch.org/888 >> > googletalk:conf+888 at conference.freeswitch.org >> > pstn:213-799-1400 >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Wed Sep 2 08:42:10 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Sep 2009 10:42:10 -0500 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <20090828001050.GA8108@jdc.jasonjgw.net> <86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com> <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org> <191c3a030909020811q6c04518au514fe8e9d6c8734e@mail.gmail.com> <191c3a030909020815k6006f96fh3df8512018f9b28d@mail.gmail.com> Message-ID: <191c3a030909020842j61369a45va923080ea0a88b40@mail.gmail.com> i mean valgrind is very intensive so you must run very slow 1-5cps yes if you have a version that only has log-file you can use that. if you find me on irc and send me the credentials privately I will examine your box for you. On Wed, Sep 2, 2009 at 10:26 AM, Benedikt Fraunhofer < fraunhofer.lists.freeswitch-001 at traced.net> wrote: > Hello Anthony, > > 2009/9/2 Anthony Minessale : > > run it slower and make sure it shuts down clean. > > i already reduced load 37% but that didnt help, now i'm down to 25% > and it's running. > > > valgrind --tool=memcheck --log-file-exactly=vg.log --leak-check=full > > --leak-resolution=high --show-reachable=yes /path/to/freeswitch -vg > > that's the line in the wiki? I used that but "log-file-exactly" is not > available in my valgrind-3.3.0-Debian, just "log-file". Does this make > any difference? > > Also it wont shut down cleanly but oom-coredump (some assertion > failed) and if i stop the loadgen after some time and try to 'fsctl > shutdown' it wont cleanly bring it down, too. (see first post, last > line printed is > > switch_core_memory.c:567 Stopping memory pool queue. > ) > > Thx. > > Beni. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/c6668b9a/attachment.html From dome at tel.co.th Wed Sep 2 08:43:15 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Wed, 2 Sep 2009 22:43:15 +0700 Subject: [Freeswitch-users] outbould PHP ESL In-Reply-To: <87f2f3b90909020825x516c2054l7a54dc6a1b61691c@mail.gmail.com> References: <8ccbff060909020539n7931ca6av6f7d9bab5ee9cdf4@mail.gmail.com> <4A9E7022.4000800@telemaque.fr> <8ccbff060909020756t167cd564qde77e849227a4f7@mail.gmail.com> <87f2f3b90909020825x516c2054l7a54dc6a1b61691c@mail.gmail.com> Message-ID: <8ccbff060909020843v425cea9ey45fce1d56154fbf5@mail.gmail.com> 2009/9/2 Michael Collins : > Are you trying to get a channel variable or capture DTMF input from the > caller? i try to make IVR by php outbound socket. in XML dialplan we can get DTMF by read application (store in channel variable) I found it's success in perl outbound (IVR.pm) but for php how do i ? Dome C. > -MC > > On Wed, Sep 2, 2009 at 7:56 AM, Dome Charoenyost wrote: >> >> I follow >> http://wiki.freeswitch.org/wiki/PHP_ESL#ivrd >> >> how to get from php ? >> >> >> Dome C. >> ------------------------------------------------------ >> #!/usr/bin/php -q >> >> > >> // set a couple of things so we dont kill the system >> ob_implicit_flush(true); >> set_time_limit(30); >> >> // Open stdin so we can read the AGI data in >> $in = fopen("php://stdin", "r"); >> >> // Connect >> echo "connect\n\n"; >> >> // Answer >> echo "sendmsg\n"; >> echo "call-command: execute\n"; >> echo "execute-app-name: answer\n\n"; >> >> ?echo "sendmsg\n"; >> ?echo "call-command: execute\n"; >> ?echo "execute-app-name: read\n"; >> ?echo "execute-app-arg: 0 20 >> /opt/freeswitch/sounds/th/tuxza/welcome.wav ?res 5000 #\n\n"; >> >> // Wait >> sleep(5); >> >> // Hangup >> echo "sendmsg\n"; >> echo "call-command: hangup\n\n"; >> >> fclose($in); >> >> ?> >> >> >> 2009/9/2 Brian West : >> > uuid_getvar >> > >> > /b >> > >> > On Sep 2, 2009, at 8:16 AM, Tristan Mah? wrote: >> > >> >> Hi, >> >> >> >> just a fast 2cent: >> >> >> >> get var via channel status ? ( variable_res ) >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From math.parent at gmail.com Wed Sep 2 08:47:33 2009 From: math.parent at gmail.com (Mathieu Parent) Date: Wed, 2 Sep 2009 17:47:33 +0200 Subject: [Freeswitch-users] Run a command on event In-Reply-To: <86a32abc0909020746n1250604ds3ebaf3a3d7abdd21@mail.gmail.com> References: <960738410909020727q3286bce9q2ff7d4edfcf6e025@mail.gmail.com> <86a32abc0909020746n1250604ds3ebaf3a3d7abdd21@mail.gmail.com> Message-ID: <960738410909020847u2562a1bei6cedba26e5b06cb6@mail.gmail.com> Hi, On Wed, Sep 2, 2009 at 4:37 PM, Brian West wrote: > Using esl + perl you could do it. > > /b > On Wed, Sep 2, 2009 at 4:46 PM, Diego Viola wrote: > Or you can do something like this with Ruby + FSR: > > http://pastebin.freeswitch.org/10184 > thanks for your suggestions, but this also requires a daemon (standalone or via inetd). I will probably use inetd if nobody suggests a better solution. creating a mod_event_* ? Mathieu From eric at newvo.com Wed Sep 2 08:25:30 2009 From: eric at newvo.com (Eric Richmond) Date: Wed, 2 Sep 2009 11:25:30 -0400 Subject: [Freeswitch-users] E1 Sangoma Card Message-ID: <20827369-2717-4755-83D0-68CDE74E1416@newvo.com> Hello all, I'm wondering if anyone has successfully deployed a freeswitch server with an E1 card attached. By looking at the wiki and chatting in IRC, I've heard that theoretically a FS + Sangoma E1 card solution should work, but I'd really like to hear from someone who actually has it working, so I know what gotchas I might encounter. Any insight into this would be great. Thanks, -Eric Richmond From eric at newvo.com Wed Sep 2 08:52:02 2009 From: eric at newvo.com (Eric Richmond) Date: Wed, 2 Sep 2009 11:52:02 -0400 Subject: [Freeswitch-users] E1 Sangoma Card Message-ID: <4F533E03-7F46-4B33-9EC2-93D8D5FCE492@newvo.com> (Sorry if this has been posted before, or even 2x. I sent it out the first time with the wrong email address, and the 2nd time before i confirmed I wanted to be on the list, so my assumption is that it hasn't gone out yet, although I might be wrong, and if it has gone out before, I'm sorry for spamming the list. It won't happen in the future.) Hello all, I'm wondering if anyone has successfully deployed a freeswitch server with an E1 card attached. By looking at the wiki and chatting in IRC, I've heard that theoretically a FS + Sangoma E1 card solution should work, but I'd really like to hear from someone who actually has it working, so I know what gotchas I might encounter. Any insight into this would be great. Thanks, -Eric Richmond From mitul at enterux.com Wed Sep 2 09:32:35 2009 From: mitul at enterux.com (Mitul Limbani) Date: Wed, 2 Sep 2009 22:02:35 +0530 Subject: [Freeswitch-users] E1 Sangoma Card In-Reply-To: <4F533E03-7F46-4B33-9EC2-93D8D5FCE492@newvo.com> References: <4F533E03-7F46-4B33-9EC2-93D8D5FCE492@newvo.com> Message-ID: <3C1FEF78-AA0C-49FE-B40C-530DFA6916F7@enterux.com> Eric, We have tried it, and it worked after a few tweaks on our PRI line, using the standard openzap plus wanpipe driver of Sangoma, the wiki on Sangoma site is old, but the one on freeswitch kinda works. Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 02-Sep-2009, at 9:22 PM, Eric Richmond wrote: > (Sorry if this has been posted before, or even 2x. I sent it out the > first time with the wrong email address, and the 2nd time before i > confirmed I wanted to be on the list, so my assumption is that it > hasn't gone out yet, although I might be wrong, and if it has gone out > before, I'm sorry for spamming the list. It won't happen in the > future.) > > Hello all, > > I'm wondering if anyone has successfully deployed a freeswitch server > with an E1 card attached. By looking at the wiki and chatting in IRC, > I've heard that theoretically a FS + Sangoma E1 card solution should > work, but I'd really like to hear from someone who actually has it > working, so I know what gotchas I might encounter. > > Any insight into this would be great. > > Thanks, > > -Eric Richmond > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From moises.silva at gmail.com Wed Sep 2 09:52:29 2009 From: moises.silva at gmail.com (Moises Silva) Date: Wed, 2 Sep 2009 12:52:29 -0400 Subject: [Freeswitch-users] E1 Sangoma Card In-Reply-To: <4F533E03-7F46-4B33-9EC2-93D8D5FCE492@newvo.com> References: <4F533E03-7F46-4B33-9EC2-93D8D5FCE492@newvo.com> Message-ID: On Wed, Sep 2, 2009 at 11:52 AM, Eric Richmond wrote: > Hello all, > > I'm wondering if anyone has successfully deployed a freeswitch server > with an E1 card attached. By looking at the wiki and chatting in IRC, > I've heard that theoretically a FS + Sangoma E1 card solution should > work, but I'd really like to hear from someone who actually has it > working, so I know what gotchas I might encounter. > > Any insight into this would be great. > > I have done several of such setups with E1/PRI and E1/MFCR2, if you find problems just drop by in #openzap or #freeswitch and ping me 'moy'. Also you can post questions in this very same list. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/e5c9b044/attachment-0001.html From eric at newvo.com Wed Sep 2 10:05:48 2009 From: eric at newvo.com (Eric Richmond) Date: Wed, 2 Sep 2009 13:05:48 -0400 Subject: [Freeswitch-users] E1 Sangoma Card In-Reply-To: References: <4F533E03-7F46-4B33-9EC2-93D8D5FCE492@newvo.com> Message-ID: Thanks for the replies guys, its much appreciated. On Sep 2, 2009, at 12:52 PM, Moises Silva wrote: > On Wed, Sep 2, 2009 at 11:52 AM, Eric Richmond wrote: > Hello all, > > I'm wondering if anyone has successfully deployed a freeswitch server > with an E1 card attached. By looking at the wiki and chatting in IRC, > I've heard that theoretically a FS + Sangoma E1 card solution should > work, but I'd really like to hear from someone who actually has it > working, so I know what gotchas I might encounter. > > Any insight into this would be great. > > > I have done several of such setups with E1/PRI and E1/MFCR2, if you > find problems just drop by in #openzap or #freeswitch and ping me > 'moy'. Also you can post questions in this very same list. > > -- > Moises Silva > Software Developer > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON > L3R 9T3 Canada > t. 1 905 474 1990 x 128 | e. moy at sangoma.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/b29c8b2f/attachment.html From msc at freeswitch.org Wed Sep 2 10:29:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Sep 2009 10:29:08 -0700 Subject: [Freeswitch-users] E1 Sangoma Card In-Reply-To: <4F533E03-7F46-4B33-9EC2-93D8D5FCE492@newvo.com> References: <4F533E03-7F46-4B33-9EC2-93D8D5FCE492@newvo.com> Message-ID: <87f2f3b90909021029g3e14b2b4o21c79a020dc6adbe@mail.gmail.com> On Wed, Sep 2, 2009 at 8:52 AM, Eric Richmond wrote: > (Sorry if this has been posted before, or even 2x. I sent it out the > first time with the wrong email address, and the 2nd time before i > confirmed I wanted to be on the list, so my assumption is that it > hasn't gone out yet, although I might be wrong, and if it has gone out > before, I'm sorry for spamming the list. It won't happen in the > future.) > > Hello all, > > I'm wondering if anyone has successfully deployed a freeswitch server > with an E1 card attached. By looking at the wiki and chatting in IRC, > I've heard that theoretically a FS + Sangoma E1 card solution should > work, but I'd really like to hear from someone who actually has it > working, so I know what gotchas I might encounter. > > Any insight into this would be great. > > Thanks, > > -Eric Richmond > Erich, Yes, FS + Sangoma E1 card has been used. Is it safe to assume that you need it for PRI? Moises from Sangoma has definitely used it and I believe we have several people in Europe who are running E1/PRI. Your biggest gotcha is probably which PRI stack to use, so for now I would use the libpri method that the devs have created. More info here: http://wiki.freeswitch.org/wiki/OpenZAP#Adding_libpri_Support Just curious - what's your application? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/b81ca5cc/attachment.html From larclap at yahoo.com Wed Sep 2 10:31:08 2009 From: larclap at yahoo.com (Lars Zeb) Date: Wed, 2 Sep 2009 10:31:08 -0700 Subject: [Freeswitch-users] Error building FreeSWITCH Message-ID: <001201ca2bf3$283d1720$78b74560$@com> I just updated using "svn up" which brought the source to 14741. After running "./configure", I ran "make" and got the following output: making all mod_lua make[5]: swig: Command not found make[5]: *** [mod_lua_wrap.cpp] Error 127 make[4]: *** [all] Error 1 make[3]: *** [mod_lua-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 What did I do wrong? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/606d6a42/attachment.html From Prometheus001 at gmx.net Wed Sep 2 11:14:16 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 02 Sep 2009 20:14:16 +0200 Subject: [Freeswitch-users] Error building FreeSWITCH In-Reply-To: <001201ca2bf3$283d1720$78b74560$@com> References: <001201ca2bf3$283d1720$78b74560$@com> Message-ID: <4A9EB5F8.7040704@gmx.net> I had the same problem. Must have been changed something in lua since this morning. Please install swig. E.g. on Debian sudo apt-get install swig That did it for me. Best regards Peter Lars Zeb schrieb: > > I just updated using ?svn up? which brought the source to 14741. After > running ?./configure?, I ran ?make? and got the following output: > > > > making all mod_lua > > make[5]: swig: Command not found > > make[5]: *** [mod_lua_wrap.cpp] Error 127 > > make[4]: *** [all] Error 1 > > make[3]: *** [mod_lua-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 > > > > What did I do wrong? > > > > Thanks, Lars > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Wed Sep 2 11:34:33 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Sep 2009 13:34:33 -0500 Subject: [Freeswitch-users] Error building FreeSWITCH In-Reply-To: <4A9EB5F8.7040704@gmx.net> References: <001201ca2bf3$283d1720$78b74560$@com> <4A9EB5F8.7040704@gmx.net> Message-ID: <191c3a030909021134w40e6e2acj4941e0c663643e8@mail.gmail.com> yikes, dont swig your own stuff just delete all the files that had merge issues and update again. On Wed, Sep 2, 2009 at 1:14 PM, Peter P GMX wrote: > I had the same problem. Must have been changed something in lua since > this morning. > > Please install swig. > > E.g. on Debian > sudo apt-get install swig > > That did it for me. > > Best regards > Peter > > Lars Zeb schrieb: > > > > I just updated using ?svn up? which brought the source to 14741. After > > running ?./configure?, I ran ?make? and got the following output: > > > > > > > > making all mod_lua > > > > make[5]: swig: Command not found > > > > make[5]: *** [mod_lua_wrap.cpp] Error 127 > > > > make[4]: *** [all] Error 1 > > > > make[3]: *** [mod_lua-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 > > > > > > > > What did I do wrong? > > > > > > > > Thanks, Lars > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/48698ea4/attachment-0001.html From Nick.Lemberger at lkfd.net Wed Sep 2 12:14:19 2009 From: Nick.Lemberger at lkfd.net (Nick Lemberger) Date: Wed, 02 Sep 2009 14:14:19 -0500 Subject: [Freeswitch-users] How can I make more sounds that sound like "Callie"? Message-ID: <4A9E7DD3.2C9A.00FE.0@lkfd.net> I'm guessing she's a Cepstral voice, but can I ask what version, khz and settings the 8khz sounds are recorded with (I ask about the khz because perhaps they were pre-transcoded)? I tried downloading 5.1 from Cepstral but they don't sound at all alike. I'd like to replace a few prompts with words more accurate for my installation but I'm having trouble getting close to the default voice. Best Regards, Nicholas Lemberger Lakefield Communications 920.973.6873 From diego.viola at gmail.com Wed Sep 2 12:21:00 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 2 Sep 2009 19:21:00 +0000 Subject: [Freeswitch-users] E1 Sangoma Card In-Reply-To: <87f2f3b90909021029g3e14b2b4o21c79a020dc6adbe@mail.gmail.com> References: <4F533E03-7F46-4B33-9EC2-93D8D5FCE492@newvo.com> <87f2f3b90909021029g3e14b2b4o21c79a020dc6adbe@mail.gmail.com> Message-ID: <86a32abc0909021221n59583210tbb34f501ed16c985@mail.gmail.com> Good to know, I am getting Sangoma cards for my FreeSWITCH as well. Diego On Wed, Sep 2, 2009 at 5:29 PM, Michael Collins wrote: > > > On Wed, Sep 2, 2009 at 8:52 AM, Eric Richmond wrote: > >> (Sorry if this has been posted before, or even 2x. I sent it out the >> first time with the wrong email address, and the 2nd time before i >> confirmed I wanted to be on the list, so my assumption is that it >> hasn't gone out yet, although I might be wrong, and if it has gone out >> before, I'm sorry for spamming the list. It won't happen in the >> future.) >> >> Hello all, >> >> I'm wondering if anyone has successfully deployed a freeswitch server >> with an E1 card attached. By looking at the wiki and chatting in IRC, >> I've heard that theoretically a FS + Sangoma E1 card solution should >> work, but I'd really like to hear from someone who actually has it >> working, so I know what gotchas I might encounter. >> >> Any insight into this would be great. >> >> Thanks, >> >> -Eric Richmond >> > > Erich, > > Yes, FS + Sangoma E1 card has been used. Is it safe to assume that you need > it for PRI? Moises from Sangoma has definitely used it and I believe we have > several people in Europe who are running E1/PRI. Your biggest gotcha is > probably which PRI stack to use, so for now I would use the libpri method > that the devs have created. More info here: > http://wiki.freeswitch.org/wiki/OpenZAP#Adding_libpri_Support > > Just curious - what's your application? > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/cacf1f6e/attachment.html From pjintheusa at gmail.com Wed Sep 2 12:25:14 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Wed, 2 Sep 2009 15:25:14 -0400 Subject: [Freeswitch-users] How can I make more sounds that sound like "Callie"? In-Reply-To: <4A9E7DD3.2C9A.00FE.0@lkfd.net> References: <4A9E7DD3.2C9A.00FE.0@lkfd.net> Message-ID: <367751820909021225s5e616c37t7eb933857f7739c0@mail.gmail.com> Taken from a similar recent posting: "Callie" is one of the voices from GM Voices. She is definitely available for custom work. Visit www.gmvoices.com for more info. Tell them that the FreeSWITCH project sent you. :) MC On Wed, Sep 2, 2009 at 3:14 PM, Nick Lemberger wrote: > I'm guessing she's a Cepstral voice, but can I ask what version, khz and > settings the 8khz sounds are recorded with (I ask about the khz because > perhaps they were pre-transcoded)? I tried downloading 5.1 from Cepstral > but they don't sound at all alike. I'd like to replace a few prompts with > words more accurate for my installation but I'm having trouble getting close > to the default voice. > > Best Regards, > Nicholas Lemberger > Lakefield Communications > 920.973.6873 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/67146492/attachment.html From csa at nowthor.com Wed Sep 2 12:27:57 2009 From: csa at nowthor.com (Carlos S. Antunes) Date: Wed, 02 Sep 2009 15:27:57 -0400 Subject: [Freeswitch-users] How can I make more sounds that sound like "Callie"? In-Reply-To: <4A9E7DD3.2C9A.00FE.0@lkfd.net> References: <4A9E7DD3.2C9A.00FE.0@lkfd.net> Message-ID: <4A9EC73D.5090206@nowthor.com> http://www.gmvoices.com She appears to be a real person! :) Nick Lemberger wrote: > I'm guessing she's a Cepstral voice, but can I ask what version, khz and settings the 8khz sounds are recorded with (I ask about the khz because perhaps they were pre-transcoded)? I tried downloading 5.1 from Cepstral but they don't sound at all alike. I'd like to replace a few prompts with words more accurate for my installation but I'm having trouble getting close to the default voice. > From msc at freeswitch.org Wed Sep 2 12:32:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 2 Sep 2009 12:32:08 -0700 Subject: [Freeswitch-users] How can I make more sounds that sound like "Callie"? In-Reply-To: <4A9E7DD3.2C9A.00FE.0@lkfd.net> References: <4A9E7DD3.2C9A.00FE.0@lkfd.net> Message-ID: <87f2f3b90909021232v5ad9c3f1v4e157f0881077ec3@mail.gmail.com> On Wed, Sep 2, 2009 at 12:14 PM, Nick Lemberger wrote: > I'm guessing she's a Cepstral voice, but can I ask what version, khz and > settings the 8khz sounds are recorded with (I ask about the khz because > perhaps they were pre-transcoded)? I tried downloading 5.1 from Cepstral > but they don't sound at all alike. I'd like to replace a few prompts with > words more accurate for my installation but I'm having trouble getting close > to the default voice. > I suppose how "close" is close enough? In any case, Cepstral has several female voices, so choose the one that you like best. FTR, we have GM record at 48k and then we downsample for 32, 16, and 8. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/91cd680f/attachment.html From tculjaga at gmail.com Wed Sep 2 12:32:31 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 2 Sep 2009 21:32:31 +0200 Subject: [Freeswitch-users] T38 <> T30 transcoding Message-ID: <65d96fc80909021232n6784ef42ra1c6e6648109c059@mail.gmail.com> Hi guys, just a quick question... is it possible to do a reliable on the fly T30 <> T38 transcoding at all ... what is the status of T.38 on FS ? T, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/2060afab/attachment.html From gerry at pstn2.net Wed Sep 2 12:32:32 2009 From: gerry at pstn2.net (Gerry Hull) Date: Wed, 2 Sep 2009 15:32:32 -0400 Subject: [Freeswitch-users] FS performance under windows In-Reply-To: References: <892489.91947.qm@web33505.mail.mud.yahoo.com> <91609A0C-AA80-4707-A62B-6FF129204DC7@freeswitch.org> Message-ID: <98a86adf0909021232p38bb0e87lf9ab39647334ab94@mail.gmail.com> I have a production application where I use FS as part of small, custom ACD solution, with about 80 incoming DIDs and 4 agent positions. It's been deployed for about 4 months now, and was in beta long before that... So far, excellent perfomrance on Windows 2003 server, 32-bit, with 4GB of memory. It's certainly not a heavy load, but it proves the stabiliy and versitility of the platform. Brian said it best -- there is just not many of us on Windows. However, I DEEPLY appreciate the support of the FS dev team, and hope they will continue to support Windows. BTW, if I were to build a high-volume app, I'd do it on CentOS also. Gerry On Wed, Sep 2, 2009 at 11:13 AM, Raffaele P. Guidi < raffaele.p.guidi at gmail.com> wrote: > I'm planning to deploy on windows a small call center (around 50 people) > and willing to help anyhow. I will be able to test on machines mounting > windows 2003 server. Is there a standard test that could be employed to > correctly benchmark the results? > > On Wed, Sep 2, 2009 at 15:50, Brian West wrote: > >> I know people that have deployed on windows... not a huge problem just >> hasn't been load tested like linux... we don't have the resources or time to >> load test every single platform, tune and tweak it. The community can help >> out with this area a lot. >> /b >> >> On Sep 2, 2009, at 8:01 AM, Diego Toro wrote: >> >> What is the reason for saying this? Perhaps the effort of the >> development group of FS has been wasted trying to support Windows as a >> platform for production systems? >> >> Diego >> http://lacarretade.blogspot.com/ >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/4db5277c/attachment-0001.html From pjintheusa at gmail.com Wed Sep 2 12:43:00 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Wed, 2 Sep 2009 15:43:00 -0400 Subject: [Freeswitch-users] 2009-09-02 15:21:44.665608 [CRIT] switch_time.c:454 Reverse Clock Skew Detected! Message-ID: <367751820909021243q45cda27cp9d7c1d048ed85a80@mail.gmail.com> Hi there, Can anyone give any insight to this following message: 2009-09-02 15:21:44.665608 [CRIT] switch_time.c:454 Reverse Clock Skew Detected! This is on a WIN2003 machine with the last call hangup exactly 20 minutes and 20 seconds earlier. Just wondering how CRITICAL this really is? Many thanks Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/637a4f97/attachment.html From brian at freeswitch.org Wed Sep 2 12:49:59 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Sep 2009 14:49:59 -0500 Subject: [Freeswitch-users] How can I make more sounds that sound like "Callie"? In-Reply-To: <4A9EC73D.5090206@nowthor.com> References: <4A9E7DD3.2C9A.00FE.0@lkfd.net> <4A9EC73D.5090206@nowthor.com> Message-ID: <96850F47-6C68-46DD-9B5D-120C76B2E475@freeswitch.org> Her real name is Katherine /b On Sep 2, 2009, at 2:27 PM, Carlos S. Antunes wrote: > http://www.gmvoices.com From brian at freeswitch.org Wed Sep 2 12:52:33 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 2 Sep 2009 14:52:33 -0500 Subject: [Freeswitch-users] 2009-09-02 15:21:44.665608 [CRIT] switch_time.c:454 Reverse Clock Skew Detected! In-Reply-To: <367751820909021243q45cda27cp9d7c1d048ed85a80@mail.gmail.com> References: <367751820909021243q45cda27cp9d7c1d048ed85a80@mail.gmail.com> Message-ID: <030E28A7-DE5A-4499-B6BA-1EFDA2046789@freeswitch.org> well your clock shouldn't be going back in time... that is unless you have figured out time travel or passed thru some star trekish temporal wake. For the most part its a harmless warning unless its happening every second or so. /b On Sep 2, 2009, at 2:43 PM, Phillip Jones wrote: > Hi there, > > Can anyone give any insight to this following message: > > 2009-09-02 15:21:44.665608 [CRIT] switch_time.c:454 Reverse Clock > Skew Detected! > > This is on a WIN2003 machine with the last call hangup exactly 20 > minutes and 20 seconds earlier. > > Just wondering how CRITICAL this really is? > > Many thanks > > > Phillip Jones From oseslija at gmail.com Wed Sep 2 13:09:43 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Wed, 2 Sep 2009 22:09:43 +0200 Subject: [Freeswitch-users] E1 Sangoma Card In-Reply-To: <87f2f3b90909021029g3e14b2b4o21c79a020dc6adbe@mail.gmail.com> References: <4F533E03-7F46-4B33-9EC2-93D8D5FCE492@newvo.com> <87f2f3b90909021029g3e14b2b4o21c79a020dc6adbe@mail.gmail.com> Message-ID: <4468a6770909021309o3e7b5d26p25dbf02c46781747@mail.gmail.com> Hello Eric, I just interconnected FS/openzap and Panasonic via E1 PRI trunk (euroisdn) using libpri stack, and it works just fine. Feel free to drop on irc for any help. Regards, Ognjen On Wed, Sep 2, 2009 at 7:29 PM, Michael Collins wrote: > > > On Wed, Sep 2, 2009 at 8:52 AM, Eric Richmond wrote: > >> (Sorry if this has been posted before, or even 2x. I sent it out the >> first time with the wrong email address, and the 2nd time before i >> confirmed I wanted to be on the list, so my assumption is that it >> hasn't gone out yet, although I might be wrong, and if it has gone out >> before, I'm sorry for spamming the list. It won't happen in the >> future.) >> >> Hello all, >> >> I'm wondering if anyone has successfully deployed a freeswitch server >> with an E1 card attached. By looking at the wiki and chatting in IRC, >> I've heard that theoretically a FS + Sangoma E1 card solution should >> work, but I'd really like to hear from someone who actually has it >> working, so I know what gotchas I might encounter. >> >> Any insight into this would be great. >> >> Thanks, >> >> -Eric Richmond >> > > Erich, > > Yes, FS + Sangoma E1 card has been used. Is it safe to assume that you need > it for PRI? Moises from Sangoma has definitely used it and I believe we have > several people in Europe who are running E1/PRI. Your biggest gotcha is > probably which PRI stack to use, so for now I would use the libpri method > that the devs have created. More info here: > http://wiki.freeswitch.org/wiki/OpenZAP#Adding_libpri_Support > > Just curious - what's your application? > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/fe541fd5/attachment.html From Tim.Meade at millicorp.com Wed Sep 2 13:35:35 2009 From: Tim.Meade at millicorp.com (Tim Meade) Date: Wed, 2 Sep 2009 16:35:35 -0400 Subject: [Freeswitch-users] T38 <> T30 transcoding In-Reply-To: <65d96fc80909021232n6784ef42ra1c6e6648109c059@mail.gmail.com> References: <65d96fc80909021232n6784ef42ra1c6e6648109c059@mail.gmail.com> Message-ID: I am very interested in a response to this. Last I knew there was only T.38 pass through and for some reason I'm not even sure it that was fully implemented. Tim From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tihomir Culjaga Sent: Wednesday, September 02, 2009 3:33 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] T38 <> T30 transcoding Hi guys, just a quick question... is it possible to do a reliable on the fly T30 <> T38 transcoding at all ... what is the status of T.38 on FS ? T, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/af9b724e/attachment.html From tculjaga at gmail.com Wed Sep 2 13:53:49 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 2 Sep 2009 22:53:49 +0200 Subject: [Freeswitch-users] T38 <> T30 transcoding In-Reply-To: References: <65d96fc80909021232n6784ef42ra1c6e6648109c059@mail.gmail.com> Message-ID: <65d96fc80909021353t67699eb8o770b3fa1e364ef98@mail.gmail.com> I will put several nickels saying it is impossible :) seriously, can it be done? T. On Wed, Sep 2, 2009 at 10:35 PM, Tim Meade wrote: > I am very interested in a response to this. Last I knew there was only > T.38 pass through and for some reason I'm not even sure it that was fully > implemented. > > > > Tim > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir > Culjaga > *Sent:* Wednesday, September 02, 2009 3:33 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] T38 <> T30 transcoding > > > > Hi guys, > > just a quick question... is it possible to do a reliable on the fly T30 <> > T38 transcoding at all ... what is the status of T.38 on FS ? > > T, > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/aa6d8134/attachment.html From eric at newvo.com Wed Sep 2 18:45:24 2009 From: eric at newvo.com (Eric Richmond) Date: Wed, 2 Sep 2009 21:45:24 -0400 Subject: [Freeswitch-users] E1 Sangoma Card In-Reply-To: <87f2f3b90909021029g3e14b2b4o21c79a020dc6adbe@mail.gmail.com> References: <4F533E03-7F46-4B33-9EC2-93D8D5FCE492@newvo.com> <87f2f3b90909021029g3e14b2b4o21c79a020dc6adbe@mail.gmail.com> Message-ID: <344DC869-99F0-479A-BD1D-5D7D734CA500@newvo.com> Hey guys, Thanks for all the great replies. After seeing all the people who've gotten it working and configured, I feel pretty confident that if we go this path, we'll be successful. I can't say too much about the app, but in essence we just need to take in traffic over a E1 connection and convert that over to a SIP connection. -Eric On Sep 2, 2009, at 1:29 PM, Michael Collins wrote: > > > On Wed, Sep 2, 2009 at 8:52 AM, Eric Richmond wrote: > (Sorry if this has been posted before, or even 2x. I sent it out the > first time with the wrong email address, and the 2nd time before i > confirmed I wanted to be on the list, so my assumption is that it > hasn't gone out yet, although I might be wrong, and if it has gone out > before, I'm sorry for spamming the list. It won't happen in the > future.) > > Hello all, > > I'm wondering if anyone has successfully deployed a freeswitch server > with an E1 card attached. By looking at the wiki and chatting in IRC, > I've heard that theoretically a FS + Sangoma E1 card solution should > work, but I'd really like to hear from someone who actually has it > working, so I know what gotchas I might encounter. > > Any insight into this would be great. > > Thanks, > > -Eric Richmond > > Erich, > > Yes, FS + Sangoma E1 card has been used. Is it safe to assume that > you need it for PRI? Moises from Sangoma has definitely used it and > I believe we have several people in Europe who are running E1/PRI. > Your biggest gotcha is probably which PRI stack to use, so for now I > would use the libpri method that the devs have created. More info > here: > http://wiki.freeswitch.org/wiki/OpenZAP#Adding_libpri_Support > > Just curious - what's your application? > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/d3598ef9/attachment-0001.html From mayamatakeshi at gmail.com Wed Sep 2 19:17:57 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Thu, 3 Sep 2009 11:17:57 +0900 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> Message-ID: <15b9404e0909021917x4d1e46b2l7823578bce795afd@mail.gmail.com> On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi wrote: > Hello, > I'm testing FS support for the header Path (FS is behind opensips). > It pretty much works: I tested calling from one user to the other and calls > work perfectly. > However, I've noticed that when I register my terminal directly with FS > without going thru the proxy, I receive an unsolicited NOTIFY containing > Message-Waiting information. But when I register via proxy, FS doesn't send > this NOTIFY. > What could be causing this difference of behavior? (enabling debug (F8) > doesn't show anything for registration handling). > I have just updated to trunk Revision 14729 and this behavior persists. > > Also, is it possible to disable this unsolicited MWI notification? (I want > to send it only to subscribed terminals). > OK. I got it this is a bad idea. Now I understand this unsolicited MWI notification is due to an implicit subscription and so, it is a good thing because we will reduce load by not having to deal with separate SUBSCRIBE requests. However I have yet to figure out why this NOTIFY is not sent when the REGISTER is relayed by a proxy. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/b8d9f71b/attachment.html From anthony.minessale at gmail.com Wed Sep 2 20:05:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 2 Sep 2009 22:05:40 -0500 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <15b9404e0909021917x4d1e46b2l7823578bce795afd@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909021917x4d1e46b2l7823578bce795afd@mail.gmail.com> Message-ID: <191c3a030909022005q54ce419au99731f064db8cf63@mail.gmail.com> Its because many phones cheat and just expect mwi without asking for it so we send one on register. There is an opt to disable it I think but I can't recall what it is atm On Sep 2, 2009 9:20 PM, "mayamatakeshi" wrote: On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi wrote: > > Hello, > I'm test... OK. I got it this is a bad idea. Now I understand this unsolicited MWI notification is due to an implicit subscription and so, it is a good thing because we will reduce load by not having to deal with separate SUBSCRIBE requests. However I have yet to figure out why this NOTIFY is not sent when the REGISTER is relayed by a proxy. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090902/336fdeae/attachment.html From mrene_lists at avgs.ca Wed Sep 2 21:52:22 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 2 Sep 2009 21:52:22 -0700 Subject: [Freeswitch-users] Run a command on event In-Reply-To: <960738410909020847u2562a1bei6cedba26e5b06cb6@mail.gmail.com> References: <960738410909020727q3286bce9q2ff7d4edfcf6e025@mail.gmail.com> <86a32abc0909020746n1250604ds3ebaf3a3d7abdd21@mail.gmail.com> <960738410909020847u2562a1bei6cedba26e5b06cb6@mail.gmail.com> Message-ID: <88D1D05C-B36D-4E7B-9DFC-58688DFD3467@avgs.ca> See http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Subscribing_to_events Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 2-Sep-09, at 8:47 AM, Mathieu Parent wrote: > Hi, > > On Wed, Sep 2, 2009 at 4:37 PM, Brian West > wrote: >> Using esl + perl you could do it. >> >> /b >> > > On Wed, Sep 2, 2009 at 4:46 PM, Diego Viola > wrote: >> Or you can do something like this with Ruby + FSR: >> >> http://pastebin.freeswitch.org/10184 >> > > thanks for your suggestions, but this also requires a daemon > (standalone or via inetd). > > I will probably use inetd if nobody suggests a better solution. > creating a mod_event_* ? > > Mathieu > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kawarod at laposte.net Wed Sep 2 23:50:45 2009 From: kawarod at laposte.net (rod) Date: Thu, 03 Sep 2009 10:50:45 +0400 Subject: [Freeswitch-users] Set disable-transcoding in dialplan In-Reply-To: <87f2f3b90908310938oa7e838dkd6e07579f224de37@mail.gmail.com> References: <4A9BE5AB.4030304@laposte.net> <87f2f3b90908310938oa7e838dkd6e07579f224de37@mail.gmail.com> Message-ID: <4A9F6745.6030607@laposte.net> Hi Michael, I did some tests but I haven't been successful, so there is what I'm trying to achieve: On A leg, my phone is using: PCMA and G729 (in this priority order) With PEER A, I want to use only G729 (thats is the only codec that this PEER support), so that the RTP flow will be: Phone-----G729----FS-----G729-----PEER_A With PEER B, I want to use only G711, so: Phone-----G711----FS-----G711-----PEER_B In fact, I'd like to force FS announcing the codec list priority based on the priority of the codec announced by the PEER, cause FS is unable to transcode G729 <--> G711. Tried a lot of things (greedy for codec-negociation, late_codec, disable_transcoding, codec-prefs) without success. If you have some clue. regards, rod Michael Collins a ?crit : > Check out this page: > http://wiki.freeswitch.org/wiki/Codec_negotiation > > Late negotiation will probably let you handle all the cases you need. > -MC > > On Mon, Aug 31, 2009 at 8:00 AM, rod > wrote: > > Hi all, > > I'm wondering if I can do something like this: > - in my internal profile, I have this because of some PEER > using G729: > - > > But for a specific PEER, I'd like to activate transcoding: > - for this PEER, only G711 is used > - I'd like to transcode DTMF SIP INFO or RFC2833 to INBAND > > So in my dialplan, I tried before bridging: > > - > - > > But I still see RFC2833 events between my FS and PEER and the DTMF are > not working. > > So 2 questions: > - does application "start_dtmf_generate" requires transcoding > - if yes, can I set the variable disable-transcoding in my dialplan > > regards, > rod > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From christian.loeschenkohl at xpirio.com Thu Sep 3 00:03:41 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Thu, 03 Sep 2009 09:03:41 +0200 Subject: [Freeswitch-users] conference question In-Reply-To: References: <4A9D79A1.5080107@xpirio.com> <7bcfdd290909011432k2e806cc5if57dd612c6825cf@mail.gmail.com> <4A9D9875.1090707@xpirio.com> <191c3a030909011552h441981cla53dae8a439f748c@mail.gmail.com> Message-ID: <4A9F6A4D.9020108@xpirio.com> thank you but we defined the conference with so no keys should be available br On 2009-09-02 17:02, Andy Spitzer wrote: > Woof! > > On Tue, 01 Sep 2009 18:52:01 -0400, Anthony Minessale wrote: > >> 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. > > Perhaps some listeners are hitting the "unmute" DTMF key? > > --Woof! > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com From christian.loeschenkohl at xpirio.com Thu Sep 3 00:29:43 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Thu, 03 Sep 2009 09:29:43 +0200 Subject: [Freeswitch-users] stability problems Message-ID: <4A9F7067.2090306@xpirio.com> hello we have regular (every 4-6 days) stability problems with freeswitch when the problme occurs - no registers are done bythe server (olny 1 ack of the initial register) - no more calls are working - the calls are all ending with a timeout (cdr caues ORIGINATOR_CANCEL) - only a restart of the whole server cures the problem the server doesn't crash or segfault my first try was to enable the crash-protection flag, but with no difference the server is restartet every night and the last stand still was after about 15h uptime the system is an sun fire 2400 with debian 64 bit system how could i offer you more information to solve this big problem br -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com From helmut.kuper at ewetel.de Thu Sep 3 02:03:02 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 03 Sep 2009 11:03:02 +0200 Subject: [Freeswitch-users] Q931 TE State Timer In-Reply-To: <86a32abc0909020828h429a98a8uabc892c42427469f@mail.gmail.com> References: <4A3266E5.2000702@ewetel.de> <4A328E8E.6030607@freeswitch.org> <4A35F0F0.50406@ewetel.de> <4A643C7D.7010209@ewetel.de> <4A658FDA.8080908@ewetel.de> <4A9E76A5.7030200@ewetel.de> <86a32abc0909020828h429a98a8uabc892c42427469f@mail.gmail.com> Message-ID: <4A9F8646.7010509@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Diege, the flowers has to go to Stefan Knoblich not to me ;) He did the coding! regards Helmut On 02.09.2009 17:28, Diego Viola wrote: > Great stuff, thanks for your hard work :). > > Keep it up. > > Regards, > > Diego -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKn4ZG4tZeNddg3dwRAt63AJ4kPRXMzCVWEZhQ+DLHePytAunGLgCeOY45 f56lUcvOv5q4jsc/Gjljt9I= =860q -----END PGP SIGNATURE----- From bruce.mcalister at blueface.ie Thu Sep 3 03:42:50 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Thu, 03 Sep 2009 11:42:50 +0100 Subject: [Freeswitch-users] "make install" failure on Solaris 10 Message-ID: <4A9F9DAA.8060702@blueface.ie> Hi, I have just managed to complete a build of FreeSWITCH 1.0.4 on Solaris 10. The problem I am now having is that it fails on the "make install" part of the installation. I have attached the complete output of the "make install". A snippet of the failure is below: --- Installing freeswitch *** Error code 1 The following command caused the error: for htdocsfile in `find htdocs -name \* | grep -v .svn` ; do \ dir=`echo $htdocsfile | sed -e 's|/[^/]*$||'`; \ filename=`echo $htdocsfile | sed -e 's|^.*/||'`; \ test -d /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/$dir || /export/home/user/packages/BUILD/freeswitch-1.0.4/build/config/install-sh -d /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/$dir ; \ test -f /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/$dir/$filename || /opt/jdsbld/bin/ginstall -c -m 644 $dir/$filename /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/$dir 2>/dev/null; \ done make: Fatal error: Command failed for target `samples-htdocs' Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4 *** Error code 1 The following command caused the error: test -d /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/htdocs || make samples-htdocs make: Fatal error: Command failed for target `install-data-local' Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4 *** Error code 1 The following command caused the error: make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` install-exec-am install-data-am make: Fatal error: Command failed for target `install-am' Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4 *** Error code 1 The following command caused the error: failcom='exit 1'; \ for f in x $MAKEFLAGS; do \ case $f in \ *=* | --[!k]*);; \ *k*) failcom='fail=yes';; \ esac; \ done; \ dot_seen=no; \ target=`echo install-recursive | sed s/-recursive//`; \ list='. src build'; for subdir in $list; do \ echo "Making $target in $subdir"; \ if test "$subdir" = "."; then \ dot_seen=yes; \ local_target="$target-am"; \ else \ local_target="$target"; \ fi; \ (cd $subdir && make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` $local_target) \ || eval $failcom; \ done; \ if test "$dot_seen" = "no"; then \ make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" || echo -s` "$target-am" || exit 1; \ fi; test -z "$fail" make: Fatal error: Command failed for target `install-recursive' Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4 *** Error code 1 make: Fatal error: Command failed for target `install' --- Any pointers would be greatly appreciated. Thanks Bruce -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: 28169-1.0.4-make-install.log Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/d6ab4e17/attachment-0001.pl From bruce.mcalister at blueface.ie Thu Sep 3 03:45:31 2009 From: bruce.mcalister at blueface.ie (Bruce McAlister) Date: Thu, 03 Sep 2009 11:45:31 +0100 Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 & v1.0.4) In-Reply-To: <4A8BF5B1.2090801@blueface.ie> References: <4A8280B8.6050308@blueface.ie> <4A891472.5060302@blueface.ie> <4A8BF5B1.2090801@blueface.ie> Message-ID: <4A9F9E4B.4060508@blueface.ie> Hi All, With the help of Vladimir I managed to get past this particular issue. I will update the JIRA as well shortly. I managed to get a successful build of FreeSWITCH 1.0.4 with the following environment being set: export 'CFLAGS=-m32 -I/usr/sfw/include -lresolv' export 'CXXFLAGS=-m32 -I/usr/sfw/include -lresolv' export 'LDFLAGS=-m32 -L/usr/sfw/lib -lresolv' Thanks Bruce Bruce McAlister wrote: > Hi All, > > JIRA FSBUILD-186 BugID has been logged for this issue. > > Thanks > Bruce > > Bruce McAlister wrote: >> Hi All, >> >> Shall I log a JIRA for this issue? >> >> Thanks >> Bruce >> >> Bruce McAlister wrote: >>> Hi All, >>> >>> I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and >>> 1.0.4. >>> >>> I am running on Solaris 10 Update 5 on x86 hardware (32-bit). >>> >>> The build fails with: >>> >>> --- snip --- >>> make: Fatal error: Command failed for target `all-recursive' >>> Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4 >>> *** Error code 1 >>> make: Fatal error: Command failed for target `all' >>> --- >>> >>> Looking back through the build I can see the following error: >>> >>> --- snip --- >>> creating libfreeswitch.la >>> (cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la >>> libfreeswitch.la) >>> /usr/bin/cc >>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include >>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src >>> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes >>> -DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch >>> freeswitch-switch.o ./.libs/libfreeswitch.so >>> -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib >>> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib/.libs/libexpat.a >>> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libapr-1.a >>> -lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp >>> -L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt >>> -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl >>> -lcurses -lsocket -R/opt/freeswitch/lib -R/usr/sfw/lib >>> Undefined first referenced >>> symbol in file >>> herror ./.libs/libfreeswitch.so >>> ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch >>> *** Error code 1 >>> The following command caused the error: >>> `if test -z "" ; then echo /bin/bash >>> /export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else >>> echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` >>> --tag=CC --mode=link /usr/bin/cc >>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include >>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src >>> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes >>> -DPATH_MAX=2048 -g -v -Xc -xc99=all -lm -R/opt/freeswitch/lib -o >>> freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib >>> freeswitch-switch.o libfreeswitch.la libs/apr/libapr-1.la >>> libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl >>> -lcrypto -lsocket -lnsl -ldl -lcurses -lsocket >>> --- snip --- >>> >>> Then a little above this error, there is the following warning that is >>> displayed (I'm not sure if it is related): >>> >>> --- snip --- >>> *** Warning: Linking the shared library libfreeswitch.la against the >>> *** static library libs/libedit/src/.libs/libedit.a is not portable! >>> --- snip --- >>> >>> My configure line is as follows: >>> >>> --- >>> ./configure --prefix=/opt/freeswitch >>> --- >>> >>> I have the complete configure and make output if anyone needs them. >>> >>> Any help/pointers would be greatly appreciated. >>> >>> Thanks >>> Bruce >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From irmatov at gmail.com Thu Sep 3 04:24:16 2009 From: irmatov at gmail.com (Timur Irmatov) Date: Thu, 3 Sep 2009 16:24:16 +0500 Subject: [Freeswitch-users] trying mod_erlang_event Message-ID: <241d382f0909030424p5b8ec789y819fa534ae45b9a5@mail.gmail.com> Hi, I have installed FreeSWITCH and mod_erlang_event. Now I'm trying to receive events from FreeSWITCH through mod_erlang_event. Simple command like 'api status' works, but I cannot receive events from FreeSWTICH: erl -sname test at localhost Erlang (BEAM) emulator version 5.6.3 [source] [async-threads:0] [kernel-poll:false] Eshell V5.6.3 (abort with ^G) (test at localhost)1> {foo, freeswitch at localhost} ! {api, status, ""}. {api,status,[]} (test at localhost)2> receive X -> X after 10000 -> timeout end. {ok,"UP 0 years, 0 days, 0 hours, 5 minutes, 10 seconds, 886 milliseconds, 421 microseconds\n0 session(s) since startup\n0 session(s) 0/30\n1000 session(s) max\n"} (test at localhost)3> {foo, freeswitch at localhost} ! {event, 'ALL'}. {event,'ALL'} (test at localhost)4> receive Y -> Y after 10000 -> timeout end. ok (test at localhost)5> receive Y -> Y after 10000 -> timeout end. timeout (test at localhost)6> receive Y -> Y after 10000 -> timeout end. timeout (test at localhost)7> Do I miss something obvious? Thanks in advance for any suggestions. -- Timur Irmatov, xmpp:irmatov at jabber.ru From irmatov at gmail.com Thu Sep 3 05:11:40 2009 From: irmatov at gmail.com (Timur Irmatov) Date: Thu, 3 Sep 2009 17:11:40 +0500 Subject: [Freeswitch-users] trying mod_erlang_event In-Reply-To: <241d382f0909030424p5b8ec789y819fa534ae45b9a5@mail.gmail.com> References: <241d382f0909030424p5b8ec789y819fa534ae45b9a5@mail.gmail.com> Message-ID: <241d382f0909030511g372f125y79e43ac2557f0317@mail.gmail.com> On Thu, Sep 3, 2009 at 4:24 PM, Timur Irmatov wrote: > I have installed FreeSWITCH and mod_erlang_event. Now I'm trying to > receive events from FreeSWITCH through mod_erlang_event. Simple > command like 'api status' works, but I cannot receive events from > FreeSWTICH: Ah, I needed to send register_event_handler before trying to receive any events. Sorry for the noise.. -- Timur Irmatov, xmpp:irmatov at jabber.ru From tculjaga at gmail.com Thu Sep 3 05:29:57 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 3 Sep 2009 14:29:57 +0200 Subject: [Freeswitch-users] T38 <> T30 transcoding In-Reply-To: <65d96fc80909021353t67699eb8o770b3fa1e364ef98@mail.gmail.com> References: <65d96fc80909021232n6784ef42ra1c6e6648109c059@mail.gmail.com> <65d96fc80909021353t67699eb8o770b3fa1e364ef98@mail.gmail.com> Message-ID: <65d96fc80909030529x1a60db4eg152a498f8b71ae50@mail.gmail.com> anyone knows anything about this? T. On Wed, Sep 2, 2009 at 10:53 PM, Tihomir Culjaga wrote: > I will put several nickels saying it is impossible :) > > > seriously, can it be done? > > T. > > On Wed, Sep 2, 2009 at 10:35 PM, Tim Meade wrote: > >> I am very interested in a response to this. Last I knew there was only >> T.38 pass through and for some reason I'm not even sure it that was fully >> implemented. >> >> >> >> Tim >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir >> Culjaga >> *Sent:* Wednesday, September 02, 2009 3:33 PM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] T38 <> T30 transcoding >> >> >> >> Hi guys, >> >> just a quick question... is it possible to do a reliable on the fly T30 <> >> T38 transcoding at all ... what is the status of T.38 on FS ? >> >> T, >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/363649cb/attachment.html From enno.egbert at googlemail.com Thu Sep 3 06:08:17 2009 From: enno.egbert at googlemail.com (NOx-WHV) Date: Thu, 3 Sep 2009 06:08:17 -0700 (PDT) Subject: [Freeswitch-users] select batchfile after call Message-ID: <25275633.post@talk.nabble.com> Hi, does anybody have a tip how to start a batchfile after hanging up. After ext. 1000 calls 1001 and hang up, i need a request to call: /../../FS/batchfile 1000 if 1001 calls 1000 i need: /../../FS/batchfile 1001 and so on... Thanks for help -- View this message in context: http://www.nabble.com/select-batchfile-after-call-tp25275633p25275633.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Thu Sep 3 07:12:43 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Sep 2009 09:12:43 -0500 Subject: [Freeswitch-users] stability problems In-Reply-To: <4A9F7067.2090306@xpirio.com> References: <4A9F7067.2090306@xpirio.com> Message-ID: <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> Sounds like you have some build skew... can you tell us how you built FreeSWITCH? /b On Sep 3, 2009, at 2:29 AM, Christian L?schenkohl wrote: > hello > > we have regular (every 4-6 days) stability problems with freeswitch > when the problme occurs > > - no registers are done bythe server (olny 1 ack of the initial > register) > - no more calls are working > - the calls are all ending with a timeout (cdr caues > ORIGINATOR_CANCEL) > - only a restart of the whole server cures the problem > > the server doesn't crash or segfault > my first try was to enable the crash-protection flag, but with no > difference > the server is restartet every night and the last stand still was > after about 15h uptime > > the system is an sun fire 2400 with debian 64 bit system > > how could i offer you more information to solve this big problem > > br > > -- > Ing. Christian L?schenkohl > 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.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From pjintheusa at gmail.com Thu Sep 3 07:26:52 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 3 Sep 2009 10:26:52 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET Message-ID: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> Hi there, mod_managed exposes EventReceivedFunction such that: Session.EventReceivedFunction = (e) => { Log.WriteLine(LogLevel.Alert, "Received Event {0}", e.ToString()); return ""; }; should trap all events to which i subscribe. But how do I subscribe to events? What is the .NET / managed equivalent of: switch_event_bind(const char *id, switch_event_types_t event, const char *subclass_name, switch_event_callback_t callback, void *user_data); Thank you! Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/ff6a6188/attachment.html From pjintheusa at gmail.com Thu Sep 3 07:29:45 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 3 Sep 2009 10:29:45 -0400 Subject: [Freeswitch-users] 2009-09-02 15:21:44.665608 [CRIT] switch_time.c:454 Reverse Clock Skew Detected! In-Reply-To: <030E28A7-DE5A-4499-B6BA-1EFDA2046789@freeswitch.org> References: <367751820909021243q45cda27cp9d7c1d048ed85a80@mail.gmail.com> <030E28A7-DE5A-4499-B6BA-1EFDA2046789@freeswitch.org> Message-ID: <367751820909030729ke289f1ap2665b197f1e5daad@mail.gmail.com> Yeah - temporal worm holes and the like are my other project :) Or may be its the full moon. Thanks for the info. On Wed, Sep 2, 2009 at 3:52 PM, Brian West wrote: > well your clock shouldn't be going back in time... that is unless you > have figured out time travel or passed thru some star trekish temporal > wake. > > For the most part its a harmless warning unless its happening every > second or so. > > /b > > > On Sep 2, 2009, at 2:43 PM, Phillip Jones wrote: > > > Hi there, > > > > Can anyone give any insight to this following message: > > > > 2009-09-02 15:21:44.665608 [CRIT] switch_time.c:454 Reverse Clock > > Skew Detected! > > > > This is on a WIN2003 machine with the last call hangup exactly 20 > > minutes and 20 seconds earlier. > > > > Just wondering how CRITICAL this really is? > > > > Many thanks > > > > > > Phillip Jones > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/abc156b1/attachment-0001.html From christian.loeschenkohl at xpirio.com Thu Sep 3 07:47:10 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Thu, 03 Sep 2009 16:47:10 +0200 Subject: [Freeswitch-users] stability problems In-Reply-To: <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> References: <4A9F7067.2090306@xpirio.com> <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> Message-ID: <4A9FD6EE.7070107@xpirio.com> on debian lenny amd64 with the build-essential package an then with ./configure --prefix=/opt/freeswitch make make install nothing else br On 2009-09-03 16:12, Brian West wrote: > Sounds like you have some build skew... can you tell us how you built > FreeSWITCH? > > /b > > On Sep 3, 2009, at 2:29 AM, Christian L?schenkohl wrote: > >> hello >> >> we have regular (every 4-6 days) stability problems with freeswitch >> when the problme occurs >> >> - no registers are done bythe server (olny 1 ack of the initial >> register) >> - no more calls are working >> - the calls are all ending with a timeout (cdr caues >> ORIGINATOR_CANCEL) >> - only a restart of the whole server cures the problem >> >> the server doesn't crash or segfault >> my first try was to enable the crash-protection flag, but with no >> difference >> the server is restartet every night and the last stand still was >> after about 15h uptime >> >> the system is an sun fire 2400 with debian 64 bit system >> >> how could i offer you more information to solve this big problem >> >> br >> >> -- >> Ing. Christian L?schenkohl >> 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.loeschenkohl at xpirio.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com From msc at freeswitch.org Thu Sep 3 07:47:41 2009 From: msc at freeswitch.org (Michael S Collins) Date: Thu, 3 Sep 2009 07:47:41 -0700 Subject: [Freeswitch-users] select batchfile after call In-Reply-To: <25275633.post@talk.nabble.com> References: <25275633.post@talk.nabble.com> Message-ID: Sent from my iPhone On Sep 3, 2009, at 6:08 AM, NOx-WHV wrote: > > Hi, > > does anybody have a tip how to start a batchfile after hanging up. > > After ext. 1000 calls 1001 and hang up, i need a request to call: > > /../../FS/batchfile 1000 > > if 1001 calls 1000 i need: > > /../../FS/batchfile 1001 > > and so on... > Try something like this in your Dialplan: -MC > > Thanks for help > -- > View this message in context: http://www.nabble.com/select-batchfile-after-call-tp25275633p25275633.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mayamatakeshi at gmail.com Thu Sep 3 07:49:29 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Thu, 3 Sep 2009 23:49:29 +0900 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <191c3a030909022005q54ce419au99731f064db8cf63@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909021917x4d1e46b2l7823578bce795afd@mail.gmail.com> <191c3a030909022005q54ce419au99731f064db8cf63@mail.gmail.com> Message-ID: <15b9404e0909030749t5229d366md0a79bb78f2f27b@mail.gmail.com> On Thu, Sep 3, 2009 at 12:05 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Its because many phones cheat and just expect mwi without asking for it so > we send one on register. > There is an opt to disable it I think but I can't recall what it is atm > Thanks. I've located the option in the mod sofia source: send-message-query-on-register It seems it is not in the wiki. I will update the mod sofia page. > On Sep 2, 2009 9:20 PM, "mayamatakeshi" wrote: > > > > On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi > wrote: > > Hello, > I'm test... > > OK. I got it this is a bad idea. Now I understand this unsolicited MWI > notification is due to an implicit subscription and so, it is a good thing > because we will reduce load by not having to deal with separate SUBSCRIBE > requests. > > However I have yet to figure out why this NOTIFY is not sent when the > REGISTER is relayed by a proxy. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/7b592639/attachment.html From diego.viola at gmail.com Thu Sep 3 07:56:22 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 3 Sep 2009 14:56:22 +0000 Subject: [Freeswitch-users] T38 <> T30 transcoding In-Reply-To: <65d96fc80909030529x1a60db4eg152a498f8b71ae50@mail.gmail.com> References: <65d96fc80909021232n6784ef42ra1c6e6648109c059@mail.gmail.com> <65d96fc80909021353t67699eb8o770b3fa1e364ef98@mail.gmail.com> <65d96fc80909030529x1a60db4eg152a498f8b71ae50@mail.gmail.com> Message-ID: <86a32abc0909030756ra0f9a6et7e8162cdcf07c201@mail.gmail.com> I believe FreeSWITCH doesn't do T.38 other than pass-through yet, there are plans for complete support I think but I'm not sure when. Diego On Thu, Sep 3, 2009 at 12:29 PM, Tihomir Culjaga wrote: > anyone knows anything about this? > > T. > > > On Wed, Sep 2, 2009 at 10:53 PM, Tihomir Culjaga wrote: > >> I will put several nickels saying it is impossible :) >> >> >> seriously, can it be done? >> >> T. >> >> On Wed, Sep 2, 2009 at 10:35 PM, Tim Meade wrote: >> >>> I am very interested in a response to this. Last I knew there was only >>> T.38 pass through and for some reason I'm not even sure it that was fully >>> implemented. >>> >>> >>> >>> Tim >>> >>> >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir >>> Culjaga >>> *Sent:* Wednesday, September 02, 2009 3:33 PM >>> *To:* freeswitch-users at lists.freeswitch.org >>> *Subject:* [Freeswitch-users] T38 <> T30 transcoding >>> >>> >>> >>> Hi guys, >>> >>> just a quick question... is it possible to do a reliable on the fly T30 >>> <> T38 transcoding at all ... what is the status of T.38 on FS ? >>> >>> T, >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/919aece2/attachment.html From anthony.minessale at gmail.com Thu Sep 3 08:02:48 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Sep 2009 10:02:48 -0500 Subject: [Freeswitch-users] stability problems In-Reply-To: <4A9FD6EE.7070107@xpirio.com> References: <4A9F7067.2090306@xpirio.com> <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> <4A9FD6EE.7070107@xpirio.com> Message-ID: <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> Which revision are you using? If you are not running the latest trunk, please upgrade to that in case your problem requires us to change the code we need it to be up to date. 1) Remove any binary files which may get mixed in from an older build rm /usr/local/freeswitch/bin/* rm /usr/local/freeswitch/lib/* rm /usr/local/freeswitch/mod 2) Build Latest Trunk 3) Reproduce the problem. If you get the problem keep FreeSWITCH running and capture a gcore back trace. ./scripts/freeswitch-gcore > gcore.txt Send us the file as an attachment or attached to a new jira issue. http://jira.freeswitch.org 2009/9/3 Christian L?schenkohl > on debian lenny amd64 with the build-essential package > > an then with > > ./configure --prefix=/opt/freeswitch > make > make install > > nothing else > > br > > On 2009-09-03 16:12, Brian West wrote: > > Sounds like you have some build skew... can you tell us how you built > > FreeSWITCH? > > > > /b > > > > On Sep 3, 2009, at 2:29 AM, Christian L?schenkohl wrote: > > > >> hello > >> > >> we have regular (every 4-6 days) stability problems with freeswitch > >> when the problme occurs > >> > >> - no registers are done bythe server (olny 1 ack of the initial > >> register) > >> - no more calls are working > >> - the calls are all ending with a timeout (cdr caues > >> ORIGINATOR_CANCEL) > >> - only a restart of the whole server cures the problem > >> > >> the server doesn't crash or segfault > >> my first try was to enable the crash-protection flag, but with no > >> difference > >> the server is restartet every night and the last stand still was > >> after about 15h uptime > >> > >> the system is an sun fire 2400 with debian 64 bit system > >> > >> how could i offer you more information to solve this big problem > >> > >> br > >> > >> -- > >> Ing. Christian L?schenkohl > >> 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.loeschenkohl at xpirio.com > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > 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.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/77272243/attachment-0001.html From mgg at giagnocavo.net Thu Sep 3 08:05:14 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 3 Sep 2009 11:05:14 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DAFA465A@mse17be1.mse17.exchange.ms> You can call switch_event_bind directly, but I doubt that'll achieve what you want. My guess is that it works the same as LUA or any of the other plugins - whatever higher level API is exposed for setting up events should make things just work. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Thursday, September 03, 2009 8:27 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET Hi there, mod_managed exposes EventReceivedFunction such that: Session.EventReceivedFunction = (e) => { Log.WriteLine(LogLevel.Alert, "Received Event {0}", e.ToString()); return ""; }; should trap all events to which i subscribe. But how do I subscribe to events? What is the .NET / managed equivalent of: switch_event_bind(const char *id, switch_event_types_t event, const char *subclass_name, switch_event_callback_t callback, void *user_data); Thank you! Phillip Jones -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/22ee5a7c/attachment.html From intralanman at freeswitch.org Thu Sep 3 08:08:31 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 3 Sep 2009 11:08:31 -0400 Subject: [Freeswitch-users] stability problems In-Reply-To: <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> References: <4A9F7067.2090306@xpirio.com> <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> <4A9FD6EE.7070107@xpirio.com> <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> Message-ID: > Send us the file as an attachment or attached to a new jira issue. > http://jira.freeswitch.org Please note the words "attached to" and not "pasted inline" :-D Thanks -Ray -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/63fcc381/attachment.html From mrene_lists at avgs.ca Thu Sep 3 08:12:07 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Thu, 3 Sep 2009 08:12:07 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DAFA465A@mse17be1.mse17.exchange.ms> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA465A@mse17be1.mse17.exchange.ms> Message-ID: <46D346CE-4B4D-47C3-8B43-2D2381B5378A@avgs.ca> Check out EventConsumer Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 3-Sep-09, at 8:05 AM, Michael Giagnocavo wrote: > You can call switch_event_bind directly, but I doubt that?ll achieve > what you want. > > My guess is that it works the same as LUA or any of the other > plugins ? whatever higher level API is exposed for setting up events > should make things just work. > > -Michael > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Phillip Jones > Sent: Thursday, September 03, 2009 8:27 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET > > Hi there, > > mod_managed exposes EventReceivedFunction such that: > > Session.EventReceivedFunction = (e) => > { > Log.WriteLine(LogLevel.Alert, "Received Event {0}", > e.ToString()); > return ""; > }; > > should trap all events to which i subscribe. > > > But how do I subscribe to events? What is the .NET / managed > equivalent of: > > switch_event_bind(const char *id, switch_event_types_t event, const > char *subclass_name, switch_event_callback_t callback, void > *user_data); > > > > Thank you! > > > > Phillip Jones > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/2460cc7a/attachment.html From trtr at 2ride.com Thu Sep 3 07:29:10 2009 From: trtr at 2ride.com (T.R. Missner) Date: Thu, 3 Sep 2009 10:29:10 -0400 Subject: [Freeswitch-users] T38 <> T30 transcoding In-Reply-To: <65d96fc80909030529x1a60db4eg152a498f8b71ae50@mail.gmail.com> References: <65d96fc80909021232n6784ef42ra1c6e6648109c059@mail.gmail.com> <65d96fc80909021353t67699eb8o770b3fa1e364ef98@mail.gmail.com> <65d96fc80909030529x1a60db4eg152a498f8b71ae50@mail.gmail.com> Message-ID: <5e757f90909030729g63c63becve6440b7ce042156b@mail.gmail.com> if you buy a sip trunk from bandwidth.com and ask them to put you on the pvp route you will get this functionality On Thu, Sep 3, 2009 at 8:29 AM, Tihomir Culjaga wrote: > anyone knows anything about this? > > T. > > > On Wed, Sep 2, 2009 at 10:53 PM, Tihomir Culjaga wrote: > >> I will put several nickels saying it is impossible :) >> >> >> seriously, can it be done? >> >> T. >> >> On Wed, Sep 2, 2009 at 10:35 PM, Tim Meade wrote: >> >>> I am very interested in a response to this. Last I knew there was only >>> T.38 pass through and for some reason I'm not even sure it that was fully >>> implemented. >>> >>> >>> >>> Tim >>> >>> >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir >>> Culjaga >>> *Sent:* Wednesday, September 02, 2009 3:33 PM >>> *To:* freeswitch-users at lists.freeswitch.org >>> *Subject:* [Freeswitch-users] T38 <> T30 transcoding >>> >>> >>> >>> Hi guys, >>> >>> just a quick question... is it possible to do a reliable on the fly T30 >>> <> T38 transcoding at all ... what is the status of T.38 on FS ? >>> >>> T, >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/57a8725d/attachment-0001.html From pjintheusa at gmail.com Thu Sep 3 08:33:11 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 3 Sep 2009 11:33:11 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <46D346CE-4B4D-47C3-8B43-2D2381B5378A@avgs.ca> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA465A@mse17be1.mse17.exchange.ms> <46D346CE-4B4D-47C3-8B43-2D2381B5378A@avgs.ca> Message-ID: <367751820909030833r2698c8fof72057becf71c190@mail.gmail.com> Where can I find EventConsumer?? I search the wiki and contrib\ Thanks On Thu, Sep 3, 2009 at 11:12 AM, Mathieu Rene wrote: > Check out EventConsumer > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 3-Sep-09, at 8:05 AM, Michael Giagnocavo wrote: > > You can call switch_event_bind directly, but I doubt that?ll achieve what > you want. > > My guess is that it works the same as LUA or any of the other plugins ? > whatever higher level API is exposed for setting up events should make > things just work. > > -Michael > > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Phillip Jones > *Sent:* Thursday, September 03, 2009 8:27 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Subscribing to events in managed C# / .NET > > Hi there, > > mod_managed exposes EventReceivedFunction such that: > > Session.EventReceivedFunction = (e) => > { > Log.WriteLine(LogLevel.Alert, "Received Event {0}", e.ToString()); > return ""; > }; > > should trap all events to which i subscribe. > > > But how do I subscribe to events? What is the .NET / managed equivalent of: > > switch_event_bind(const char *id, switch_event_types_t event, const char > *subclass_name, switch_event_callback_t callback, void *user_data); > > > > Thank you! > > > > Phillip Jones > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/b6577997/attachment.html From rupa at rupa.com Thu Sep 3 08:56:47 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 3 Sep 2009 10:56:47 -0500 Subject: [Freeswitch-users] stability problems In-Reply-To: <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> References: <4A9F7067.2090306@xpirio.com> <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> <4A9FD6EE.7070107@xpirio.com> <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> Message-ID: On Thu, Sep 3, 2009 at 10:02 AM, Anthony Minessale wrote: > 1) Remove any binary files which may get mixed in from an older build > ?? rm /usr/local/freeswitch/bin/* > ?? rm /usr/local/freeswitch/lib/* > ?? rm /usr/local/freeswitch/mod Since you are using the debian package, the files will be in /opt/freeswitch not /usr/local/freeswitch. -- -Rupa From jerry.richards at teotech.com Thu Sep 3 09:21:30 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 3 Sep 2009 09:21:30 -0700 Subject: [Freeswitch-users] Duplicate Extension Registration Message-ID: I submitted this to the dev-list, but maybe it should be in the user-list: Can I register two phones to the same Line-ID? That is, does Freeswitch support a configuration where multiple endpoints have the same extension number, auth-id and password? And if so, do I have control over whether an inbound call causes both to ring or not? Thanks and Best Regards, Jerry From brian at freeswitch.org Thu Sep 3 09:25:10 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Sep 2009 11:25:10 -0500 Subject: [Freeswitch-users] stability problems In-Reply-To: References: <4A9F7067.2090306@xpirio.com> <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> <4A9FD6EE.7070107@xpirio.com> <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> Message-ID: <3194DE12-AB44-4F7D-8BD7-D7171B41A735@freeswitch.org> Can you try NOT using a package? I have a theory that the package has a few optimization flags in it that breaks things. /b On Sep 3, 2009, at 10:56 AM, Rupa Schomaker wrote: > Since you are using the debian package, the files will be in > /opt/freeswitch not /usr/local/freeswitch. From intralanman at freeswitch.org Thu Sep 3 09:35:25 2009 From: intralanman at freeswitch.org (Raymond Chandler) Date: Thu, 3 Sep 2009 12:35:25 -0400 Subject: [Freeswitch-users] Duplicate Extension Registration In-Reply-To: References: Message-ID: yes you can, and yes it should go to -users not -dev just make sure you have multiple-registrations turned on in the profile you want to allow that on Raymond Chandler http://freeswitchsolutions.com http://cluecon.com On Sep 3, 2009, at 12:21 PM, Jerry Richards wrote: > > I submitted this to the dev-list, but maybe it should be in the user- > list: > > Can I register two phones to the same Line-ID? That is, does > Freeswitch > support a configuration where multiple endpoints have the same > extension > number, auth-id and password? And if so, do I have control over > whether an > inbound call causes both to ring or not? > > Thanks and Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From christian.loeschenkohl at xpirio.com Thu Sep 3 09:43:59 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Thu, 03 Sep 2009 18:43:59 +0200 Subject: [Freeswitch-users] stability problems In-Reply-To: <3194DE12-AB44-4F7D-8BD7-D7171B41A735@freeswitch.org> References: <4A9F7067.2090306@xpirio.com> <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> <4A9FD6EE.7070107@xpirio.com> <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> <3194DE12-AB44-4F7D-8BD7-D7171B41A735@freeswitch.org> Message-ID: <4A9FF24F.6000001@xpirio.com> sorry i can not follow you i build everthing from scratch (download source, unpack and build) what i mean with the build-essential package is a debian meta package that contains gcc, make and so on br On 2009-09-03 18:25, Brian West wrote: > Can you try NOT using a package? I have a theory that the package has > a few optimization flags in it that breaks things. > > /b > > On Sep 3, 2009, at 10:56 AM, Rupa Schomaker wrote: > >> Since you are using the debian package, the files will be in >> /opt/freeswitch not /usr/local/freeswitch. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com From brian at freeswitch.org Thu Sep 3 09:50:45 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Sep 2009 11:50:45 -0500 Subject: [Freeswitch-users] stability problems In-Reply-To: <4A9FF24F.6000001@xpirio.com> References: <4A9F7067.2090306@xpirio.com> <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> <4A9FD6EE.7070107@xpirio.com> <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> <3194DE12-AB44-4F7D-8BD7-D7171B41A735@freeswitch.org> <4A9FF24F.6000001@xpirio.com> Message-ID: Please join IRC if you experience the issue again #freeswitch on irc.freenode.net /b On Sep 3, 2009, at 11:43 AM, Christian L?schenkohl wrote: > sorry i can not follow you > i build everthing from scratch (download source, unpack and build) > > what i mean with the build-essential package is a debian meta package > that contains gcc, make and so on > > br From anthony.minessale at gmail.com Thu Sep 3 10:11:51 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Sep 2009 12:11:51 -0500 Subject: [Freeswitch-users] stability problems In-Reply-To: References: <4A9F7067.2090306@xpirio.com> <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> <4A9FD6EE.7070107@xpirio.com> <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> <3194DE12-AB44-4F7D-8BD7-D7171B41A735@freeswitch.org> <4A9FF24F.6000001@xpirio.com> Message-ID: <191c3a030909031011y2d73ed4bsd6c259297f8ccd5e@mail.gmail.com> I already provided exact instructions. On Thu, Sep 3, 2009 at 11:50 AM, Brian West wrote: > Please join IRC if you experience the issue again #freeswitch on > irc.freenode.net > > /b > > > On Sep 3, 2009, at 11:43 AM, Christian L?schenkohl wrote: > > > sorry i can not follow you > > i build everthing from scratch (download source, unpack and build) > > > > what i mean with the build-essential package is a debian meta package > > that contains gcc, make and so on > > > > br > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/d81b2f9d/attachment-0001.html From rupa at rupa.com Thu Sep 3 10:18:04 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Thu, 3 Sep 2009 12:18:04 -0500 Subject: [Freeswitch-users] stability problems In-Reply-To: <4A9FF24F.6000001@xpirio.com> References: <4A9F7067.2090306@xpirio.com> <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> <4A9FD6EE.7070107@xpirio.com> <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> <3194DE12-AB44-4F7D-8BD7-D7171B41A735@freeswitch.org> <4A9FF24F.6000001@xpirio.com> Message-ID: There are two ways to build. One is to use the debian dir that is in package. That builds a package and puts everything in /opt. Is that what you are doing? The other is to do a simple ./configure && make && make install by default that'll put everything in /usr/local 2009/9/3 Christian L?schenkohl : > sorry i can not follow you > i build everthing from scratch (download source, unpack and build) > > what i mean with the build-essential package is a debian meta package > that contains gcc, make and so on > > br > > On 2009-09-03 18:25, Brian West wrote: >> Can you try NOT using a package? ?I have a theory that the package has >> a few optimization flags in it that breaks things. >> >> /b >> >> On Sep 3, 2009, at 10:56 AM, Rupa Schomaker wrote: >> >>> Since you are using the debian package, the files will be in >>> /opt/freeswitch not /usr/local/freeswitch. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > 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.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From Prometheus001 at gmx.net Thu Sep 3 10:31:16 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Thu, 03 Sep 2009 19:31:16 +0200 Subject: [Freeswitch-users] mod_xml_curl.c Oversized file detected [1056100 bytes] Message-ID: <4A9FFD64.2000701@gmx.net> Hello, in a B2BUA scenario we have 2000 defined gateways (defined but not registered yet). When reloading mod_sofia Freeswitch complains about the XML-Curl File size > 1MB and deactivates all gateways: mod_xml_curl.c:121 Oversized file detected [1056100 bytes] Is there any way to overcome this? Currently we have 2000 gateways defined. Finally we will have about 10.000. And we will not be able to reduce the file size below 1 MB. It will become ~ 2-3 MB maybe. Best Regards Peter From jlenk at frontiernet.net Thu Sep 3 10:54:05 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Thu, 3 Sep 2009 12:54:05 -0500 (CDT) Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> Message-ID: <1252000445541-3574945.post@n2.nabble.com> try something like this EventConsumer con = new EventConsumer("all", ""); Event ev = con.pop(0); see lua sample - http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer Phillip Jones-2 wrote: > > Hi there, > > mod_managed exposes EventReceivedFunction such that: > > Session.EventReceivedFunction = (e) => > { > Log.WriteLine(LogLevel.Alert, "Received Event {0}", e.ToString()); > return ""; > }; > > should trap all events to which i subscribe. > > > But how do I subscribe to events? What is the .NET / managed equivalent > of: > > switch_event_bind(const char *id, switch_event_types_t event, const char > *subclass_name, switch_event_callback_t callback, void *user_data); > > > > Thank you! > > > > Phillip Jones > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/Subscribing-to-events-in-managed-C-NET-tp3573619p3574945.html Sent from the freeswitch-users mailing list archive at Nabble.com. From pjintheusa at gmail.com Thu Sep 3 11:05:43 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 3 Sep 2009 14:05:43 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <1252000445541-3574945.post@n2.nabble.com> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <1252000445541-3574945.post@n2.nabble.com> Message-ID: <367751820909031105s13028b09gfddf3435e65ba166@mail.gmail.com> Exactly what I was after - thank you! On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk wrote: > > try something like this > > EventConsumer con = new EventConsumer("all", ""); > Event ev = con.pop(0); > > see lua sample - > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer > > > Phillip Jones-2 wrote: > > > > Hi there, > > > > mod_managed exposes EventReceivedFunction such that: > > > > Session.EventReceivedFunction = (e) => > > { > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", e.ToString()); > > return ""; > > }; > > > > should trap all events to which i subscribe. > > > > > > But how do I subscribe to events? What is the .NET / managed equivalent > > of: > > > > switch_event_bind(const char *id, switch_event_types_t event, const char > > *subclass_name, switch_event_callback_t callback, void *user_data); > > > > > > > > Thank you! > > > > > > > > Phillip Jones > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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/Subscribing-to-events-in-managed-C-NET-tp3573619p3574945.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/d20242ae/attachment.html From mike at jerris.com Thu Sep 3 11:27:59 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 3 Sep 2009 14:27:59 -0400 Subject: [Freeswitch-users] Incorrect method of PHP call control? In-Reply-To: <40C6E7F6-D49A-4902-B3F8-224567325875@cgicommunications.com> References: <4E10B925-CEFD-4AF1-8527-282DCDEC0603@cgicommunications.com> <87f2f3b90908310955u399f5c5v1214d58e6004e392@mail.gmail.com> <40C6E7F6-D49A-4902-B3F8-224567325875@cgicommunications.com> Message-ID: In esl you get events for each dtmf. Mike On Sep 1, 2009, at 9:54 AM, Greg Thoen wrote: > 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 > > 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; >> >> 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/9d9f9cca/attachment.html From anthony.minessale at gmail.com Thu Sep 3 11:30:25 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 3 Sep 2009 13:30:25 -0500 Subject: [Freeswitch-users] mod_xml_curl.c Oversized file detected [1056100 bytes] In-Reply-To: <4A9FFD64.2000701@gmx.net> References: <4A9FFD64.2000701@gmx.net> Message-ID: <191c3a030909031130n453786ccyd71ee0b9f92a3955@mail.gmail.com> you can edit mod_xml_curl.c line 64 and increase XML_CURL_MAX_BYTES On Thu, Sep 3, 2009 at 12:31 PM, Peter P GMX wrote: > Hello, > > in a B2BUA scenario we have 2000 defined gateways (defined but not > registered yet). > When reloading mod_sofia Freeswitch complains about the XML-Curl File > size > 1MB and deactivates all gateways: > mod_xml_curl.c:121 Oversized file detected [1056100 bytes] > > Is there any way to overcome this? Currently we have 2000 gateways > defined. Finally we will have about 10.000. And we will not be able to > reduce the file size below 1 MB. It will become ~ 2-3 MB maybe. > > Best Regards > Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/302899ec/attachment-0001.html From jerry.richards at teotech.com Thu Sep 3 11:40:36 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Thu, 3 Sep 2009 11:40:36 -0700 Subject: [Freeswitch-users] Presence Feature Message-ID: Does Freeswitch support Presence via SIMPLE protocol? Can it maintain presence? I presume this would be a SUBSCRIBE/NOTIFY arrangement? Best Regards, Jerry From tculjaga at gmail.com Thu Sep 3 11:59:44 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 3 Sep 2009 20:59:44 +0200 Subject: [Freeswitch-users] T38 <> T30 transcoding In-Reply-To: <5e757f90909030729g63c63becve6440b7ce042156b@mail.gmail.com> References: <65d96fc80909021232n6784ef42ra1c6e6648109c059@mail.gmail.com> <65d96fc80909021353t67699eb8o770b3fa1e364ef98@mail.gmail.com> <65d96fc80909030529x1a60db4eg152a498f8b71ae50@mail.gmail.com> <5e757f90909030729g63c63becve6440b7ce042156b@mail.gmail.com> Message-ID: <65d96fc80909031159l5ab6f88fhf32d19b6b390af00@mail.gmail.com> nice to know that... anyhow i'd like to have this on FS as well :) T. On Thu, Sep 3, 2009 at 4:29 PM, T.R. Missner wrote: > if you buy a sip trunk from bandwidth.com and ask them to put you on the > pvp route you will get this functionality > > > On Thu, Sep 3, 2009 at 8:29 AM, Tihomir Culjaga wrote: > >> anyone knows anything about this? >> >> T. >> >> >> On Wed, Sep 2, 2009 at 10:53 PM, Tihomir Culjaga wrote: >> >>> I will put several nickels saying it is impossible :) >>> >>> >>> seriously, can it be done? >>> >>> T. >>> >>> On Wed, Sep 2, 2009 at 10:35 PM, Tim Meade wrote: >>> >>>> I am very interested in a response to this. Last I knew there was >>>> only T.38 pass through and for some reason I'm not even sure it that was >>>> fully implemented. >>>> >>>> >>>> >>>> Tim >>>> >>>> >>>> >>>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Tihomir >>>> Culjaga >>>> *Sent:* Wednesday, September 02, 2009 3:33 PM >>>> *To:* freeswitch-users at lists.freeswitch.org >>>> *Subject:* [Freeswitch-users] T38 <> T30 transcoding >>>> >>>> >>>> >>>> Hi guys, >>>> >>>> just a quick question... is it possible to do a reliable on the fly T30 >>>> <> T38 transcoding at all ... what is the status of T.38 on FS ? >>>> >>>> T, >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/434f0761/attachment.html From diego.viola at gmail.com Thu Sep 3 12:27:57 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 3 Sep 2009 19:27:57 +0000 Subject: [Freeswitch-users] Presence Feature In-Reply-To: References: Message-ID: <86a32abc0909031227u62940a30w17a440f6fbafb241@mail.gmail.com> Not sure if it does via SIMPLE protocol, but if you do "event plain all" you should see the PRESENCE_IN and PRESENCE_OUT events. Diego On Thu, Sep 3, 2009 at 6:40 PM, Jerry Richards wrote: > Does Freeswitch support Presence via SIMPLE protocol? Can it maintain > presence? I presume this would be a SUBSCRIBE/NOTIFY arrangement? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/a7875627/attachment.html From diego.viola at gmail.com Thu Sep 3 12:30:10 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 3 Sep 2009 19:30:10 +0000 Subject: [Freeswitch-users] Presence Feature In-Reply-To: <86a32abc0909031227u62940a30w17a440f6fbafb241@mail.gmail.com> References: <86a32abc0909031227u62940a30w17a440f6fbafb241@mail.gmail.com> Message-ID: <86a32abc0909031230q799e247clfa7fac4a29d0561f@mail.gmail.com> I have made a simple script that catches the PRESENCE IN/OUT events and passes them to a method to get the information for the phones, you can take a look at it here. http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/presence/presence.rb?r=14679 Diego On Thu, Sep 3, 2009 at 7:27 PM, Diego Viola wrote: > Not sure if it does via SIMPLE protocol, but if you do "event plain all" > you should see the PRESENCE_IN and PRESENCE_OUT events. > > Diego > > > On Thu, Sep 3, 2009 at 6:40 PM, Jerry Richards > wrote: > >> Does Freeswitch support Presence via SIMPLE protocol? Can it maintain >> presence? I presume this would be a SUBSCRIBE/NOTIFY arrangement? >> >> Best Regards, >> Jerry >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/3f482f38/attachment.html From christian.loeschenkohl at xpirio.com Thu Sep 3 12:39:46 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Thu, 03 Sep 2009 21:39:46 +0200 Subject: [Freeswitch-users] stability problems In-Reply-To: <191c3a030909031011y2d73ed4bsd6c259297f8ccd5e@mail.gmail.com> References: <4A9F7067.2090306@xpirio.com> <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> <4A9FD6EE.7070107@xpirio.com> <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> <3194DE12-AB44-4F7D-8BD7-D7171B41A735@freeswitch.org> <4A9FF24F.6000001@xpirio.com> <191c3a030909031011y2d73ed4bsd6c259297f8ccd5e@mail.gmail.com> Message-ID: <4AA01B82.4040805@xpirio.com> yes, thank you for this i will follow your instructions an remove any files from older builds and reinstall - then I'll give feedback hope this resolves my problem br On 2009-09-03 19:11, Anthony Minessale wrote: > I already provided exact instructions. > > > On Thu, Sep 3, 2009 at 11:50 AM, Brian West > wrote: > > Please join IRC if you experience the issue again #freeswitch on > irc.freenode.net > > /b > > > On Sep 3, 2009, at 11:43 AM, Christian L?schenkohl wrote: > > > sorry i can not follow you > > i build everthing from scratch (download source, unpack and build) > > > > what i mean with the build-essential package is a debian meta package > > that contains gcc, make and so on > > > > br > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com From pjintheusa at gmail.com Thu Sep 3 12:55:31 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 3 Sep 2009 15:55:31 -0400 Subject: [Freeswitch-users] uuid_exists - does it still exist? Message-ID: <367751820909031255l1c6947c6t1bf577c77551f9b3@mail.gmail.com> Hi there: The wiki states: uuid_exists Checks whether a given UUID exists. Usage: uuid_exists However when I call via an API call I get: INVALID COMMAND! I also don't see it in MOD_COMMAND.C Has it gone? Thanks Phillip Jones. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/e9317d21/attachment.html From pedroprado at msn.com Thu Sep 3 13:05:52 2009 From: pedroprado at msn.com (Pedro Prado) Date: Thu, 3 Sep 2009 17:05:52 -0300 Subject: [Freeswitch-users] Digium boards Message-ID: Hi, Does Freeswith work with any Digium boards? Ps: Sorry for the newbie question. Thanks, Pedro Prado _________________________________________________________________ Acesse o Portal MSN do seu celular e se mantenha sempre atualizado. Clique aqui. http://www.windowslive.com.br/celular/home.asp?utm_source=MSN_Hotmail&utm_medium=Tagline&utm_campaign=MobileServices200908 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/6b3f075c/attachment-0001.html From diego.viola at gmail.com Thu Sep 3 13:15:58 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 3 Sep 2009 20:15:58 +0000 Subject: [Freeswitch-users] Digium boards In-Reply-To: References: Message-ID: <86a32abc0909031315i4242391eo65325aa552136069@mail.gmail.com> Hi Pedro, Yes it should, it does. Diego 2009/9/3 Pedro Prado > Hi, > > Does Freeswith work with any Digium boards? > > Ps: Sorry for the newbie question. > > Thanks, > Pedro Prado > > ------------------------------ > Novo Internet Explorer 8: traduza com apenas um clique. Baixe agora, ? > gr?tis! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/11cfadf0/attachment.html From brian at freeswitch.org Thu Sep 3 13:20:44 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Sep 2009 15:20:44 -0500 Subject: [Freeswitch-users] uuid_exists - does it still exist? In-Reply-To: <367751820909031255l1c6947c6t1bf577c77551f9b3@mail.gmail.com> References: <367751820909031255l1c6947c6t1bf577c77551f9b3@mail.gmail.com> Message-ID: You're not on the latest SVN that was recently added. /b On Sep 3, 2009, at 2:55 PM, Phillip Jones wrote: > Hi there: > > The wiki states: > > uuid_exists > > Checks whether a given UUID exists. > > Usage: uuid_exists > > > > > However when I call via an API call I get: > > INVALID COMMAND! > > I also don't see it in MOD_COMMAND.C > > > Has it gone? > > > Thanks > > > Phillip Jones. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/fbbcd27f/attachment.html From diego.viola at gmail.com Thu Sep 3 13:21:01 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 3 Sep 2009 20:21:01 +0000 Subject: [Freeswitch-users] Digium boards In-Reply-To: <86a32abc0909031315i4242391eo65325aa552136069@mail.gmail.com> References: <86a32abc0909031315i4242391eo65325aa552136069@mail.gmail.com> Message-ID: <86a32abc0909031321i61700bc8rcd273c73681230b4@mail.gmail.com> Any Zaptel-compatible card should work. Diego On Thu, Sep 3, 2009 at 8:15 PM, Diego Viola wrote: > Hi Pedro, > > Yes it should, it does. > > Diego > > 2009/9/3 Pedro Prado > >> Hi, >> >> Does Freeswith work with any Digium boards? >> >> Ps: Sorry for the newbie question. >> >> Thanks, >> Pedro Prado >> >> ------------------------------ >> Novo Internet Explorer 8: traduza com apenas um clique. Baixe agora, ? >> gr?tis! >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/49d805c2/attachment.html From lfurrea at gmail.com Thu Sep 3 13:29:30 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Thu, 3 Sep 2009 14:29:30 -0600 Subject: [Freeswitch-users] Presence Feature In-Reply-To: <86a32abc0909031230q799e247clfa7fac4a29d0561f@mail.gmail.com> References: <86a32abc0909031227u62940a30w17a440f6fbafb241@mail.gmail.com> <86a32abc0909031230q799e247clfa7fac4a29d0561f@mail.gmail.com> Message-ID: Diego can you explain a little bit further what your script does. Not really versed into ruby but I certainly interested on these events On Thu, Sep 3, 2009 at 1:30 PM, Diego Viola wrote: > I have made a simple script that catches the PRESENCE IN/OUT events and > passes them to a method to get the information for the phones, you can take > a look at it here. > > > http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/presence/presence.rb?r=14679 > > Diego > > > On Thu, Sep 3, 2009 at 7:27 PM, Diego Viola wrote: > >> Not sure if it does via SIMPLE protocol, but if you do "event plain all" >> you should see the PRESENCE_IN and PRESENCE_OUT events. >> >> Diego >> >> >> On Thu, Sep 3, 2009 at 6:40 PM, Jerry Richards < >> jerry.richards at teotech.com> wrote: >> >>> Does Freeswitch support Presence via SIMPLE protocol? Can it maintain >>> presence? I presume this would be a SUBSCRIBE/NOTIFY arrangement? >>> >>> Best Regards, >>> Jerry >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/d3abd4e2/attachment.html From diego.viola at gmail.com Thu Sep 3 13:46:24 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 3 Sep 2009 20:46:24 +0000 Subject: [Freeswitch-users] Presence Feature In-Reply-To: References: <86a32abc0909031227u62940a30w17a440f6fbafb241@mail.gmail.com> <86a32abc0909031230q799e247clfa7fac4a29d0561f@mail.gmail.com> Message-ID: <86a32abc0909031346lc9a541eqf24bd57bdce7f06e@mail.gmail.com> Hi Luis, My script simply adds two event hooks for presence (PRESENCE_IN and PRESENCE_OUT) and then it passes those events to a method (presence_handler) and it prints the "from" which is something like: 1000%192.168.0.2 (the user and ip) and the status: which can be Registered or Unregistered. The script is a daemon, so when you register/unregister a phone you see something like this: *I, [2009-08-29T03:05:20.351467 #3092] INFO -- : Phone 1000%192.168.0.2 - Unregistered I, [2009-08-29T03:05:20.864802 #3092] INFO -- : Phone 1000%192.168.0.2 - Unregistered I, [2009-08-29T03:05:20.865763 #3092] INFO -- : Phone 1000%192.168.0.2 - Registered(UDP) I, [2009-08-29T03:05:21.390911 #3092] INFO -- : Phone 1000%192.168.0.2 - Registered(UDP) I, [2009-08-29T03:05:21.391308 #3092] INFO -- : Phone 1000%192.168.0.2 - Registered(UDP) I, [2009-08-29T03:05:21.548433 #3092] INFO -- : Phone 1000%192.168.0.2 - Unregistered* That's pretty much what it does, but this makes it possible to make apps to report the status of phones, etc. I was thinking to send that information on a database and then show it on a web interface, with Ajax, etc. on "near real time", it would be pretty cool :-). Let me know if you have any questions or need any help. Regards, Diego On Thu, Sep 3, 2009 at 8:29 PM, Luis F Urrea wrote: > Diego can you explain a little bit further what your script does. > > Not really versed into ruby but I certainly interested on these events > > > On Thu, Sep 3, 2009 at 1:30 PM, Diego Viola wrote: > >> I have made a simple script that catches the PRESENCE IN/OUT events and >> passes them to a method to get the information for the phones, you can take >> a look at it here. >> >> >> http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/presence/presence.rb?r=14679 >> >> Diego >> >> >> On Thu, Sep 3, 2009 at 7:27 PM, Diego Viola wrote: >> >>> Not sure if it does via SIMPLE protocol, but if you do "event plain all" >>> you should see the PRESENCE_IN and PRESENCE_OUT events. >>> >>> Diego >>> >>> >>> On Thu, Sep 3, 2009 at 6:40 PM, Jerry Richards < >>> jerry.richards at teotech.com> wrote: >>> >>>> Does Freeswitch support Presence via SIMPLE protocol? Can it maintain >>>> presence? I presume this would be a SUBSCRIBE/NOTIFY arrangement? >>>> >>>> Best Regards, >>>> Jerry >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/551a5e13/attachment-0001.html From pedroprado at msn.com Thu Sep 3 13:49:01 2009 From: pedroprado at msn.com (Pedro Prado) Date: Thu, 3 Sep 2009 17:49:01 -0300 Subject: [Freeswitch-users] Digium boards In-Reply-To: <86a32abc0909031321i61700bc8rcd273c73681230b4@mail.gmail.com> References: <86a32abc0909031315i4242391eo65325aa552136069@mail.gmail.com> <86a32abc0909031321i61700bc8rcd273c73681230b4@mail.gmail.com> Message-ID: Hi Diego, I have a Digium AEX410 to test in laboratory. I think, that it works, not? Thanks, Pedro Prado Date: Thu, 3 Sep 2009 20:21:01 +0000 From: diego.viola at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Digium boards Any Zaptel-compatible card should work. Diego On Thu, Sep 3, 2009 at 8:15 PM, Diego Viola wrote: Hi Pedro, Yes it should, it does. Diego 2009/9/3 Pedro Prado Hi, Does Freeswith work with any Digium boards? Ps: Sorry for the newbie question. Thanks, Pedro Prado Novo Internet Explorer 8: traduza com apenas um clique. Baixe agora, ? gr?tis! _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________ Acesse seu Hotmail de onde quer que esteja atrav?s do celular. Clique aqui. http://www.windowslive.com.br/celular/home.asp?utm_source=MSN_Hotmail&utm_medium=Tagline&utm_campaign=MobileServices200908 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/400b28ad/attachment.html From diego.viola at gmail.com Thu Sep 3 13:56:23 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 3 Sep 2009 20:56:23 +0000 Subject: [Freeswitch-users] Digium boards In-Reply-To: References: <86a32abc0909031315i4242391eo65325aa552136069@mail.gmail.com> <86a32abc0909031321i61700bc8rcd273c73681230b4@mail.gmail.com> Message-ID: <86a32abc0909031356r43806f3j9ea982aaabe9a7ca@mail.gmail.com> Sure. Diego On Thu, Sep 3, 2009 at 8:49 PM, Pedro Prado wrote: > Hi Diego, > > I have a Digium AEX410 to test in laboratory. I think, that it works, not? > > Thanks, > Pedro Prado > > ------------------------------ > Date: Thu, 3 Sep 2009 20:21:01 +0000 > From: diego.viola at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Digium boards > > > Any Zaptel-compatible card should work. > > Diego > > On Thu, Sep 3, 2009 at 8:15 PM, Diego Viola wrote: > > Hi Pedro, > > Yes it should, it does. > > Diego > > 2009/9/3 Pedro Prado > > Hi, > > Does Freeswith work with any Digium boards? > > Ps: Sorry for the newbie question. > > Thanks, > Pedro Prado > > ------------------------------ > Novo Internet Explorer 8: traduza com apenas um clique. Baixe agora, ? > gr?tis! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > ------------------------------ > Novo Internet Explorer 8: fa?a tudo com menos cliques. Baixe agora, ? > gratis! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/29f11b87/attachment.html From diego.viola at gmail.com Thu Sep 3 14:08:59 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 3 Sep 2009 21:08:59 +0000 Subject: [Freeswitch-users] Presence Feature In-Reply-To: <86a32abc0909031346lc9a541eqf24bd57bdce7f06e@mail.gmail.com> References: <86a32abc0909031227u62940a30w17a440f6fbafb241@mail.gmail.com> <86a32abc0909031230q799e247clfa7fac4a29d0561f@mail.gmail.com> <86a32abc0909031346lc9a541eqf24bd57bdce7f06e@mail.gmail.com> Message-ID: <86a32abc0909031408k40a8ac1fwe1dfbf58bd78066c@mail.gmail.com> You should try to do something like "event plain all" from the freeswitch CLI and see all the events that you can use in your apps, etc. The sky is the limit ;) Diego On Thu, Sep 3, 2009 at 8:46 PM, Diego Viola wrote: > Hi Luis, > > My script simply adds two event hooks for presence (PRESENCE_IN and > PRESENCE_OUT) and then it passes those events to a method (presence_handler) > and it prints the "from" which is something like: 1000%192.168.0.2 (the user > and ip) and the status: which can be Registered or Unregistered. > > The script is a daemon, so when you register/unregister a phone you see > something like this: > > *I, [2009-08-29T03:05:20.351467 #3092] INFO -- : Phone 1000%192.168.0.2 - > Unregistered > > I, [2009-08-29T03:05:20.864802 #3092] INFO -- : Phone 1000%192.168.0.2 - > Unregistered > > I, [2009-08-29T03:05:20.865763 #3092] INFO -- : Phone 1000%192.168.0.2 - > Registered(UDP) > > I, [2009-08-29T03:05:21.390911 #3092] INFO -- : Phone 1000%192.168.0.2 - > Registered(UDP) > > I, [2009-08-29T03:05:21.391308 #3092] INFO -- : Phone 1000%192.168.0.2 - > Registered(UDP) > > I, [2009-08-29T03:05:21.548433 #3092] INFO -- : Phone 1000%192.168.0.2 - > Unregistered* > > That's pretty much what it does, but this makes it possible to make apps to > report the status of phones, etc. I was thinking to send that information on > a database and then show it on a web interface, with Ajax, etc. on "near > real time", it would be pretty cool :-). > > Let me know if you have any questions or need any help. > > Regards, > > Diego > > > On Thu, Sep 3, 2009 at 8:29 PM, Luis F Urrea wrote: > >> Diego can you explain a little bit further what your script does. >> >> Not really versed into ruby but I certainly interested on these events >> >> >> On Thu, Sep 3, 2009 at 1:30 PM, Diego Viola wrote: >> >>> I have made a simple script that catches the PRESENCE IN/OUT events and >>> passes them to a method to get the information for the phones, you can take >>> a look at it here. >>> >>> >>> http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/presence/presence.rb?r=14679 >>> >>> Diego >>> >>> >>> On Thu, Sep 3, 2009 at 7:27 PM, Diego Viola wrote: >>> >>>> Not sure if it does via SIMPLE protocol, but if you do "event plain all" >>>> you should see the PRESENCE_IN and PRESENCE_OUT events. >>>> >>>> Diego >>>> >>>> >>>> On Thu, Sep 3, 2009 at 6:40 PM, Jerry Richards < >>>> jerry.richards at teotech.com> wrote: >>>> >>>>> Does Freeswitch support Presence via SIMPLE protocol? Can it maintain >>>>> presence? I presume this would be a SUBSCRIBE/NOTIFY arrangement? >>>>> >>>>> Best Regards, >>>>> Jerry >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/b63a4867/attachment.html From dmitry.bely at gmail.com Thu Sep 3 13:26:05 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Fri, 4 Sep 2009 00:26:05 +0400 Subject: [Freeswitch-users] Proxy authorization Message-ID: <90823c940909031326g5e9d5a7bsd2650a59a7a5eaf@mail.gmail.com> Hi, My SIP provider's gateway requires authorization with correct realm specified. So I configured gateway with "realm" parameter: (default_provider_register = true) Unfortunately even after that there is no "Authorization:" header in the REGISTER message: REGISTER sip:1.2.3.4 SIP/2.0 Via: SIP/2.0/UDP 5.6.7.8:5080;rport;branch=z9hG4bKNBB3ygD85y3eF Max-Forwards: 70 From: ;tag=Nrc6Z9yrNBS3H To: Call-ID: a93d949a-98c1-11de-b6b8-8321249ad8d4 CSeq: 119885384 REGISTER Contact: Expires: 600 User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14707M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 and the provider responds with 403 User '1111111' is disabled (as there is no correct authorization realm there). How to force Authorization header? Am I missing something? - Dmitry Bely From diego.viola at gmail.com Thu Sep 3 14:11:49 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 3 Sep 2009 21:11:49 +0000 Subject: [Freeswitch-users] Digium boards In-Reply-To: <86a32abc0909031356r43806f3j9ea982aaabe9a7ca@mail.gmail.com> References: <86a32abc0909031315i4242391eo65325aa552136069@mail.gmail.com> <86a32abc0909031321i61700bc8rcd273c73681230b4@mail.gmail.com> <86a32abc0909031356r43806f3j9ea982aaabe9a7ca@mail.gmail.com> Message-ID: <86a32abc0909031411i24ab9b21sf73f098a85f90779@mail.gmail.com> Join #freeswitch at irc.freenode.net if you have any questions, my nick is "diegoviola" there :). Diego On Thu, Sep 3, 2009 at 8:56 PM, Diego Viola wrote: > Sure. > > Diego > > > On Thu, Sep 3, 2009 at 8:49 PM, Pedro Prado wrote: > >> Hi Diego, >> >> I have a Digium AEX410 to test in laboratory. I think, that it works, not? >> >> Thanks, >> Pedro Prado >> >> ------------------------------ >> Date: Thu, 3 Sep 2009 20:21:01 +0000 >> From: diego.viola at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Digium boards >> >> >> Any Zaptel-compatible card should work. >> >> Diego >> >> On Thu, Sep 3, 2009 at 8:15 PM, Diego Viola wrote: >> >> Hi Pedro, >> >> Yes it should, it does. >> >> Diego >> >> 2009/9/3 Pedro Prado >> >> Hi, >> >> Does Freeswith work with any Digium boards? >> >> Ps: Sorry for the newbie question. >> >> Thanks, >> Pedro Prado >> >> ------------------------------ >> Novo Internet Explorer 8: traduza com apenas um clique. Baixe agora, ? >> gr?tis! >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> ------------------------------ >> Novo Internet Explorer 8: fa?a tudo com menos cliques. Baixe agora, ? >> gratis! >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/8fa5fa1a/attachment-0001.html From brian at freeswitch.org Thu Sep 3 14:19:35 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Sep 2009 16:19:35 -0500 Subject: [Freeswitch-users] Proxy authorization In-Reply-To: <90823c940909031326g5e9d5a7bsd2650a59a7a5eaf@mail.gmail.com> References: <90823c940909031326g5e9d5a7bsd2650a59a7a5eaf@mail.gmail.com> Message-ID: <7B180BCD-82F4-43BD-A8C9-48CC2A38D899@freeswitch.org> There will not be an authorization header on the first register attempt... it only happens once we are 401/407'ed and the phone comes back and registers again. /b On Sep 3, 2009, at 3:26 PM, Dmitry Bely wrote: > Unfortunately even after that there is no "Authorization:" header in > the REGISTER message: > > REGISTER sip:1.2.3.4 SIP/2.0 > Via: SIP/2.0/UDP 5.6.7.8:5080;rport;branch=z9hG4bKNBB3ygD85y3eF > Max-Forwards: 70 > From: ;tag=Nrc6Z9yrNBS3H > To: > Call-ID: a93d949a-98c1-11de-b6b8-8321249ad8d4 > CSeq: 119885384 REGISTER > Contact: > Expires: 600 > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14707M > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/f76d26a0/attachment.html From fraunhofer.lists.freeswitch-001 at traced.net Thu Sep 3 14:23:26 2009 From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer) Date: Thu, 3 Sep 2009 23:23:26 +0200 Subject: [Freeswitch-users] uuid_exists - does it still exist? In-Reply-To: References: <367751820909031255l1c6947c6t1bf577c77551f9b3@mail.gmail.com> Message-ID: Hi, > Usage: uuid_exists > However when I call via an API call I get: > INVALID COMMAND! > I also don't see it in MOD_COMMAND.C As a workaround or if your are unable to upgrade, you can use "uuid_getvar [some_uuid] thisVariableDoesNotExist" ("thisVariableDoesNotExist" is any variable you can think of, a valid one or literally 'thisVariableDoesNotExist' :) You'll either get an error that this channel does not exist any longer or "undef" for the channel variable. The api will return "+OK" in case the channel still exists, and "ERROR" in case it does not. Beni. From msc at freeswitch.org Thu Sep 3 14:50:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 Sep 2009 14:50:14 -0700 Subject: [Freeswitch-users] mod_xml_curl.c Oversized file detected [1056100 bytes] In-Reply-To: <191c3a030909031130n453786ccyd71ee0b9f92a3955@mail.gmail.com> References: <4A9FFD64.2000701@gmx.net> <191c3a030909031130n453786ccyd71ee0b9f92a3955@mail.gmail.com> Message-ID: <87f2f3b90909031450u1957edb0xd9631431ab6dce6d@mail.gmail.com> Okay I added this tidbit to the mod_xml_curl page for future reference. -MC On Thu, Sep 3, 2009 at 11:30 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you can edit mod_xml_curl.c line 64 > and increase XML_CURL_MAX_BYTES > > > > On Thu, Sep 3, 2009 at 12:31 PM, Peter P GMX wrote: > >> Hello, >> >> in a B2BUA scenario we have 2000 defined gateways (defined but not >> registered yet). >> When reloading mod_sofia Freeswitch complains about the XML-Curl File >> size > 1MB and deactivates all gateways: >> mod_xml_curl.c:121 Oversized file detected [1056100 bytes] >> >> Is there any way to overcome this? Currently we have 2000 gateways >> defined. Finally we will have about 10.000. And we will not be able to >> reduce the file size below 1 MB. It will become ~ 2-3 MB maybe. >> >> Best Regards >> Peter >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/d3d9ae98/attachment.html From msc at freeswitch.org Thu Sep 3 14:53:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 Sep 2009 14:53:31 -0700 Subject: [Freeswitch-users] Digium boards In-Reply-To: References: <86a32abc0909031315i4242391eo65325aa552136069@mail.gmail.com> <86a32abc0909031321i61700bc8rcd273c73681230b4@mail.gmail.com> Message-ID: <87f2f3b90909031453s14ccf065m528ff913397658e3@mail.gmail.com> On Thu, Sep 3, 2009 at 1:49 PM, Pedro Prado wrote: > Hi Diego, > > I have a Digium AEX410 to test in laboratory. I think, that it works, not? > Okay, it looks like that's the next-gen version of the TDM400. I'm pretty sure you'll be okay. Check out the wiki page on OpenZAP which is the TDM abstraction layer that FS uses: http://wiki.freeswitch.org/wiki/OpenZAP Have fun! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/44a70bcb/attachment.html From msc at freeswitch.org Thu Sep 3 14:56:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 Sep 2009 14:56:23 -0700 Subject: [Freeswitch-users] uuid_exists - does it still exist? In-Reply-To: References: <367751820909031255l1c6947c6t1bf577c77551f9b3@mail.gmail.com> Message-ID: <87f2f3b90909031456s76a12874xe0a6ebfa26babb10@mail.gmail.com> On Thu, Sep 3, 2009 at 2:23 PM, Benedikt Fraunhofer < fraunhofer.lists.freeswitch-001 at traced.net> wrote: > Hi, > > > > Usage: uuid_exists > > However when I call via an API call I get: > > INVALID COMMAND! > > I also don't see it in MOD_COMMAND.C > > As a workaround or if your are unable to upgrade, you can use > "uuid_getvar [some_uuid] thisVariableDoesNotExist" > ("thisVariableDoesNotExist" is any variable you can think of, a valid > one or literally 'thisVariableDoesNotExist' :) > You'll either get an error that this channel does not exist any longer > or "undef" for the channel variable. > The api will return "+OK" in case the channel still exists, and > "ERROR" in case it does not. > > Beni. > > You could even do this: uuid_getvar uuid If it exists then the return will be the uuid. :) Although I must say I recommend this instead: make current :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/6b4decfa/attachment.html From hads at nice.net.nz Thu Sep 3 16:42:25 2009 From: hads at nice.net.nz (Hadley Rich) Date: Fri, 4 Sep 2009 11:42:25 +1200 Subject: [Freeswitch-users] Digium boards In-Reply-To: <87f2f3b90909031453s14ccf065m528ff913397658e3@mail.gmail.com> References: <87f2f3b90909031453s14ccf065m528ff913397658e3@mail.gmail.com> Message-ID: <200909041142.26011.hads@nice.net.nz> On Fri, 04 Sep 2009 09:53:31 Michael Collins wrote: > Okay, it looks like that's the next-gen version of the TDM400. It is, the TDM410 is the new TDM400 and the AEX is the PCI-ex version. The TDM410 at least and so I assume the AEX410 uses the same driver as the TDM2400 i.e. not the same driver as the TDM400. hads -- https://nicegear.co.nz VoIP and Open Source Hardware From dujinfang at gmail.com Thu Sep 3 16:51:48 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 4 Sep 2009 07:51:48 +0800 Subject: [Freeswitch-users] select batchfile after call In-Reply-To: <25275633.post@talk.nabble.com> References: <25275633.post@talk.nabble.com> Message-ID: <5EC4D247-DB33-441C-A622-14FED6ADD1FC@gmail.com> hangup hook api? system or system_bg ? not sure On Sep 3, 2009, at 9:08 PM, NOx-WHV wrote: > > Hi, > > does anybody have a tip how to start a batchfile after hanging up. > > After ext. 1000 calls 1001 and hang up, i need a request to call: > > /../../FS/batchfile 1000 > > if 1001 calls 1000 i need: > > /../../FS/batchfile 1001 > > and so on... > > > Thanks for help > -- > View this message in context: http://www.nabble.com/select-batchfile-after-call-tp25275633p25275633.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tayeb.meftah at gmail.com Thu Sep 3 18:14:10 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Fri, 04 Sep 2009 01:14:10 +0000 Subject: [Freeswitch-users] Presence Feature In-Reply-To: References: Message-ID: <4AA069E2.3060507@gmail.com> freeswitch support all sip standard featurs including presence/ blf/... and all featurs that sofia sip support is supported by freeswitch, + aditional featurs thanks and welcome to freeswitch! Jerry Richards wrote: > Does Freeswitch support Presence via SIMPLE protocol? Can it maintain > presence? I presume this would be a SUBSCRIBE/NOTIFY arrangement? > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From nandy1925 at gmail.com Thu Sep 3 17:26:20 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Fri, 4 Sep 2009 08:26:20 +0800 Subject: [Freeswitch-users] Set disable-transcoding in dialplan In-Reply-To: <4A9F6745.6030607@laposte.net> References: <4A9BE5AB.4030304@laposte.net> <87f2f3b90908310938oa7e838dkd6e07579f224de37@mail.gmail.com> <4A9F6745.6030607@laposte.net> Message-ID: <7d0bfd8c0909031726t53f9900dldf7f2c0eab97305@mail.gmail.com> rod, have you tried this? http://lists.freeswitch.org/pipermail/freeswitch-users/2008-March/002199.html /nandy On Thu, Sep 3, 2009 at 2:50 PM, rod wrote: > Hi Michael, > > I did some tests but I haven't been successful, so there is what I'm > trying to achieve: > > On A leg, my phone is using: PCMA and G729 (in this priority order) > > With PEER A, I want to use only G729 (thats is the only codec that this > PEER support), so that the RTP flow will be: > Phone-----G729----FS-----G729-----PEER_A > > With PEER B, I want to use only G711, so: > Phone-----G711----FS-----G711-----PEER_B > > In fact, I'd like to force FS announcing the codec list priority based > on the priority of the codec announced by the PEER, cause FS is unable > to transcode G729 <--> G711. > > Tried a lot of things (greedy for codec-negociation, late_codec, > disable_transcoding, codec-prefs) without success. > > If you have some clue. > > regards, > rod > > Michael Collins a ?crit : > > Check out this page: > > http://wiki.freeswitch.org/wiki/Codec_negotiation > > > > Late negotiation will probably let you handle all the cases you need. > > -MC > > > > On Mon, Aug 31, 2009 at 8:00 AM, rod > > wrote: > > > > Hi all, > > > > I'm wondering if I can do something like this: > > - in my internal profile, I have this because of some PEER > > using G729: > > - > > > > But for a specific PEER, I'd like to activate transcoding: > > - for this PEER, only G711 is used > > - I'd like to transcode DTMF SIP INFO or RFC2833 to INBAND > > > > So in my dialplan, I tried before bridging: > > > > - > > - > > > > But I still see RFC2833 events between my FS and PEER and the DTMF > are > > not working. > > > > So 2 questions: > > - does application "start_dtmf_generate" requires transcoding > > - if yes, can I set the variable disable-transcoding in my > dialplan > > > > regards, > > rod > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/6916ca84/attachment.html From larclap at yahoo.com Thu Sep 3 18:27:53 2009 From: larclap at yahoo.com (Lars Zeb) Date: Thu, 3 Sep 2009 18:27:53 -0700 Subject: [Freeswitch-users] Trouble parsing sip:888@conference.freeswitch.org? Message-ID: <01b301ca2cfe$ec8d02a0$c5a707e0$@com> I tried to dial sip:888 at conference.freeswitch.org via a Bria softphone. Why does the parsed regex look like: mod_dialplan_xml.c:315 Processing 1009->888 in context default http://pastebin.freeswitch.org/10199 It looks like the domain is stripped off. What am I missing? Thanks, Lars -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/954677d6/attachment.html From brian at freeswitch.org Thu Sep 3 20:00:56 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 3 Sep 2009 22:00:56 -0500 Subject: [Freeswitch-users] Trouble parsing sip:888@conference.freeswitch.org? In-Reply-To: <01b301ca2cfe$ec8d02a0$c5a707e0$@com> References: <01b301ca2cfe$ec8d02a0$c5a707e0$@com> Message-ID: <4505E4EE-71CB-4DCE-92CC-81B5807097CA@freeswitch.org> FreeSWITCH is not a proxy so you'll have to look at all the variables to see if you need to send the invite out.... use the info app. /b On Sep 3, 2009, at 8:27 PM, Lars Zeb wrote: > I tried to dial sip:888 at conference.freeswitch.org via a Bria > softphone. Why does the parsed regex look like: > > mod_dialplan_xml.c:315 Processing 1009->888 in context default > > http://pastebin.freeswitch.org/10199 > > It looks like the domain is stripped off. What am I missing? > > Thanks, Lars > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/a716b6c1/attachment.html From kawarod at laposte.net Thu Sep 3 22:54:40 2009 From: kawarod at laposte.net (rod) Date: Fri, 04 Sep 2009 09:54:40 +0400 Subject: [Freeswitch-users] Set disable-transcoding in dialplan In-Reply-To: <7d0bfd8c0909031726t53f9900dldf7f2c0eab97305@mail.gmail.com> References: <4A9BE5AB.4030304@laposte.net> <87f2f3b90908310938oa7e838dkd6e07579f224de37@mail.gmail.com> <4A9F6745.6030607@laposte.net> <7d0bfd8c0909031726t53f9900dldf7f2c0eab97305@mail.gmail.com> Message-ID: <4AA0ABA0.5020901@laposte.net> Hi Nandy, yes already tried this, but if I use proxy_media=true, FS makes no control on the content of the RTP stream. But the pbm is that I need to use this: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate This function enables transcoding of SIP_INFO or RFC2833 to Inband DTMF in G711 But this feature doesn't work if I'm using proxy_media=true. In fact my setup is the following: CPE using G711A, G729 and SIP INFO for DTMF PEER_A using G729 only and RFC_2833 PEER_B using G711 and SIP INFO I have been able to make this works, with proxy_media=true for PEER_B cause I don't need transcoding of DTMF (SIP INFO to SIP INFO). For PEER_A, proxy_media is set to false (default) cause I need transcoding SIP INFO to RFC2833. I'm able to use G729 using codec_negotiation=greedy and setting G729 with highest priority on my internal profile. But the pbm is that I need to add PEER_C. PEER_C needs G711 with transcoding DTMF from SIP_INFO to Inband. And this is where I'm stuck, cause using "greedy settings and G729 with priority 1 in my codec list and proxy_media=false" force FS to negotiate G729 on leg A. But Leg B is willing to use G711 and FS is unable to transcode G729 <---> G711. I was wondering if there is a way for FS to force the codec order on Leg A with some knowledge of the preferred codec on Leg B, ie I know that Leg B will always use G711 so that I want to biase the SDP answer on Leg A based on this fact. regards, rod Nandy Dagondon a ?crit : > rod, > > have you tried this? > http://lists.freeswitch.org/pipermail/freeswitch-users/2008-March/002199.html > > /nandy > > > On Thu, Sep 3, 2009 at 2:50 PM, rod > wrote: > > Hi Michael, > > I did some tests but I haven't been successful, so there is what I'm > trying to achieve: > > On A leg, my phone is using: PCMA and G729 (in this priority order) > > With PEER A, I want to use only G729 (thats is the only codec that > this > PEER support), so that the RTP flow will be: > Phone-----G729----FS-----G729-----PEER_A > > With PEER B, I want to use only G711, so: > Phone-----G711----FS-----G711-----PEER_B > > In fact, I'd like to force FS announcing the codec list priority based > on the priority of the codec announced by the PEER, cause FS is unable > to transcode G729 <--> G711. > > Tried a lot of things (greedy for codec-negociation, late_codec, > disable_transcoding, codec-prefs) without success. > > If you have some clue. > > regards, > rod > > Michael Collins a ?crit : > > Check out this page: > > http://wiki.freeswitch.org/wiki/Codec_negotiation > > > > Late negotiation will probably let you handle all the cases you > need. > > -MC > > > > On Mon, Aug 31, 2009 at 8:00 AM, rod > > >> wrote: > > > > Hi all, > > > > I'm wondering if I can do something like this: > > - in my internal profile, I have this because of some PEER > > using G729: > > - > > > > But for a specific PEER, I'd like to activate transcoding: > > - for this PEER, only G711 is used > > - I'd like to transcode DTMF SIP INFO or RFC2833 to > INBAND > > > > So in my dialplan, I tried before bridging: > > > > - data="disable-transcoding=false"/> > > - > > > > But I still see RFC2833 events between my FS and PEER and > the DTMF are > > not working. > > > > So 2 questions: > > - does application "start_dtmf_generate" requires transcoding > > - if yes, can I set the variable disable-transcoding in > my dialplan > > > > regards, > > rod > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From panayotov.vd at gmail.com Thu Sep 3 22:59:03 2009 From: panayotov.vd at gmail.com (Vassil Panayotov) Date: Fri, 4 Sep 2009 08:59:03 +0300 Subject: [Freeswitch-users] Python error: is a directory ?! Message-ID: <8a9b664c0909032259v39c5fd22j10700ec281d8848@mail.gmail.com> Hi, Sometimes when I try to restart FS, I get the error while the mod_python is loaded: Python error: is a directory, cannot continue In some cases the error doesn't disappear until system reboot. What causes such errors? Thank you, Vassil Panayotov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/cd55c500/attachment.html From christian.loeschenkohl at xpirio.com Thu Sep 3 23:03:26 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Fri, 04 Sep 2009 08:03:26 +0200 Subject: [Freeswitch-users] restart when convenient Message-ID: <4AA0ADAE.2090709@xpirio.com> hello i'm looking for a possibility to restart freeswitch like it is possible with asterisk. for me i tried to created a script that looks for open channels and if no channel is open it restarts freeswitch with the init script (not the most efficient way). i think i would be great if we would have a buildin function for this, i think such command would help with maintenance and not only for me. br -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com From shaheryarkh at googlemail.com Thu Sep 3 22:59:02 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 4 Sep 2009 10:59:02 +0500 Subject: [Freeswitch-users] XML Dial Plan vs Language Modules Message-ID: Hi, I couple of my team members are working on translating a very long Asterisk Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link below, http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Variables The dial plan variables are not getting initialized as expected. I was just wondering if we move this variable get and set stuff to any language module say mod_perl, will that make any difference performance wise? I mean we will be invoking a Perl interpreter for each incoming call, won't that be expensive in terms of RAM and CPU usage and thus reducing number of calls this FS deployment can handle? I have guys with programming skills in Perl, PHP, Python, Java and LUA languages. Which language do you recommend for this, again in terms of speed and performance? Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/088e21ce/attachment.html From ahmedmunir007 at gmail.com Thu Sep 3 23:05:25 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Fri, 4 Sep 2009 12:05:25 +0600 Subject: [Freeswitch-users] Passing Variables in FS Message-ID: Hi, I'm newbie in FS. As far as I know for setting up custom variables in FS we use this syntex in dialplan XML i.e. But when I call this variable using eval application i.e. the value I get from variable ABC is undefined means no values are passed to the variable. So kindly do let me know how I can pass values in variables in FS. -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/1cac0461/attachment.html From demuel at thephinix.org Thu Sep 3 23:19:46 2009 From: demuel at thephinix.org (demuel at thephinix.org) Date: Fri, 4 Sep 2009 07:19:46 +0100 Subject: [Freeswitch-users] XML Dial Plan vs Language Modules In-Reply-To: References: Message-ID: Why not make your gays have with all those programming skills try each one of them. Say, one guy programs in Perl, the other in PHP, still the other in Python, still again the other in Java and finally one in LUA. Take note, same dialplan project and let them not compare notes or translate the code of the other. Then at the end of the day, testing day, let the code produced be subjected to which one does the job well. Sure there could be one and those code that's not worthy enough well just do a "rm -rf " in your ranks. > Hi, > > I couple of my team members are working on translating a very long Asterisk > Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link below, > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Variables > > The dial plan variables are not getting initialized as expected. I was just > wondering if we move this variable get and set stuff to any language module > say mod_perl, will that make any difference performance wise? I mean we will > be invoking a Perl interpreter for each incoming call, won't that be > expensive in terms of RAM and CPU usage and thus reducing number of calls > this FS deployment can handle? > > I have guys with programming skills in Perl, PHP, Python, Java and LUA > languages. Which language do you recommend for this, again in terms of speed > and performance? > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jmesquita at freeswitch.org Thu Sep 3 23:35:50 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 4 Sep 2009 03:35:50 -0300 Subject: [Freeswitch-users] restart when convenient In-Reply-To: <4AA0ADAE.2090709@xpirio.com> References: <4AA0ADAE.2090709@xpirio.com> Message-ID: Look at the fsctl api on the wiki. It has what you need. jmesquita On 9/4/09, Christian L?schenkohl wrote: > hello > > i'm looking for a possibility to restart freeswitch like it is possible with > asterisk. > for me i tried to created a script that looks for open channels and if no > channel > is open it restarts freeswitch with the init script (not the most efficient > way). > > i think i would be great if we would have a buildin function for this, i > think such > command would help with maintenance and not only for me. > > br > > -- > Ing. Christian L?schenkohl > 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.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device From lfurrea at gmail.com Thu Sep 3 23:44:14 2009 From: lfurrea at gmail.com (Luis F Urrea) Date: Fri, 4 Sep 2009 00:44:14 -0600 Subject: [Freeswitch-users] FS on 5400zl Procurve module Message-ID: Has anyone out there has the opportunity to get hands on a 5400zl series Procurve? FS on the Intel based module That would be a sweet application!! As far as I understand applications need to undergo some testing before they can be run on the module. anyone can comment?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/e046f46c/attachment.html From nandy1925 at gmail.com Thu Sep 3 23:46:14 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Fri, 4 Sep 2009 14:46:14 +0800 Subject: [Freeswitch-users] Set disable-transcoding in dialplan In-Reply-To: <4AA0ABA0.5020901@laposte.net> References: <4A9BE5AB.4030304@laposte.net> <87f2f3b90908310938oa7e838dkd6e07579f224de37@mail.gmail.com> <4A9F6745.6030607@laposte.net> <7d0bfd8c0909031726t53f9900dldf7f2c0eab97305@mail.gmail.com> <4AA0ABA0.5020901@laposte.net> Message-ID: <7d0bfd8c0909032346u5b12614fuba5be2b1aa4f1fc@mail.gmail.com> rod, it looks more complicated now when PEER C comes to the picture. i think we'll have to wait for the availability of g729 on FS, as per Anthony's post. /nandy On Fri, Sep 4, 2009 at 1:54 PM, rod wrote: > Hi Nandy, > > yes already tried this, but if I use proxy_media=true, FS makes no > control on the content of the RTP stream. But the pbm is that I need to > use this: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate > This function enables transcoding of SIP_INFO or RFC2833 to Inband DTMF > in G711 > > But this feature doesn't work if I'm using proxy_media=true. > > In fact my setup is the following: > > CPE using G711A, G729 and SIP INFO for DTMF > PEER_A using G729 only and RFC_2833 > PEER_B using G711 and SIP INFO > > I have been able to make this works, with proxy_media=true for PEER_B > cause I don't need transcoding of DTMF (SIP INFO to SIP INFO). > For PEER_A, proxy_media is set to false (default) cause I need > transcoding SIP INFO to RFC2833. I'm able to use G729 using > codec_negotiation=greedy and setting G729 with highest priority on my > internal profile. > > But the pbm is that I need to add PEER_C. > PEER_C needs G711 with transcoding DTMF from SIP_INFO to Inband. > > And this is where I'm stuck, cause using "greedy settings and G729 with > priority 1 in my codec list and proxy_media=false" force FS to negotiate > G729 on leg A. But Leg B is willing to use G711 and FS is unable to > transcode G729 <---> G711. > > I was wondering if there is a way for FS to force the codec order on Leg > A with some knowledge of the preferred codec on Leg B, ie I know that > Leg B will always use G711 so that I want to biase the SDP answer on Leg > A based on this fact. > > regards, > rod > > Nandy Dagondon a ?crit : > > rod, > > > > have you tried this? > > > http://lists.freeswitch.org/pipermail/freeswitch-users/2008-March/002199.html > > > > /nandy > > > > > > On Thu, Sep 3, 2009 at 2:50 PM, rod > > wrote: > > > > Hi Michael, > > > > I did some tests but I haven't been successful, so there is what I'm > > trying to achieve: > > > > On A leg, my phone is using: PCMA and G729 (in this priority order) > > > > With PEER A, I want to use only G729 (thats is the only codec that > > this > > PEER support), so that the RTP flow will be: > > Phone-----G729----FS-----G729-----PEER_A > > > > With PEER B, I want to use only G711, so: > > Phone-----G711----FS-----G711-----PEER_B > > > > In fact, I'd like to force FS announcing the codec list priority > based > > on the priority of the codec announced by the PEER, cause FS is > unable > > to transcode G729 <--> G711. > > > > Tried a lot of things (greedy for codec-negociation, late_codec, > > disable_transcoding, codec-prefs) without success. > > > > If you have some clue. > > > > regards, > > rod > > > > Michael Collins a ?crit : > > > Check out this page: > > > http://wiki.freeswitch.org/wiki/Codec_negotiation > > > > > > Late negotiation will probably let you handle all the cases you > > need. > > > -MC > > > > > > On Mon, Aug 31, 2009 at 8:00 AM, rod > > > > >> wrote: > > > > > > Hi all, > > > > > > I'm wondering if I can do something like this: > > > - in my internal profile, I have this because of some PEER > > > using G729: > > > - > > > > > > But for a specific PEER, I'd like to activate transcoding: > > > - for this PEER, only G711 is used > > > - I'd like to transcode DTMF SIP INFO or RFC2833 to > > INBAND > > > > > > So in my dialplan, I tried before bridging: > > > > > > - > data="disable-transcoding=false"/> > > > - > > > > > > But I still see RFC2833 events between my FS and PEER and > > the DTMF are > > > not working. > > > > > > So 2 questions: > > > - does application "start_dtmf_generate" requires > transcoding > > > - if yes, can I set the variable disable-transcoding in > > my dialplan > > > > > > regards, > > > rod > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/1550c3ca/attachment-0001.html From msc at freeswitch.org Thu Sep 3 23:59:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 3 Sep 2009 23:59:05 -0700 Subject: [Freeswitch-users] Passing Variables in FS In-Reply-To: References: Message-ID: <87f2f3b90909032359p399ff13dq5ec10070d01b3380@mail.gmail.com> On Thu, Sep 3, 2009 at 11:05 PM, Ahmed Munir wrote: > Hi, > > I'm newbie in FS. As far as I know for setting up custom variables in FS we > use this syntex in dialplan XML i.e. > > > > But when I call this variable using eval application i.e. > > > > the value I get from variable ABC is undefined means no values are passed > to the variable. > > So kindly do let me know how I can pass values in variables in FS. > > Try using the info app instead of eval. Most likely this is just a case of the dialplan being parsed prior to the variable being assigned. Do this in your dialplan after you set the variable: You'll see a whole list of variables and your custom variable(s) should be shown with their values. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090903/4b3c0708/attachment.html From msc at freeswitch.org Fri Sep 4 00:06:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 Sep 2009 00:06:29 -0700 Subject: [Freeswitch-users] XML Dial Plan vs Language Modules In-Reply-To: References: Message-ID: <87f2f3b90909040006s15c1f733lff93b4549dee6da7@mail.gmail.com> On Thu, Sep 3, 2009 at 10:59 PM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > Hi, > > I couple of my team members are working on translating a very long Asterisk > Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link below, > Before you go through all the trouble of translating the dialplan be sure to review the application itself. In many cases just doing a dialplan translation results in less efficient use of FreeSWITCH's powerful features. Be sure that you are looking at the way FreeSWITCH handles various situations and take advantage of its power and ease of use. > > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Variables > > The dial plan variables are not getting initialized as expected. I was just > wondering if we move this variable get and set stuff to any language module > say mod_perl, will that make any difference performance wise? I mean we will > be invoking a Perl interpreter for each incoming call, won't that be > expensive in terms of RAM and CPU usage and thus reducing number of calls > this FS deployment can handle? > > I have guys with programming skills in Perl, PHP, Python, Java and LUA > languages. Which language do you recommend for this, again in terms of speed > and performance? > > Lua is very portable and we've done tests with hundreds of concurrent Lua scripts running. The other languages are heavier but they'll still handle quite a few concurrent sessions. Just be sure that you don't do the bridge app right in the script, use transfer instead and have the dialplan process any bridging that you need to do. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/ec3dbf2a/attachment.html From msc at freeswitch.org Fri Sep 4 00:14:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 Sep 2009 00:14:03 -0700 Subject: [Freeswitch-users] restart when convenient In-Reply-To: <4AA0ADAE.2090709@xpirio.com> References: <4AA0ADAE.2090709@xpirio.com> Message-ID: <87f2f3b90909040014r5c3dfa7ai9b5bbd2413107ec6@mail.gmail.com> 2009/9/3 Christian L?schenkohl > hello > > i'm looking for a possibility to restart freeswitch like it is possible > with > asterisk. > for me i tried to created a script that looks for open channels and if no > channel > is open it restarts freeswitch with the init script (not the most efficient > way). > > i think i would be great if we would have a buildin function for this, i > think such > command would help with maintenance and not only for me. > > br > > Like JM said, the fsctl API can help. If you're in Linux you can do a shell script with a command like ths: fs_cli -x 'fsctl shutdown restart' -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/91606f9d/attachment.html From shaheryarkh at googlemail.com Fri Sep 4 00:25:04 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 4 Sep 2009 12:25:04 +0500 Subject: [Freeswitch-users] XML Dial Plan vs Language Modules In-Reply-To: <87f2f3b90909040006s15c1f733lff93b4549dee6da7@mail.gmail.com> References: <87f2f3b90909040006s15c1f733lff93b4549dee6da7@mail.gmail.com> Message-ID: Thank you so much. Of course we are not doing a blind translation, but at the very basic we will need to get and set certain variable at different stage of call processing. Another question in same context, Can we do post-hangup call processing? I mean like in Asterisk, we have extension "h" which is called after hangup. Can you guide a bit how to do it in FS? Does FS has any such special extensions? Thank you. On Fri, Sep 4, 2009 at 12:06 PM, Michael Collins wrote: > > > On Thu, Sep 3, 2009 at 10:59 PM, Muhammad Shahzad < > shaheryarkh at googlemail.com> wrote: > >> Hi, >> >> I couple of my team members are working on translating a very long >> Asterisk Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link >> below, >> > > Before you go through all the trouble of translating the dialplan be sure > to review the application itself. In many cases just doing a dialplan > translation results in less efficient use of FreeSWITCH's powerful features. > Be sure that you are looking at the way FreeSWITCH handles various > situations and take advantage of its power and ease of use. > >> >> >> http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Variables >> >> The dial plan variables are not getting initialized as expected. I was >> just wondering if we move this variable get and set stuff to any language >> module say mod_perl, will that make any difference performance wise? I mean >> we will be invoking a Perl interpreter for each incoming call, won't that be >> expensive in terms of RAM and CPU usage and thus reducing number of calls >> this FS deployment can handle? >> >> I have guys with programming skills in Perl, PHP, Python, Java and LUA >> languages. Which language do you recommend for this, again in terms of speed >> and performance? >> >> > Lua is very portable and we've done tests with hundreds of concurrent Lua > scripts running. The other languages are heavier but they'll still handle > quite a few concurrent sessions. Just be sure that you don't do the bridge > app right in the script, use transfer instead and have the dialplan process > any bridging that you need to do. > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/56f4efa7/attachment.html From msc at freeswitch.org Fri Sep 4 00:32:50 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 Sep 2009 00:32:50 -0700 Subject: [Freeswitch-users] XML Dial Plan vs Language Modules In-Reply-To: References: <87f2f3b90909040006s15c1f733lff93b4549dee6da7@mail.gmail.com> Message-ID: <87f2f3b90909040032p76c3f8acvdbfe8b6ce13855b8@mail.gmail.com> On Fri, Sep 4, 2009 at 12:25 AM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > Thank you so much. Of course we are not doing a blind translation, but at > the very basic we will need to get and set certain variable at different > stage of call processing. > > Another question in same context, Can we do post-hangup call processing? I > mean like in Asterisk, we have extension "h" which is called after hangup. > Can you guide a bit how to do it in FS? Does FS has any such special > extensions? > > Thank you. > > Yes, you can post hangup processing. See the wiki channel_variables page and look at "api_hangup_hook" for more information. Just know that it can get tricky to try and post-process calls from right inside the dialplan. In most cases we recommend using the event socket and having absolute control over the call, including what happens at hangup. Lua is especially good at this. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/54f43a14/attachment.html From anatoliy at kounitskiy.com Fri Sep 4 00:35:59 2009 From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy) Date: Fri, 4 Sep 2009 10:35:59 +0300 Subject: [Freeswitch-users] restart when convenient In-Reply-To: <87f2f3b90909040014r5c3dfa7ai9b5bbd2413107ec6@mail.gmail.com> References: <4AA0ADAE.2090709@xpirio.com> <87f2f3b90909040014r5c3dfa7ai9b5bbd2413107ec6@mail.gmail.com> Message-ID: <1cd828b60909040035k264a1e0bw16f8d1771d3ac931@mail.gmail.com> After some testing (fs_cli -x 'fsctl shutdown restart') I'm seeing that all active calls are dropped and the freeswitch is restarted On Fri, Sep 4, 2009 at 10:14 AM, Michael Collins wrote: > > > 2009/9/3 Christian L?schenkohl >> >> hello >> >> i'm looking for a possibility to restart freeswitch like it is possible >> with >> asterisk. >> for me i tried to created a script that looks for open channels and if no >> channel >> is open it restarts freeswitch with the init script (not the most >> efficient way). >> >> i think i would be great if we would have a buildin function for this, i >> think such >> command would help with maintenance and not only for me. >> >> br >> > Like JM said, the fsctl API can help. If you're in Linux you can do a shell > script with a command like ths: > > fs_cli -x 'fsctl shutdown restart' > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy at kounitskiy.com Mobile: +359898913540 From dujinfang at gmail.com Fri Sep 4 00:46:17 2009 From: dujinfang at gmail.com (Seven Du) Date: Fri, 4 Sep 2009 15:46:17 +0800 Subject: [Freeswitch-users] restart when convenient In-Reply-To: <1cd828b60909040035k264a1e0bw16f8d1771d3ac931@mail.gmail.com> References: <4AA0ADAE.2090709@xpirio.com> <87f2f3b90909040014r5c3dfa7ai9b5bbd2413107ec6@mail.gmail.com> <1cd828b60909040035k264a1e0bw16f8d1771d3ac931@mail.gmail.com> Message-ID: freeswitch at foosball> fsctl -USAGE: [send_sighup|hupall|pause|resume|shutdown [cancel|elegant|asap| restart]|sps|sync_clock|reclaim_mem|max_sessions|max_dtmf_duration [num]|loglevel [level]] On Sep 4, 2009, at 3:35 PM, Anatoliy Kounitskiy wrote: > After some testing (fs_cli -x 'fsctl shutdown restart') I'm seeing > that all active calls are dropped and the freeswitch is restarted > > > > On Fri, Sep 4, 2009 at 10:14 AM, Michael Collins > wrote: >> >> >> 2009/9/3 Christian L?schenkohl >>> >>> hello >>> >>> i'm looking for a possibility to restart freeswitch like it is >>> possible >>> with >>> asterisk. >>> for me i tried to created a script that looks for open channels >>> and if no >>> channel >>> is open it restarts freeswitch with the init script (not the most >>> efficient way). >>> >>> i think i would be great if we would have a buildin function for >>> this, i >>> think such >>> command would help with maintenance and not only for me. >>> >>> br >>> >> Like JM said, the fsctl API can help. If you're in Linux you can do >> a shell >> script with a command like ths: >> >> fs_cli -x 'fsctl shutdown restart' >> >> -MC >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > Anatoliy Kounitskiy > ------------------------- > E-mail: anatoliy at kounitskiy.com > Mobile: +359898913540 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jingwei.yang at gmail.com Fri Sep 4 01:01:14 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Fri, 4 Sep 2009 16:01:14 +0800 Subject: [Freeswitch-users] skypiax error Message-ID: <13529f9d0909040101i2ea5ae66n8dfe9c6a89236ef0@mail.gmail.com> Hi Folks, I just tried to install FS with mod_skypiax on a new CentOS 5.2 machine. After having followed the big help doc from the wiki page ( http://wiki.freeswitch.org/wiki/Skypiax#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS.2C_from_scratch), I hit an error when running multi.sh (under freeswitch/src/mod/endpoints/mod_skypiax/configs/multiple-instance-same-skype-username). *Couldn't open RGB_DB '/usr/share/X11/rgb' error opening security policy file /usr/lib64/xserver/SecurityPolicy * This error seems not stopping xvfb from getting started. Then I started FS and loaded mod_skypiax. However, when I initiated a test call (originate skypiax/ANY/userAAA &echo), I saw a bunch of ALSA lib errors popping up: *ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi* Again, this error doesn't prohibit the call from reaching me. It's just too annoying and it keeps popping up after a while. Does anyone know how to get rid of those errors? I found a similar post here: http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/013956.html. Phil, if you happen to see my question, could you please reply and let me know what the cause was and what you have done to solve it? Thanks! By the way, I started xvfb and FS using root. Regards, -Jingwei -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/6c9c7cb4/attachment.html From gmaruzz at celliax.org Fri Sep 4 01:25:40 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 4 Sep 2009 10:25:40 +0200 Subject: [Freeswitch-users] skypiax error In-Reply-To: <13529f9d0909040101i2ea5ae66n8dfe9c6a89236ef0@mail.gmail.com> References: <13529f9d0909040101i2ea5ae66n8dfe9c6a89236ef0@mail.gmail.com> Message-ID: <7b197bef0909040125g627a684epc48f965268b4415d@mail.gmail.com> Jingwei, those are normal warnings made by the Skype client (not by mod_skypiax), you just have to edit /etc/alsa/alsa.conf and comment out hdmi lines. Is a problem with a lazy implementation of that file, that supposes you got an hdmi. The other warning is because there are some files missing from the Xvfb installation made by centos, but are completely harmless. In the future I will make the script to redirect them to /dev/null :-) Bottom line: all is OK. -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 On Fri, Sep 4, 2009 at 10:01 AM, Jingwei Yang wrote: > Hi Folks, > > I just tried to install FS with mod_skypiax on a new CentOS 5.2 machine. > After having followed the big help doc from the wiki page > (http://wiki.freeswitch.org/wiki/Skypiax#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS.2C_from_scratch), > I hit an error when running multi.sh (under > freeswitch/src/mod/endpoints/mod_skypiax/configs/multiple-instance-same-skype-username). > > Couldn't open RGB_DB '/usr/share/X11/rgb' > error opening security policy file /usr/lib64/xserver/SecurityPolicy > > This error seems not stopping xvfb from getting started. Then I started FS > and loaded mod_skypiax. However, when I initiated a test call (originate > skypiax/ANY/userAAA &echo), I saw a bunch of ALSA lib errors popping up: > > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > > Again, this error doesn't prohibit the call from reaching me. It's just too > annoying and it keeps popping up after a while. Does anyone know how to get > rid of those errors? > > I found a similar post here: > http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/013956.html. > Phil, if you happen to see my question, could you please reply and let me > know what the cause was and what you have done to solve it? Thanks! > > By the way, I started xvfb and FS using root. > > Regards, > -Jingwei > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From jingwei.yang at gmail.com Fri Sep 4 01:38:06 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Fri, 4 Sep 2009 16:38:06 +0800 Subject: [Freeswitch-users] skypiax error In-Reply-To: <7b197bef0909040125g627a684epc48f965268b4415d@mail.gmail.com> References: <13529f9d0909040101i2ea5ae66n8dfe9c6a89236ef0@mail.gmail.com> <7b197bef0909040125g627a684epc48f965268b4415d@mail.gmail.com> Message-ID: <13529f9d0909040138hcf63eane312c42d32e4f38b@mail.gmail.com> Hi Giovanni, That's a big relief. Thanks a lot for the reply :) Regards, -Jingwei On Fri, Sep 4, 2009 at 4:25 PM, Giovanni Maruzzelli wrote: > Jingwei, > > those are normal warnings made by the Skype client (not by > mod_skypiax), you just have to edit /etc/alsa/alsa.conf and comment > out hdmi lines. Is a problem with a lazy implementation of that file, > that supposes you got an hdmi. > > The other warning is because there are some files missing from the > Xvfb installation made by centos, but are completely harmless. In the > future I will make the script to redirect them to /dev/null :-) > > Bottom line: all is OK. > > -giovanni > > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > > > On Fri, Sep 4, 2009 at 10:01 AM, Jingwei Yang > wrote: > > Hi Folks, > > > > I just tried to install FS with mod_skypiax on a new CentOS 5.2 machine. > > After having followed the big help doc from the wiki page > > ( > http://wiki.freeswitch.org/wiki/Skypiax#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS.2C_from_scratch > ), > > I hit an error when running multi.sh (under > > > freeswitch/src/mod/endpoints/mod_skypiax/configs/multiple-instance-same-skype-username). > > > > Couldn't open RGB_DB '/usr/share/X11/rgb' > > error opening security policy file /usr/lib64/xserver/SecurityPolicy > > > > This error seems not stopping xvfb from getting started. Then I started > FS > > and loaded mod_skypiax. However, when I initiated a test call (originate > > skypiax/ANY/userAAA &echo), I saw a bunch of ALSA lib errors popping up: > > > > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi > > > > Again, this error doesn't prohibit the call from reaching me. It's just > too > > annoying and it keeps popping up after a while. Does anyone know how to > get > > rid of those errors? > > > > I found a similar post here: > > > http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/013956.html > . > > Phil, if you happen to see my question, could you please reply and let me > > know what the cause was and what you have done to solve it? Thanks! > > > > By the way, I started xvfb and FS using root. > > > > Regards, > > -Jingwei > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/27a086a5/attachment.html From dmitry.bely at gmail.com Fri Sep 4 01:43:59 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Fri, 4 Sep 2009 12:43:59 +0400 Subject: [Freeswitch-users] Proxy authorization In-Reply-To: <7B180BCD-82F4-43BD-A8C9-48CC2A38D899@freeswitch.org> References: <90823c940909031326g5e9d5a7bsd2650a59a7a5eaf@mail.gmail.com> <7B180BCD-82F4-43BD-A8C9-48CC2A38D899@freeswitch.org> Message-ID: <90823c940909040143ie0c175fy5702a9c280676d28@mail.gmail.com> On Fri, Sep 4, 2009 at 1:19 AM, Brian West wrote: > There will not be an authorization header on the first register attempt... > it only happens once we are 401/407'ed and the phone comes back and > registers again. > /b Alas, I cannot change the way the provider's gateway works. It immediately responses with 403... BTW, it's Mera Damos (http://www.mera-systems.com ?). No workaround possible? > On Sep 3, 2009, at 3:26 PM, Dmitry Bely wrote: > > Unfortunately even after that there is no "Authorization:" header in > the REGISTER message: > > ??REGISTER?sip:1.2.3.4?SIP/2.0 > ??Via: SIP/2.0/UDP 5.6.7.8:5080;rport;branch=z9hG4bKNBB3ygD85y3eF > ??Max-Forwards: 70 > ??From: ;tag=Nrc6Z9yrNBS3H > ??To: > ??Call-ID: a93d949a-98c1-11de-b6b8-8321249ad8d4 > ??CSeq: 119885384 REGISTER > ??Contact: > ??Expires: 600 > ??User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14707M > ??Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > ??Supported: timer, precondition, path, replaces > ??Content-Length: 0 - Dmitry Bely From gmaruzz at celliax.org Fri Sep 4 01:56:54 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 4 Sep 2009 10:56:54 +0200 Subject: [Freeswitch-users] skypiax error In-Reply-To: <13529f9d0909040138hcf63eane312c42d32e4f38b@mail.gmail.com> References: <13529f9d0909040101i2ea5ae66n8dfe9c6a89236ef0@mail.gmail.com> <7b197bef0909040125g627a684epc48f965268b4415d@mail.gmail.com> <13529f9d0909040138hcf63eane312c42d32e4f38b@mail.gmail.com> Message-ID: <7b197bef0909040156j5aa66a29sac58f25f26acb14b@mail.gmail.com> :-) My fault, I would have to document this. I'll do pretty soon. Sorry about that, and thanks for reporting!!! -gm Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 On Fri, Sep 4, 2009 at 10:38 AM, Jingwei Yang wrote: > Hi Giovanni, > > That's a big relief. Thanks a lot for the reply :) > > Regards, > -Jingwei > > On Fri, Sep 4, 2009 at 4:25 PM, Giovanni Maruzzelli > wrote: >> >> Jingwei, >> >> those are normal warnings made by the Skype client (not by >> mod_skypiax), you just have to edit /etc/alsa/alsa.conf and comment >> out hdmi lines. Is a problem with a lazy implementation of that file, >> that supposes you got an hdmi. >> >> The other warning is because there are some files missing from the >> Xvfb installation made by centos, but are completely harmless. In the >> future I will make the script to redirect them to /dev/null :-) >> >> Bottom line: all is OK. >> >> -giovanni >> >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> >> >> >> On Fri, Sep 4, 2009 at 10:01 AM, Jingwei Yang >> wrote: >> > Hi Folks, >> > >> > I just tried to install FS with mod_skypiax on a new CentOS 5.2 machine. >> > After having followed the big help doc from the wiki page >> > >> > (http://wiki.freeswitch.org/wiki/Skypiax#An_example_of_Skypiax_and_FreeSWITCH_installation_on_CentOS.2C_from_scratch), >> > I hit an error when running multi.sh (under >> > >> > freeswitch/src/mod/endpoints/mod_skypiax/configs/multiple-instance-same-skype-username). >> > >> > Couldn't open RGB_DB '/usr/share/X11/rgb' >> > error opening security policy file /usr/lib64/xserver/SecurityPolicy >> > >> > This error seems not stopping xvfb from getting started. Then I started >> > FS >> > and loaded mod_skypiax. However, when I initiated a test call (originate >> > skypiax/ANY/userAAA &echo), I saw a bunch of ALSA lib errors popping up: >> > >> > ALSA lib pcm.c:2184:(snd_pcm_open_noupdate) Unknown PCM cards.pcm.hdmi >> > >> > Again, this error doesn't prohibit the call from reaching me. It's just >> > too >> > annoying and it keeps popping up after a while. Does anyone know how to >> > get >> > rid of those errors? >> > >> > I found a similar post here: >> > >> > http://lists.freeswitch.org/pipermail/freeswitch-users/2009-May/013956.html. >> > Phil, if you happen to see my question, could you please reply and let >> > me >> > know what the cause was and what you have done to solve it? Thanks! >> > >> > By the way, I started xvfb and FS using root. >> > >> > Regards, >> > -Jingwei >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anatoliy at kounitskiy.com Fri Sep 4 02:03:18 2009 From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy) Date: Fri, 4 Sep 2009 12:03:18 +0300 Subject: [Freeswitch-users] restart when convenient In-Reply-To: References: <4AA0ADAE.2090709@xpirio.com> <87f2f3b90909040014r5c3dfa7ai9b5bbd2413107ec6@mail.gmail.com> <1cd828b60909040035k264a1e0bw16f8d1771d3ac931@mail.gmail.com> Message-ID: <1cd828b60909040203of05e0d3u9fea57834d96bb98@mail.gmail.com> What you're looking for is : fs_cli -x 'fsctl shutdown elegant restart' :) It will restart the freeswitch after all calls are hanged up. On Fri, Sep 4, 2009 at 10:46 AM, Seven Du wrote: > > freeswitch at foosball> fsctl > -USAGE: [send_sighup|hupall|pause|resume|shutdown [cancel|elegant|asap| > restart]|sps|sync_clock|reclaim_mem|max_sessions|max_dtmf_duration > [num]|loglevel [level]] > > > On Sep 4, 2009, at 3:35 PM, Anatoliy Kounitskiy wrote: >> After some testing (fs_cli -x 'fsctl shutdown restart') I'm seeing >> that all active calls are dropped and the freeswitch is restarted >> >> >> >> On Fri, Sep 4, 2009 at 10:14 AM, Michael Collins >> wrote: >>> >>> >>> 2009/9/3 Christian L?schenkohl >>>> >>>> hello >>>> >>>> i'm looking for a possibility to restart freeswitch like it is >>>> possible >>>> with >>>> asterisk. >>>> for me i tried to created a script that looks for open channels >>>> and if no >>>> channel >>>> is open it restarts freeswitch with the init script (not the most >>>> efficient way). >>>> >>>> i think i would be great if we would have a buildin function for >>>> this, i >>>> think such >>>> command would help with maintenance and not only for me. >>>> >>>> br >>>> >>> Like JM said, the fsctl API can help. If you're in Linux you can do >>> a shell >>> script with a command like ths: >>> >>> fs_cli -x 'fsctl shutdown restart' >>> >>> -MC >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> >> -- >> Anatoliy Kounitskiy >> ------------------------- >> E-mail: anatoliy at kounitskiy.com >> Mobile: +359898913540 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy at kounitskiy.com Mobile: +359898913540 From gmaruzz at celliax.org Fri Sep 4 02:47:41 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 4 Sep 2009 11:47:41 +0200 Subject: [Freeswitch-users] skypiax error In-Reply-To: <7b197bef0909040156j5aa66a29sac58f25f26acb14b@mail.gmail.com> References: <13529f9d0909040101i2ea5ae66n8dfe9c6a89236ef0@mail.gmail.com> <7b197bef0909040125g627a684epc48f965268b4415d@mail.gmail.com> <13529f9d0909040138hcf63eane312c42d32e4f38b@mail.gmail.com> <7b197bef0909040156j5aa66a29sac58f25f26acb14b@mail.gmail.com> Message-ID: <7b197bef0909040247vd74dfb2o718cb25fbac46180@mail.gmail.com> On Fri, Sep 4, 2009 at 10:56 AM, Giovanni Maruzzelli wrote: > :-) My fault, I would have to document this. http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#Error_and_warnings_at_the_starting_of_Skype_clients_on_Linux -giovanni From Prometheus001 at gmx.net Fri Sep 4 02:51:28 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 04 Sep 2009 11:51:28 +0200 Subject: [Freeswitch-users] mod_xml_curl.c Oversized file detected [1056100 bytes] In-Reply-To: <191c3a030909031130n453786ccyd71ee0b9f92a3955@mail.gmail.com> References: <4A9FFD64.2000701@gmx.net> <191c3a030909031130n453786ccyd71ee0b9f92a3955@mail.gmail.com> Message-ID: <4AA0E320.8000708@gmx.net> Thanks Anthony, that did the trick. Best regards Peter Anthony Minessale schrieb: > you can edit mod_xml_curl.c line 64 > and increase XML_CURL_MAX_BYTES > > > On Thu, Sep 3, 2009 at 12:31 PM, Peter P GMX > wrote: > > Hello, > > in a B2BUA scenario we have 2000 defined gateways (defined but not > registered yet). > When reloading mod_sofia Freeswitch complains about the XML-Curl File > size > 1MB and deactivates all gateways: > mod_xml_curl.c:121 Oversized file detected [1056100 bytes] > > Is there any way to overcome this? Currently we have 2000 gateways > defined. Finally we will have about 10.000. And we will not be able to > reduce the file size below 1 MB. It will become ~ 2-3 MB maybe. > > Best Regards > Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jingwei.yang at gmail.com Fri Sep 4 02:59:10 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Fri, 4 Sep 2009 17:59:10 +0800 Subject: [Freeswitch-users] skypiax error In-Reply-To: <7b197bef0909040247vd74dfb2o718cb25fbac46180@mail.gmail.com> References: <13529f9d0909040101i2ea5ae66n8dfe9c6a89236ef0@mail.gmail.com> <7b197bef0909040125g627a684epc48f965268b4415d@mail.gmail.com> <13529f9d0909040138hcf63eane312c42d32e4f38b@mail.gmail.com> <7b197bef0909040156j5aa66a29sac58f25f26acb14b@mail.gmail.com> <7b197bef0909040247vd74dfb2o718cb25fbac46180@mail.gmail.com> Message-ID: <13529f9d0909040259j3ab0a11t2a8ac8a80971b18@mail.gmail.com> That's efficient :) By the way, do you have any idea about this warning? ALSA lib pcm_dmix.c:1008:(snd_pcm_dmix_open) unable to open slave On Fri, Sep 4, 2009 at 5:47 PM, Giovanni Maruzzelli wrote: > On Fri, Sep 4, 2009 at 10:56 AM, Giovanni Maruzzelli > wrote: > > :-) My fault, I would have to document this. > > > http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk#Error_and_warnings_at_the_starting_of_Skype_clients_on_Linux > > -giovanni > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/1922a360/attachment.html From gmaruzz at celliax.org Fri Sep 4 03:21:21 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 4 Sep 2009 12:21:21 +0200 Subject: [Freeswitch-users] skypiax error In-Reply-To: <7b197bef0909040247vd74dfb2o718cb25fbac46180@mail.gmail.com> References: <13529f9d0909040101i2ea5ae66n8dfe9c6a89236ef0@mail.gmail.com> <7b197bef0909040125g627a684epc48f965268b4415d@mail.gmail.com> <13529f9d0909040138hcf63eane312c42d32e4f38b@mail.gmail.com> <7b197bef0909040156j5aa66a29sac58f25f26acb14b@mail.gmail.com> <7b197bef0909040247vd74dfb2o718cb25fbac46180@mail.gmail.com> Message-ID: <7b197bef0909040321s1400f31cwde4db0fd7ad85e65@mail.gmail.com> Updated the wiki page with references to other errors/warnings as well :-) -giovanni From mayamatakeshi at gmail.com Fri Sep 4 03:28:23 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Fri, 4 Sep 2009 19:28:23 +0900 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> Message-ID: <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi wrote: > Hello, > I'm testing FS support for the header Path (FS is behind opensips). > It pretty much works: I tested calling from one user to the other and calls > work perfectly. > However, I've noticed that when I register my terminal directly with FS > without going thru the proxy, I receive an unsolicited NOTIFY containing > Message-Waiting information. But when I register via proxy, FS doesn't send > this NOTIFY. > What could be causing this difference of behavior? (enabling debug (F8) > doesn't show anything for registration handling). > I have enabled Sofia debug and I can see NTA is complaining about invalid URI when building the NOTIFY: nua: nua_notify: entering nua(0x9b3c1e8): sent signal r_notify nua(0x9b3c1e8): recv signal r_notify nua: nua_stack_set_params: entering nua(0x9b3c1e8): adding notify usage with event message-summary nta_leg_tcreate(0x9b74c68) nta outgoing create: invalid URI nta: outgoing_free(0x9b74928) nua(0x9b3c1e8): event r_notify 900 Internal error at nua_client.c:711 nua(0x9b3c1e8): removing notify usage with event message-summary My REGISTER relayed by opensips is this: REGISTER sip:test.com SIP/2.0 Record-Route: Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKe62c.49e9f6c1.0 Via: SIP/2.0/UDP 192.168.2.121:5060 ;received=192.168.2.121;rport=5060;branch=z9hG4bKPj4uAYgDuRbilYy4lCWcjlDKIDAtf-9RdS Max-Forwards: 69 From: >;tag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5 To: > Call-ID: JvQ.apMLiJtfHa7z4ShIfgBPi5jIbtBv CSeq: 14872 REGISTER Contact: Expires: 60 Authorization: Digest username="user1", realm="test.com", nonce="7d911eef-2c16-4deb-99f6-afcff9968a19", uri="sip:192.168.2.100", response="df29caeb78790b4527f1176622cbf192", algorithm=MD5, cnonce="5.EXCbM3RZTx6iOh1cvUzUvEZTs2eheG", qop=auth, nc=00000001 Content-Length: 0 Path: ;lr;received=sip:192.168.2.121:5060> I hope someone can point out a problem. I'm looking at NTA with gdb but I'm slow on this. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/a15a28a1/attachment.html From dmitry.bely at gmail.com Fri Sep 4 04:26:39 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Fri, 4 Sep 2009 15:26:39 +0400 Subject: [Freeswitch-users] Set disable-transcoding in dialplan In-Reply-To: <4AA0ABA0.5020901@laposte.net> References: <4A9BE5AB.4030304@laposte.net> <87f2f3b90908310938oa7e838dkd6e07579f224de37@mail.gmail.com> <4A9F6745.6030607@laposte.net> <7d0bfd8c0909031726t53f9900dldf7f2c0eab97305@mail.gmail.com> <4AA0ABA0.5020901@laposte.net> Message-ID: <90823c940909040426s23c36c61vf4d14da939247ff@mail.gmail.com> I had a similar problem when I needed to talk to a gateway using g729 while g711 was used by default. The following works for me: vars.xml (...) sip_profiles/internal.xml (...) dialplan/default/01_example.com.xml (...) On Fri, Sep 4, 2009 at 9:54 AM, rod wrote: > Hi Nandy, > > yes already tried this, but if I use proxy_media=true, FS makes no > control on the content of the RTP stream. But the pbm is that I need to > use this: > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate > This function enables transcoding of SIP_INFO or RFC2833 to Inband DTMF > in G711 > > But this feature doesn't work if I'm using proxy_media=true. > > In fact my setup is the following: > > CPE using G711A, G729 and SIP INFO for DTMF > PEER_A using G729 only and RFC_2833 > PEER_B using G711 and SIP INFO > > I have been able to make this works, with proxy_media=true for PEER_B > cause I don't need transcoding of DTMF (SIP INFO to SIP INFO). > For PEER_A, proxy_media is set to false (default) cause ?I need > transcoding SIP INFO to RFC2833. I'm able to use G729 using > codec_negotiation=greedy and setting G729 with highest priority on my > internal profile. > > But the pbm is that I need to add PEER_C. > PEER_C needs G711 with transcoding DTMF from SIP_INFO to Inband. > > And this is where I'm stuck, cause using "greedy settings and G729 with > priority 1 in my codec list and proxy_media=false" force FS to negotiate > G729 on leg A. But Leg B is willing to use G711 and FS is unable to > transcode G729 <---> G711. > > I was wondering if there is a way for FS to force the codec order on Leg > A with some knowledge of the preferred codec on Leg B, ie I know that > Leg B will always use G711 so that I want to biase the SDP answer on Leg > A based on this fact. > > regards, > rod > > Nandy Dagondon a ?crit : >> rod, >> >> have you tried this? >> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-March/002199.html >> >> /nandy >> >> >> On Thu, Sep 3, 2009 at 2:50 PM, rod > > wrote: >> >> ? ? Hi Michael, >> >> ? ? I did some tests but I haven't been successful, so there is what I'm >> ? ? trying to achieve: >> >> ? ? On A leg, my phone is using: PCMA and G729 (in this priority order) >> >> ? ? With PEER A, I want to use only G729 (thats is the only codec that >> ? ? this >> ? ? PEER support), so that the RTP flow will be: >> ? ? ? ?Phone-----G729----FS-----G729-----PEER_A >> >> ? ? With PEER B, I want to use only G711, so: >> ? ? ? ?Phone-----G711----FS-----G711-----PEER_B >> >> ? ? In fact, I'd like to force FS announcing the codec list priority based >> ? ? on the priority of the codec announced by the PEER, cause FS is unable >> ? ? to transcode G729 <--> G711. >> >> ? ? Tried a lot of things (greedy for codec-negociation, late_codec, >> ? ? disable_transcoding, codec-prefs) without success. >> >> ? ? If you have some clue. >> >> ? ? regards, >> ? ? rod >> >> ? ? Michael Collins a ?crit : >> ? ? > Check out this page: >> ? ? > http://wiki.freeswitch.org/wiki/Codec_negotiation >> ? ? > >> ? ? > Late negotiation will probably let you handle all the cases you >> ? ? need. >> ? ? > -MC >> ? ? > >> ? ? > On Mon, Aug 31, 2009 at 8:00 AM, rod > ? ? >> ? ? > >> wrote: >> ? ? > >> ? ? > ? ? Hi all, >> ? ? > >> ? ? > ? ? I'm wondering if I can do something like this: >> ? ? > ? ? ? ?- in my internal profile, I have this because of some PEER >> ? ? > ? ? using G729: >> ? ? > ? ? ? ? ? ? ?- >> ? ? > >> ? ? > ? ? But for a specific PEER, I'd like to activate transcoding: >> ? ? > ? ? ? ? ? ? ?- for this PEER, only G711 is used >> ? ? > ? ? ? ? ? ? ?- I'd like to transcode DTMF SIP INFO or RFC2833 to >> ? ? INBAND >> ? ? > >> ? ? > ? ? So in my dialplan, I tried before bridging: >> ? ? > >> ? ? > ? ? ? ?- > ? ? data="disable-transcoding=false"/> >> ? ? > ? ? ? ?- >> ? ? > >> ? ? > ? ? But I still see RFC2833 events between my FS and PEER and >> ? ? the DTMF are >> ? ? > ? ? not working. >> ? ? > >> ? ? > ? ? So 2 questions: >> ? ? > ? ? ? ?- does application "start_dtmf_generate" requires transcoding >> ? ? > ? ? ? ?- if yes, can I set the variable disable-transcoding in >> ? ? my dialplan >> ? ? > >> ? ? > ? ? regards, >> ? ? > ? ? rod - Dmitry Bely From shahzad at vopium.com Fri Sep 4 05:35:29 2009 From: shahzad at vopium.com (Muhammad Shahzad) Date: Fri, 4 Sep 2009 17:35:29 +0500 Subject: [Freeswitch-users] mod_dingaling: dl_login command syntax Message-ID: Hi, i am have FS SVN revision 14760, i am trying to use mod_xml_curl against mod_dingaling. When i call xml_curl url in browser i get mod_dingaling configuration correctly, also when i do reload mod_dingaling it fetches its configuration from xml_curl correctly. BUT when i try to use dl_login command to login a jingle profile it does not work. I have tried both syntax, Syntax 1: ======= dl_login profile=abcd Where abcd is a valid jingle profile fetch-able from xml_curl. Syntax 2: ======= dl_login name=abcd;login= XXX at gmail.com/talk;pass=YYY;dialplan=XML;context=public;rtp-ip=auto;sasl=plain;tls=true;exten=1001 All these values are correct and work if i reload mod_dingaling but they don't work with dl_login, and give following output. USAGE: Existing Profile: dl_login profile= Dynamic Profile: dl_login var1=val1;var2=val2;varN=valN I don't think xml_curl has any role in this syntax. Can you please correct me if i am doing something wrong in here or is it a bug in mod_dingaling. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/8eec0e7a/attachment-0001.html From fraunhofer.lists.freeswitch-001 at traced.net Fri Sep 4 05:42:11 2009 From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer) Date: Fri, 4 Sep 2009 14:42:11 +0200 Subject: [Freeswitch-users] memory leak In-Reply-To: <191c3a030909020842j61369a45va923080ea0a88b40@mail.gmail.com> References: <86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com> <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org> <191c3a030909020811q6c04518au514fe8e9d6c8734e@mail.gmail.com> <191c3a030909020815k6006f96fh3df8512018f9b28d@mail.gmail.com> <191c3a030909020842j61369a45va923080ea0a88b40@mail.gmail.com> Message-ID: Hello Anthony, 2009/9/2 Anthony Minessale : > yes if you have a version that only has log-file you can use that. > > if you find me on irc and send me the credentials privately I will examine > your box for you. thanks for that offer, but the box is pretty deep inside our internal network with no routing to the outside, several stepping-stones in between and all that "security" stuff. I finally found the right amount of load where the memory leak builds up quickly enough and was able to stop freeswitch before it started swapping. The result is available on http://ns42.ath.cx/B0GdWh/vg-2.log.bz2 (19k) Thx in advance Beni. From brian at freeswitch.org Fri Sep 4 05:45:50 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Sep 2009 07:45:50 -0500 Subject: [Freeswitch-users] Proxy authorization In-Reply-To: <90823c940909040143ie0c175fy5702a9c280676d28@mail.gmail.com> References: <90823c940909031326g5e9d5a7bsd2650a59a7a5eaf@mail.gmail.com> <7B180BCD-82F4-43BD-A8C9-48CC2A38D899@freeswitch.org> <90823c940909040143ie0c175fy5702a9c280676d28@mail.gmail.com> Message-ID: <9147D03C-22CC-4990-8DA6-701D28FC49B1@freeswitch.org> I cannot change the way SIP Authentication works. The first register is always sent without an authorization header then is challenged. If you're getting an instant 403 then you have something wrong in your config and the remote system doesn't like it. Please contact your provider and ask them to troubleshoot it with you. /b On Sep 4, 2009, at 3:43 AM, Dmitry Bely wrote: > > Alas, I cannot change the way the provider's gateway works. It > immediately responses with 403... BTW, it's Mera Damos > (http://www.mera-systems.com ?). No workaround possible? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/1016bf78/attachment.html From rupa at rupa.com Fri Sep 4 07:10:24 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 4 Sep 2009 09:10:24 -0500 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org> <191c3a030909020811q6c04518au514fe8e9d6c8734e@mail.gmail.com> <191c3a030909020815k6006f96fh3df8512018f9b28d@mail.gmail.com> <191c3a030909020842j61369a45va923080ea0a88b40@mail.gmail.com> Message-ID: Worst offenders (leakers over 100K). The last one is the worst (672M) -- looks like a lua script. What are you doing in lua again? ==28624== 105,725 bytes in 1,804 blocks are still reachable in loss record 497 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x50384F2: xmlrpc_strdupnull (asprintf.c:92) ==28624== by 0x503F86D: RequestRead (http.c:57) ==28624== by 0x5044413: ??? (server.c:538) ==28624== by 0x5039FAF: ??? (conn.c:37) ==28624== by 0x50486F1: ??? (thread_pthread.c:48) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 116,772 bytes in 3,156 blocks are definitely lost in loss record 498 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x4088664: switch_channel_perform_hangup (switch_channel.c:1674) ==28624== by 0x64A12EA: ??? (mod_dptools.c:633) ==28624== by 0x409AA45: switch_core_session_exec (switch_core_session.c:1476) ==28624== by 0x409AF88: switch_core_session_execute_application (switch_core_session.c:1398) ==28624== by 0x409E674: switch_core_session_run (switch_core_state_machine.c:166) ==28624== by 0x409A48E: switch_core_session_thread (switch_core_session.c:1066) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== ==28624== ==28624== 119,658 (119,621 direct, 37 indirect) bytes in 3,233 blocks are definitely lost in loss record 499 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x4088664: switch_channel_perform_hangup (switch_channel.c:1674) ==28624== by 0x50B6790: sofia_event_callback (sofia.c:3863) ==28624== by 0x5146787: nua_application_event (nua_stack.c:393) ==28624== by 0x519DB28: su_base_port_execute_msgs (su_base_port.c:280) ==28624== by 0x519D8CF: su_base_port_getmsgs (su_base_port.c:202) ==28624== by 0x519DD75: su_base_port_step (su_base_port.c:454) ==28624== by 0x5190968: su_port_step (su_port.h:340) ==28624== by 0x5190938: su_root_step (su_root.c:858) ==28624== ==28624== ==28624== 124,209 bytes in 3,357 blocks are still reachable in loss record 500 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x409A5EC: switch_core_session_thread (switch_core_session.c:1086) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 124,290 bytes in 4,143 blocks are still reachable in loss record 501 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x443B957: vasprintf (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x5038532: xmlrpc_vasprintf (asprintf.c:61) ==28624== by 0x5038581: xmlrpc_asprintf (asprintf.c:81) ==28624== by 0x503B881: DateToString (date.c:43) ==28624== by 0x5036D09: handler_hook (mod_xml_rpc.c:733) ==28624== by 0x504456F: ??? (server.c:515) ==28624== by 0x5039FAF: ??? (conn.c:37) ==28624== by 0x50486F1: ??? (thread_pthread.c:48) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 137,085 bytes in 3,705 blocks are still reachable in loss record 502 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x409921F: switch_core_session_perform_destroy (switch_core_session.c:947) ==28624== by 0x409A60D: switch_core_session_thread (switch_core_session.c:1088) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 145,589 bytes in 1,837 blocks are possibly lost in loss record 503 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x50384F2: xmlrpc_strdupnull (asprintf.c:92) ==28624== by 0x503F86D: RequestRead (http.c:57) ==28624== by 0x5044413: ??? (server.c:538) ==28624== by 0x5039FAF: ??? (conn.c:37) ==28624== by 0x50486F1: ??? (thread_pthread.c:48) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 151,929 (151,922 direct, 7 indirect) bytes in 4,106 blocks are definitely lost in loss record 504 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x4088664: switch_channel_perform_hangup (switch_channel.c:1674) ==28624== by 0x40C124A: audio_bridge_on_exchange_media (switch_ivr_bridge.c:503) ==28624== by 0x409DB8B: switch_core_session_run (switch_core_state_machine.c:494) ==28624== by 0x409A48E: switch_core_session_thread (switch_core_session.c:1066) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 200,704 bytes in 1 blocks are still reachable in loss record 505 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x410C0EC: apr_palloc (apr_pools.c:300) ==28624== by 0x4101A3A: apr_queue_create (apr_queue.c:129) ==28624== by 0x407FFEA: switch_queue_create (switch_apr.c:897) ==28624== by 0x509F2B5: mod_sofia_load (mod_sofia.c:3371) ==28624== by 0x40AF30D: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==28624== by 0x40AFCCF: switch_loadable_module_init (switch_loadable_module.c:1174) ==28624== by 0x40A8320: switch_core_init_and_modload (switch_core.c:1451) ==28624== by 0x804A7EC: main (switch.c:731) ==28624== ==28624== ==28624== 200,704 bytes in 1 blocks are still reachable in loss record 506 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x410C0EC: apr_palloc (apr_pools.c:300) ==28624== by 0x4101A3A: apr_queue_create (apr_queue.c:129) ==28624== by 0x407FFEA: switch_queue_create (switch_apr.c:897) ==28624== by 0x509F295: mod_sofia_load (mod_sofia.c:3370) ==28624== by 0x40AF30D: switch_loadable_module_load_module_ex (switch_loadable_module.c:846) ==28624== by 0x40AFCCF: switch_loadable_module_init (switch_loadable_module.c:1174) ==28624== by 0x40A8320: switch_core_init_and_modload (switch_core.c:1451) ==28624== by 0x804A7EC: main (switch.c:731) ==28624== ==28624== ==28624== 225,280 bytes in 11 blocks are still reachable in loss record 507 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x503AC39: ??? (mallocvar.h:43) ==28624== by 0x503ACBD: PoolAlloc (data.c:602) ==28624== by 0x503AD2C: PoolStrdup (data.c:674) ==28624== by 0x5043442: MIMETypeAdd2 (response.c:356) ==28624== by 0x50434DD: MIMETypeAdd (response.c:415) ==28624== by 0x503586D: mod_xml_rpc_runtime (mod_xml_rpc.c:936) ==28624== by 0x40AF772: switch_loadable_module_exec (switch_loadable_module.c:94) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 246,312 bytes in 311 blocks are possibly lost in loss record 508 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x50D7759: sofia_glue_do_invite (sofia_glue.c:1677) ==28624== by 0x50A2400: sofia_on_init (mod_sofia.c:102) ==28624== by 0x409D587: switch_core_session_run (switch_core_state_machine.c:481) ==28624== by 0x409A48E: switch_core_session_thread (switch_core_session.c:1066) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 303,104 bytes in 37 blocks are still reachable in loss record 509 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x410BB2F: apr_pool_create_ex (apr_pools.c:300) ==28624== by 0x4092416: switch_core_perform_new_memory_pool (switch_core_memory.c:357) ==28624== by 0x40AF24E: switch_loadable_module_load_module_ex (switch_loadable_module.c:785) ==28624== by 0x40AFCCF: switch_loadable_module_init (switch_loadable_module.c:1174) ==28624== by 0x40A8320: switch_core_init_and_modload (switch_core.c:1451) ==28624== by 0x804A7EC: main (switch.c:731) ==28624== ==28624== ==28624== 399,637 (399,600 direct, 37 indirect) bytes in 10,800 blocks are definitely lost in loss record 510 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x4088664: switch_channel_perform_hangup (switch_channel.c:1674) ==28624== by 0x50B2A5E: sofia_handle_sip_i_bye (sofia.c:327) ==28624== by 0x50B4A76: sofia_event_callback (sofia.c:508) ==28624== by 0x5146787: nua_application_event (nua_stack.c:393) ==28624== by 0x519DB28: su_base_port_execute_msgs (su_base_port.c:280) ==28624== by 0x519D8CF: su_base_port_getmsgs (su_base_port.c:202) ==28624== by 0x519DD75: su_base_port_step (su_base_port.c:454) ==28624== by 0x5190968: su_port_step (su_port.h:340) ==28624== ==28624== ==28624== 444,777 bytes in 12,021 blocks are definitely lost in loss record 511 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x4088664: switch_channel_perform_hangup (switch_channel.c:1674) ==28624== by 0x50B6790: sofia_event_callback (sofia.c:3863) ==28624== by 0x5146787: nua_application_event (nua_stack.c:393) ==28624== by 0x519DB28: su_base_port_execute_msgs (su_base_port.c:280) ==28624== by 0x519D8CF: su_base_port_getmsgs (su_base_port.c:202) ==28624== by 0x519DE27: su_base_port_step (su_base_port.c:473) ==28624== by 0x5190968: su_port_step (su_port.h:340) ==28624== by 0x5190938: su_root_step (su_root.c:858) ==28624== ==28624== ==28624== 2,261,847 (2,261,810 direct, 37 indirect) bytes in 61,130 blocks are definitely lost in loss record 512 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x408A4B7: switch_channel_perform_mark_answered (switch_channel.c:1914) ==28624== by 0x50B8528: sofia_event_callback (sofia.c:3807) ==28624== by 0x5146787: nua_application_event (nua_stack.c:393) ==28624== by 0x519DB28: su_base_port_execute_msgs (su_base_port.c:280) ==28624== by 0x519D8CF: su_base_port_getmsgs (su_base_port.c:202) ==28624== by 0x519DD75: su_base_port_step (su_base_port.c:454) ==28624== by 0x5190968: su_port_step (su_port.h:340) ==28624== by 0x5190938: su_root_step (su_root.c:858) ==28624== ==28624== ==28624== 2,413,880 (2,413,843 direct, 37 indirect) bytes in 65,239 blocks are definitely lost in loss record 513 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x4088664: switch_channel_perform_hangup (switch_channel.c:1674) ==28624== by 0x50B2A5E: sofia_handle_sip_i_bye (sofia.c:327) ==28624== by 0x50B4A76: sofia_event_callback (sofia.c:508) ==28624== by 0x5146787: nua_application_event (nua_stack.c:393) ==28624== by 0x519DB28: su_base_port_execute_msgs (su_base_port.c:280) ==28624== by 0x519D8CF: su_base_port_getmsgs (su_base_port.c:202) ==28624== by 0x519DE27: su_base_port_step (su_base_port.c:473) ==28624== by 0x5190968: su_port_step (su_port.h:340) ==28624== ==28624== ==28624== 2,416,640 bytes in 295 blocks are still reachable in loss record 514 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x410C0EC: apr_palloc (apr_pools.c:300) ==28624== by 0x41110E6: apr_thread_create (thread.c:150) ==28624== by 0x4080878: switch_thread_create (switch_apr.c:631) ==28624== by 0x6C278E9: lua_thread (mod_lua.cpp:372) ==28624== by 0x6C27948: ??? (mod_lua.cpp:407) ==28624== by 0x40AADFC: switch_api_execute (switch_loadable_module.c:1567) ==28624== by 0x583A7FC: ??? (mod_commands.c:2426) ==28624== by 0x40A8881: switch_scheduler_execute (switch_scheduler.c:61) ==28624== by 0x40A8DE0: task_thread_loop (switch_scheduler.c:127) ==28624== by 0x40A8EA3: switch_scheduler_task_thread (switch_scheduler.c:168) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== ==28624== ==28624== 2,671,215 (2,671,178 direct, 37 indirect) bytes in 72,194 blocks are definitely lost in loss record 515 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x408ACF4: switch_channel_perform_mark_ring_ready (switch_channel.c:1697) ==28624== by 0x50B6E29: sofia_event_callback (sofia.c:3366) ==28624== by 0x5146787: nua_application_event (nua_stack.c:393) ==28624== by 0x519DB28: su_base_port_execute_msgs (su_base_port.c:280) ==28624== by 0x519D8CF: su_base_port_getmsgs (su_base_port.c:202) ==28624== by 0x519DD75: su_base_port_step (su_base_port.c:454) ==28624== by 0x5190968: su_port_step (su_port.h:340) ==28624== by 0x5190938: su_root_step (su_root.c:858) ==28624== ==28624== ==28624== 2,826,171 bytes in 76,383 blocks are definitely lost in loss record 516 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x4088664: switch_channel_perform_hangup (switch_channel.c:1674) ==28624== by 0x40C1049: signal_bridge_on_hangup (switch_ivr_bridge.c:710) ==28624== by 0x409D7CF: switch_core_session_run (switch_core_state_machine.c:434) ==28624== by 0x409A48E: switch_core_session_thread (switch_core_session.c:1066) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 2,968,658 bytes in 80,234 blocks are definitely lost in loss record 517 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x692F07B: ??? (mod_dialplan_xml.c:315) ==28624== by 0x409EFFD: switch_core_session_run (switch_core_state_machine.c:109) ==28624== by 0x409A48E: switch_core_session_thread (switch_core_session.c:1066) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 2,974,282 (2,974,245 direct, 37 indirect) bytes in 80,385 blocks are definitely lost in loss record 518 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x40DDF32: switch_ivr_session_transfer (switch_ivr.c:1350) ==28624== by 0x5840B54: ??? (mod_commands.c:2319) ==28624== by 0x40AADFC: switch_api_execute (switch_loadable_module.c:1567) ==28624== by 0x5036FAA: handler_hook (mod_xml_rpc.c:777) ==28624== by 0x504456F: ??? (server.c:515) ==28624== by 0x5039FAF: ??? (conn.c:37) ==28624== by 0x50486F1: ??? (thread_pthread.c:48) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== ==28624== ==28624== 2,976,983 (2,976,909 direct, 74 indirect) bytes in 80,457 blocks are definitely lost in loss record 519 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x408B204: switch_channel_set_name (switch_channel.c:602) ==28624== by 0x50D3FDD: sofia_glue_attach_private (sofia_glue.c:527) ==28624== by 0x50A35EF: sofia_outgoing_channel (mod_sofia.c:2854) ==28624== by 0x409B970: switch_core_session_outgoing_channel (switch_core_session.c:410) ==28624== by 0x40C53D8: switch_ivr_originate (switch_ivr_originate.c:1508) ==28624== by 0x64A7F2F: ??? (mod_dptools.c:2092) ==28624== by 0x409AA45: switch_core_session_exec (switch_core_session.c:1476) ==28624== by 0x409AF88: switch_core_session_execute_application (switch_core_session.c:1398) ==28624== ==28624== ==28624== 3,085,393 bytes in 83,389 blocks are definitely lost in loss record 520 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x408ACF4: switch_channel_perform_mark_ring_ready (switch_channel.c:1697) ==28624== by 0x50B6E29: sofia_event_callback (sofia.c:3366) ==28624== by 0x5146787: nua_application_event (nua_stack.c:393) ==28624== by 0x519DB28: su_base_port_execute_msgs (su_base_port.c:280) ==28624== by 0x519D8CF: su_base_port_getmsgs (su_base_port.c:202) ==28624== by 0x519DE27: su_base_port_step (su_base_port.c:473) ==28624== by 0x5190968: su_port_step (su_port.h:340) ==28624== by 0x5190938: su_root_step (su_root.c:858) ==28624== ==28624== ==28624== 3,110,961 (3,107,016 direct, 3,945 indirect) bytes in 3,923 blocks are definitely lost in loss record 521 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x50D7759: sofia_glue_do_invite (sofia_glue.c:1677) ==28624== by 0x50A2400: sofia_on_init (mod_sofia.c:102) ==28624== by 0x409D587: switch_core_session_run (switch_core_state_machine.c:481) ==28624== by 0x409A48E: switch_core_session_thread (switch_core_session.c:1066) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 3,550,402 (3,550,261 direct, 141 indirect) bytes in 95,953 blocks are definitely lost in loss record 522 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x408B204: switch_channel_set_name (switch_channel.c:602) ==28624== by 0x50D3FDD: sofia_glue_attach_private (sofia_glue.c:527) ==28624== by 0x50A35EF: sofia_outgoing_channel (mod_sofia.c:2854) ==28624== by 0x409B970: switch_core_session_outgoing_channel (switch_core_session.c:410) ==28624== by 0x40C53D8: switch_ivr_originate (switch_ivr_originate.c:1508) ==28624== by 0x5840AF1: ??? (mod_commands.c:2285) ==28624== by 0x40AADFC: switch_api_execute (switch_loadable_module.c:1567) ==28624== by 0x5036FAA: handler_hook (mod_xml_rpc.c:777) ==28624== ==28624== ==28624== 3,687,864 (3,687,790 direct, 74 indirect) bytes in 99,670 blocks are definitely lost in loss record 523 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x408A4B7: switch_channel_perform_mark_answered (switch_channel.c:1914) ==28624== by 0x50B8528: sofia_event_callback (sofia.c:3807) ==28624== by 0x5146787: nua_application_event (nua_stack.c:393) ==28624== by 0x519DB28: su_base_port_execute_msgs (su_base_port.c:280) ==28624== by 0x519D8CF: su_base_port_getmsgs (su_base_port.c:202) ==28624== by 0x519DE27: su_base_port_step (su_base_port.c:473) ==28624== by 0x5190968: su_port_step (su_port.h:340) ==28624== by 0x5190938: su_root_step (su_root.c:858) ==28624== ==28624== ==28624== 4,833,280 bytes in 590 blocks are still reachable in loss record 524 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x410C0EC: apr_palloc (apr_pools.c:300) ==28624== by 0x41112CF: apr_threadattr_create (thread.c:45) ==28624== by 0x4080953: switch_threadattr_create (switch_apr.c:589) ==28624== by 0x6C27891: lua_thread (mod_lua.cpp:369) ==28624== by 0x6C27948: ??? (mod_lua.cpp:407) ==28624== by 0x40AADFC: switch_api_execute (switch_loadable_module.c:1567) ==28624== by 0x583A7FC: ??? (mod_commands.c:2426) ==28624== by 0x40A8881: switch_scheduler_execute (switch_scheduler.c:61) ==28624== by 0x40A8DE0: task_thread_loop (switch_scheduler.c:127) ==28624== by 0x40A8EA3: switch_scheduler_task_thread (switch_scheduler.c:168) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== ==28624== ==28624== 6,502,484 (6,502,269 direct, 215 indirect) bytes in 175,737 blocks are definitely lost in loss record 525 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x409921F: switch_core_session_perform_destroy (switch_core_session.c:947) ==28624== by 0x409A60D: switch_core_session_thread (switch_core_session.c:1088) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 6,515,323 (6,515,219 direct, 104 indirect) bytes in 176,087 blocks are definitely lost in loss record 526 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x40E4EA9: switch_log_vprintf (switch_log.c:438) ==28624== by 0x40E5130: switch_log_printf (switch_log.c:308) ==28624== by 0x409A5EC: switch_core_session_thread (switch_core_session.c:1086) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 7,557,720 bytes in 251,924 blocks are definitely lost in loss record 527 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x443B957: vasprintf (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x5038532: xmlrpc_vasprintf (asprintf.c:61) ==28624== by 0x5038581: xmlrpc_asprintf (asprintf.c:81) ==28624== by 0x503B881: DateToString (date.c:43) ==28624== by 0x5036D09: handler_hook (mod_xml_rpc.c:733) ==28624== by 0x504456F: ??? (server.c:515) ==28624== by 0x5039FAF: ??? (conn.c:37) ==28624== by 0x50486F1: ??? (thread_pthread.c:48) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 8,700,546 (8,700,436 direct, 110 indirect) bytes in 253,570 blocks are definitely lost in loss record 528 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x444AFCF: strdup (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== by 0x50384F2: xmlrpc_strdupnull (asprintf.c:92) ==28624== by 0x503F86D: RequestRead (http.c:57) ==28624== by 0x5044413: ??? (server.c:538) ==28624== by 0x5039FAF: ??? (conn.c:37) ==28624== by 0x50486F1: ??? (thread_pthread.c:48) ==28624== by 0x42114FA: start_thread (in /lib/tls/i686/cmov/libpthread-2.7.so) ==28624== by 0x44AFE5D: clone (in /lib/tls/i686/cmov/libc-2.7.so) ==28624== ==28624== ==28624== 672,268,288 bytes in 82,064 blocks are still reachable in loss record 529 of 529 ==28624== at 0x4022AB8: malloc (vg_replace_malloc.c:207) ==28624== by 0x410BB2F: apr_pool_create_ex (apr_pools.c:300) ==28624== by 0x4111176: apr_thread_create (thread.c:171) ==28624== by 0x4080878: switch_thread_create (switch_apr.c:631) ==28624== by 0x6C278E9: lua_thread (mod_lua.cpp:372) ==28624== by 0x6C27948: ??? (mod_lua.cpp:407) ==28624== by 0x40AADFC: switch_api_execute (switch_loadable_module.c:1567) ==28624== by 0x583A7FC: ??? (mod_commands.c:2426) ==28624== by 0x40A8881: switch_scheduler_execute (switch_scheduler.c:61) ==28624== by 0x40A8DE0: task_thread_loop (switch_scheduler.c:127) ==28624== by 0x40A8EA3: switch_scheduler_task_thread (switch_scheduler.c:168) ==28624== by 0x4110E05: dummy_worker (thread.c:138) ==28624== ==28624== LEAK SUMMARY: ==28624== definitely lost: 63,113,740 bytes in 1,690,880 blocks. ==28624== indirectly lost: 35,632 bytes in 491 blocks. ==28624== possibly lost: 645,758 bytes in 9,150 blocks. ==28624== still reachable: 681,849,684 bytes in 113,077 blocks. ==28624== suppressed: 0 bytes in 0 blocks. On Fri, Sep 4, 2009 at 7:42 AM, Benedikt Fraunhofer wrote: > Hello Anthony, > > 2009/9/2 Anthony Minessale : > >> yes if you have a version that only has log-file you can use that. >> >> if you find me on irc and send me the credentials privately I will examine >> your box for you. > > thanks for that offer, but the box is pretty deep inside our internal > network with no routing to the outside, several stepping-stones in > between and all that "security" stuff. > > I finally found the right amount of load where the memory leak builds > up quickly enough and was able to stop freeswitch before it started > swapping. The result is available on > > ?http://ns42.ath.cx/B0GdWh/vg-2.log.bz2 > > (19k) > > Thx in advance > ?Beni. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From rupa at rupa.com Fri Sep 4 07:29:07 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 4 Sep 2009 09:29:07 -0500 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org> <191c3a030909020811q6c04518au514fe8e9d6c8734e@mail.gmail.com> <191c3a030909020815k6006f96fh3df8512018f9b28d@mail.gmail.com> <191c3a030909020842j61369a45va923080ea0a88b40@mail.gmail.com> Message-ID: Doesn't that look like a pool that isn't being destroyed? On Fri, Sep 4, 2009 at 9:10 AM, Rupa Schomaker wrote: > Worst offenders (leakers over 100K). ?The last one is the worst (672M) > -- looks like a lua script. ?What are you doing in lua again? > ==28624== 672,268,288 bytes in 82,064 blocks are still reachable in > loss record 529 of 529 > ==28624== ? ?at 0x4022AB8: malloc (vg_replace_malloc.c:207) > ==28624== ? ?by 0x410BB2F: apr_pool_create_ex (apr_pools.c:300) > ==28624== ? ?by 0x4111176: apr_thread_create (thread.c:171) > ==28624== ? ?by 0x4080878: switch_thread_create (switch_apr.c:631) > ==28624== ? ?by 0x6C278E9: lua_thread (mod_lua.cpp:372) > ==28624== ? ?by 0x6C27948: ??? (mod_lua.cpp:407) > ==28624== ? ?by 0x40AADFC: switch_api_execute (switch_loadable_module.c:1567) > ==28624== ? ?by 0x583A7FC: ??? (mod_commands.c:2426) > ==28624== ? ?by 0x40A8881: switch_scheduler_execute (switch_scheduler.c:61) > ==28624== ? ?by 0x40A8DE0: task_thread_loop (switch_scheduler.c:127) > ==28624== ? ?by 0x40A8EA3: switch_scheduler_task_thread (switch_scheduler.c:168) > ==28624== ? ?by 0x4110E05: dummy_worker (thread.c:138) > ==28624== > ==28624== LEAK SUMMARY: > ==28624== ? ?definitely lost: 63,113,740 bytes in 1,690,880 blocks. > ==28624== ? ?indirectly lost: 35,632 bytes in 491 blocks. > ==28624== ? ? ?possibly lost: 645,758 bytes in 9,150 blocks. > ==28624== ? ?still reachable: 681,849,684 bytes in 113,077 blocks. > ==28624== ? ? ? ? suppressed: 0 bytes in 0 blocks. -- -Rupa From math.parent at gmail.com Fri Sep 4 07:32:13 2009 From: math.parent at gmail.com (Mathieu Parent) Date: Fri, 4 Sep 2009 16:32:13 +0200 Subject: [Freeswitch-users] Run a command on event In-Reply-To: <88D1D05C-B36D-4E7B-9DFC-58688DFD3467@avgs.ca> References: <960738410909020727q3286bce9q2ff7d4edfcf6e025@mail.gmail.com> <86a32abc0909020746n1250604ds3ebaf3a3d7abdd21@mail.gmail.com> <960738410909020847u2562a1bei6cedba26e5b06cb6@mail.gmail.com> <88D1D05C-B36D-4E7B-9DFC-58688DFD3467@avgs.ca> Message-ID: <960738410909040732o3f08ec28x6d1869aa6f4649b9@mail.gmail.com> Hi On Thu, Sep 3, 2009 at 6:52 AM, Mathieu Rene wrote: > See http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Subscribing_to_events > Thanks. I have tried this method without success and finally replaced the voicemail section in dialplan by a spidermonkey script with session.setHangupHook(). Test passed! Mathieu Parent From fraunhofer.lists.freeswitch-001 at traced.net Fri Sep 4 07:32:53 2009 From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer) Date: Fri, 4 Sep 2009 16:32:53 +0200 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org> <191c3a030909020811q6c04518au514fe8e9d6c8734e@mail.gmail.com> <191c3a030909020815k6006f96fh3df8512018f9b28d@mail.gmail.com> <191c3a030909020842j61369a45va923080ea0a88b40@mail.gmail.com> Message-ID: 2009/9/4 Rupa Schomaker : > Worst offenders (leakers over 100K). ?The last one is the worst (672M) > -- looks like a lua script. ?What are you doing in lua again? i feel kinda dumb to double post, but here it is again :) the setup is the same as in http://jira.freeswitch.org/browse/MODSOFIA-22 one is ----- local reason = session:getVariable("originate_disposition"); session:setAutoHangup(false); if(reason) then if(reason == "NO_ANSWER") then -- nothing end if(reason == "USER_BUSY") then -- nothing end end freeswitch.consoleLog(... ------ anotherone is ---- local sess = "nil"; if(argv[1]) then sess=argv[1]; end freeswitch.consoleLog(... api = freeswitch.API(); local res = api:execute("sched_api" ... freeswitch.consoleLog(... ---- and the scheduled script does --- function log(msg) freeswitch.consoleLog("notice", "c2c-hangup-timeout.lua: " .. msg .. "\n"); end local sess = argv[1]; if(sess) then freeswitch.consoleLog("INFO", "hangup-timeout.lua for uuid " .. sess .. "\n"); api = freeswitch.API(); local stillValid = api:execute("uuid_getvar", sess .. " Dummy-DoesChannelExists"); if(stillValid:sub(1,4) == "-ERR") then log("session uuid " .. sess .. " disappeared (nothing bad)"); else -- this is important!!! Otherwise the aleg get's just hung up! api:execute("uuid_media", sess); api:execute("uuid_transfer", sess .. " -both timeout"); end else -- /if(sess) log("called with nil session?"); end -- /if(sess) --- at least there's no fancy db-connection-thingi which could make debugging harder :) Cheers Beni. From fraunhofer.lists.freeswitch-001 at traced.net Fri Sep 4 07:35:08 2009 From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer) Date: Fri, 4 Sep 2009 16:35:08 +0200 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <191c3a030909020811q6c04518au514fe8e9d6c8734e@mail.gmail.com> <191c3a030909020815k6006f96fh3df8512018f9b28d@mail.gmail.com> <191c3a030909020842j61369a45va923080ea0a88b40@mail.gmail.com> Message-ID: personally i would blame xmlrpc (which is no xml :) for it. Just my 2cent Beni. From gshfreesw at gmail.com Fri Sep 4 07:37:34 2009 From: gshfreesw at gmail.com (Shameem Shiek) Date: Fri, 4 Sep 2009 10:37:34 -0400 Subject: [Freeswitch-users] XML Dial Plan vs Language Modules In-Reply-To: <87f2f3b90909040006s15c1f733lff93b4549dee6da7@mail.gmail.com> References: <87f2f3b90909040006s15c1f733lff93b4549dee6da7@mail.gmail.com> Message-ID: <5070fcbd0909040737v6d39a234j62d6e1610a001b47@mail.gmail.com> Hi Michael, Why is it not recommended to do the brdge app right in the script? The reason I ask this, I did have lot of trouble using Park/Fifo app in the script and the whole thing started working after I did the UUID transfer and have the things I wanted executed as part of the Dial plan. Also, How many concurrent sessions can one support in ESL using Python/Ruby compared to using Lua? Thanks. On Fri, Sep 4, 2009 at 3:06 AM, Michael Collins wrote: > > > On Thu, Sep 3, 2009 at 10:59 PM, Muhammad Shahzad < > shaheryarkh at googlemail.com> wrote: > >> Hi, >> >> I couple of my team members are working on translating a very long >> Asterisk Dial Plan to FreeSWITCH XML Dial Plan. Now reference to wiki link >> below, >> > > Before you go through all the trouble of translating the dialplan be sure > to review the application itself. In many cases just doing a dialplan > translation results in less efficient use of FreeSWITCH's powerful features. > Be sure that you are looking at the way FreeSWITCH handles various > situations and take advantage of its power and ease of use. > >> >> >> http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Variables >> >> The dial plan variables are not getting initialized as expected. I was >> just wondering if we move this variable get and set stuff to any language >> module say mod_perl, will that make any difference performance wise? I mean >> we will be invoking a Perl interpreter for each incoming call, won't that be >> expensive in terms of RAM and CPU usage and thus reducing number of calls >> this FS deployment can handle? >> >> I have guys with programming skills in Perl, PHP, Python, Java and LUA >> languages. Which language do you recommend for this, again in terms of speed >> and performance? >> >> > Lua is very portable and we've done tests with hundreds of concurrent Lua > scripts running. The other languages are heavier but they'll still handle > quite a few concurrent sessions. Just be sure that you don't do the bridge > app right in the script, use transfer instead and have the dialplan process > any bridging that you need to do. > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/8aa5052a/attachment.html From rupa at rupa.com Fri Sep 4 07:42:51 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 4 Sep 2009 09:42:51 -0500 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <191c3a030909020811q6c04518au514fe8e9d6c8734e@mail.gmail.com> <191c3a030909020815k6006f96fh3df8512018f9b28d@mail.gmail.com> <191c3a030909020842j61369a45va923080ea0a88b40@mail.gmail.com> Message-ID: There are other smaller leakers. xmlrpc is leaking, but the leaks are very small compared to the lua leak. Same with spidermonkey_curl - it is leaking but not too terribly much. I'll hop on #freeswitch in a bit and see if anyone has an idea. On Fri, Sep 4, 2009 at 9:35 AM, Benedikt Fraunhofer wrote: > personally i would blame xmlrpc (which is no xml :) for it. > > Just my 2cent > > ?Beni. > http://www.freeswitch.org > -- -Rupa From anthony.minessale at gmail.com Fri Sep 4 07:57:31 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Sep 2009 09:57:31 -0500 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <191c3a030909020811q6c04518au514fe8e9d6c8734e@mail.gmail.com> <191c3a030909020815k6006f96fh3df8512018f9b28d@mail.gmail.com> <191c3a030909020842j61369a45va923080ea0a88b40@mail.gmail.com> Message-ID: <191c3a030909040757y93c105bs8939559f90142b9e@mail.gmail.com> that looks to me like luarun being called on a script that never terminates. could your script be ending up caught in an endless loop or blocking on something? On Fri, Sep 4, 2009 at 9:42 AM, Rupa Schomaker wrote: > There are other smaller leakers. xmlrpc is leaking, but the leaks are > very small compared to the lua leak. Same with spidermonkey_curl - it > is leaking but not too terribly much. I'll hop on #freeswitch in a > bit and see if anyone has an idea. > > On Fri, Sep 4, 2009 at 9:35 AM, Benedikt > Fraunhofer wrote: > > personally i would blame xmlrpc (which is no xml :) for it. > > > > Just my 2cent > > > > Beni. > > > http://www.freeswitch.org > > > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/4a04b013/attachment-0001.html From brian.stafford at lattice-voice.com Fri Sep 4 04:53:58 2009 From: brian.stafford at lattice-voice.com (Brian Stafford) Date: Fri, 04 Sep 2009 12:53:58 +0100 Subject: [Freeswitch-users] UK English wav files Message-ID: <4AA0FFD6.9070804@lattice-voice.com> Hi all anyone know where I can find UK English recordings for the FS prompts (assuming there are any)? (I've googled to no avail). Alternatively is there a list of the text used so we can record our own? Regards Brian From mike at jerris.com Fri Sep 4 09:17:54 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 4 Sep 2009 12:17:54 -0400 Subject: [Freeswitch-users] UK English wav files In-Reply-To: <4AA0FFD6.9070804@lattice-voice.com> References: <4AA0FFD6.9070804@lattice-voice.com> Message-ID: <1420B76C-36EE-4D13-B456-38E8B8A542DB@jerris.com> We don't currently have a full set of UK English prompts, the prompts list (soon to be updated with some new prompts) is available at: http://svn.freeswitch.org/svn/freeswitch/trunk/docs/phrase/phrase_en.xml If you are going to get a set professionally recorded, we would be happy to host those files and integrate into the build system like we did the russian sounds. Mike On Sep 4, 2009, at 7:53 AM, Brian Stafford wrote: > Hi all > > anyone know where I can find UK English recordings for the FS prompts > (assuming there are any)? (I've googled to no avail). > Alternatively is > there a list of the text used so we can record our own? > > Regards > Brian From msc at freeswitch.org Fri Sep 4 09:18:36 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 Sep 2009 09:18:36 -0700 Subject: [Freeswitch-users] REMINDER: Weekly call is now happening. Join us! Message-ID: <87f2f3b90909040918l7f62779eu3655edb9f8b7faa8@mail.gmail.com> Hello all, We are now on line and welcoming callers. Here's the agenda so far: http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_04 Come join the conference sip:888 at conference.freeswitch.org 1-213-799-1400 -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/31e3a322/attachment.html From diego.viola at gmail.com Fri Sep 4 09:43:51 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 4 Sep 2009 16:43:51 +0000 Subject: [Freeswitch-users] REMINDER: Weekly call is now happening. Join us! In-Reply-To: <87f2f3b90909040918l7f62779eu3655edb9f8b7faa8@mail.gmail.com> References: <87f2f3b90909040918l7f62779eu3655edb9f8b7faa8@mail.gmail.com> Message-ID: <86a32abc0909040943p42186dd6q57960727045e6567@mail.gmail.com> I'm in, very cool =D Diego On Fri, Sep 4, 2009 at 4:18 PM, Michael Collins wrote: > Hello all, > > We are now on line and welcoming callers. Here's the agenda so far: > http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_04 > > Come join the conference > sip:888 at conference.freeswitch.org > 1-213-799-1400 > > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/7f230cfe/attachment.html From gmaruzz at celliax.org Fri Sep 4 09:58:51 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 4 Sep 2009 18:58:51 +0200 Subject: [Freeswitch-users] REMINDER: Weekly call is now happening. Join us! In-Reply-To: <86a32abc0909040943p42186dd6q57960727045e6567@mail.gmail.com> References: <87f2f3b90909040918l7f62779eu3655edb9f8b7faa8@mail.gmail.com> <86a32abc0909040943p42186dd6q57960727045e6567@mail.gmail.com> Message-ID: <7b197bef0909040958o25272435sb5b48d21558ee137@mail.gmail.com> For the ones SIP challenged: call Skype the skypeuser "skypiax5" and then press 1 -gm On Fri, Sep 4, 2009 at 6:43 PM, Diego Viola wrote: > I'm in, very cool =D > > Diego > > On Fri, Sep 4, 2009 at 4:18 PM, Michael Collins wrote: >> >> Hello all, >> >> We are now on line and welcoming callers. Here's the agenda so far: >> http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_04 >> >> Come join the conference >> sip:888 at conference.freeswitch.org >> 1-213-799-1400 >> >> -MC >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From lon at kickasspixels.com Fri Sep 4 11:17:02 2009 From: lon at kickasspixels.com (Lon Baker) Date: Fri, 4 Sep 2009 11:17:02 -0700 Subject: [Freeswitch-users] SIP provider directory? Message-ID: <5d3e0dc60909041117p36d4962ek10b51643503202b9@mail.gmail.com> Does anyone know of a SIP provider or network directory? A list of all the public service provider or networks? Gizmo, Google Voice, etc? Or Vitelity, iCall? Lon Baker Kickass Pixels - +1-415-894-0184 - http://kickasspixels.com http://twitter.com/kickasspixels http://www.linkedin.com/in/lonbaker -- From jerry.richards at teotech.com Fri Sep 4 11:33:51 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Fri, 4 Sep 2009 11:33:51 -0700 Subject: [Freeswitch-users] Minimum/Recommended Freeswitch System Configuration Message-ID: <93C3A49CAB7743DFB845CFA0A8714A4E@greyhawk.tonecommander.com> Under the Minimum/Recommended System Requirements, what is meant by "We recommend you plan for 50% duty cycle"? What is this duty cycle? Also, I see that the system requirements indicate Freeswitch recommends 1GB RAM and 50MB disk space. I guess I'm wondering how the number of extensions and external interfaces drive size of RAM and disk space? For example, would these recommendations support 100 extensions and one external interface? 500 extensions and 10 external interfaces? Etc.? Best Regards, Jerry From dmitry.bely at gmail.com Fri Sep 4 12:22:47 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Fri, 4 Sep 2009 23:22:47 +0400 Subject: [Freeswitch-users] Proxy authorization In-Reply-To: <9147D03C-22CC-4990-8DA6-701D28FC49B1@freeswitch.org> References: <90823c940909031326g5e9d5a7bsd2650a59a7a5eaf@mail.gmail.com> <7B180BCD-82F4-43BD-A8C9-48CC2A38D899@freeswitch.org> <90823c940909040143ie0c175fy5702a9c280676d28@mail.gmail.com> <9147D03C-22CC-4990-8DA6-701D28FC49B1@freeswitch.org> Message-ID: <90823c940909041222q3ad30f88iadf9d067502b2076@mail.gmail.com> On Fri, Sep 4, 2009 at 4:45 PM, Brian West wrote: > I cannot change the way SIP Authentication works. ?The first register is > always sent without an authorization header then is challenged. ?If you're > getting an instant 403 then you have something wrong in your config and the > remote system doesn't like it. ?Please contact your provider and ask them to > troubleshoot it with you. > /b Well, you are right. Looks like the problem is not with authorization but in the line Contact: that the gateway would like to see as Contact: I've found (almost undocumented) parameter extension-in-contact, but it still gives Contact: (1.2.3.4 is my IP address, 5.6.7.8 is gateway's one). Any idea how to overcome this? > On Sep 4, 2009, at 3:43 AM, Dmitry Bely wrote: > > Alas, I cannot change the way the provider's gateway works. It > immediately responses with 403... BTW, it's Mera Damos > (http://www.mera-systems.com ?). No workaround possible? - Dmitry Bely From brian at freeswitch.org Fri Sep 4 12:37:54 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Sep 2009 14:37:54 -0500 Subject: [Freeswitch-users] Proxy authorization In-Reply-To: <90823c940909041222q3ad30f88iadf9d067502b2076@mail.gmail.com> References: <90823c940909031326g5e9d5a7bsd2650a59a7a5eaf@mail.gmail.com> <7B180BCD-82F4-43BD-A8C9-48CC2A38D899@freeswitch.org> <90823c940909040143ie0c175fy5702a9c280676d28@mail.gmail.com> <9147D03C-22CC-4990-8DA6-701D28FC49B1@freeswitch.org> <90823c940909041222q3ad30f88iadf9d067502b2076@mail.gmail.com> Message-ID: <533FE634-F83A-402D-87E7-9CC61BAA9D09@freeswitch.org> Try filling out contact-host too. But if the far end gets pissed about your contact they are broken. /b On Sep 4, 2009, at 2:22 PM, Dmitry Bely wrote: > Well, you are right. Looks like the problem is not with authorization > but in the line > > Contact: > > that the gateway would like to see as > > Contact: > > I've found (almost undocumented) parameter extension-in-contact, but > it still gives > > Contact: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/eba59805/attachment.html From dmitry.bely at gmail.com Fri Sep 4 13:43:18 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Sat, 5 Sep 2009 00:43:18 +0400 Subject: [Freeswitch-users] Proxy authorization In-Reply-To: <533FE634-F83A-402D-87E7-9CC61BAA9D09@freeswitch.org> References: <90823c940909031326g5e9d5a7bsd2650a59a7a5eaf@mail.gmail.com> <7B180BCD-82F4-43BD-A8C9-48CC2A38D899@freeswitch.org> <90823c940909040143ie0c175fy5702a9c280676d28@mail.gmail.com> <9147D03C-22CC-4990-8DA6-701D28FC49B1@freeswitch.org> <90823c940909041222q3ad30f88iadf9d067502b2076@mail.gmail.com> <533FE634-F83A-402D-87E7-9CC61BAA9D09@freeswitch.org> Message-ID: <90823c940909041343t724c7804o62fcf2794699e102@mail.gmail.com> I'm started to suspect another thing.. Successful register (SIP phone) contains REGISTER sip:Domain SIP/2.0 while unsuccessful one is REGISTER sip:1.2.3.4 SIP/2.0 What parameter is responsible for Request-URI? Note that I need both IP address for proxy and symbolic name for SIP domain (which is not mapped the resolvable DNS name). On Fri, Sep 4, 2009 at 11:37 PM, Brian West wrote: > Try filling out contact-host too. ?But if the far end gets pissed about your > contact they are broken. > /b > On Sep 4, 2009, at 2:22 PM, Dmitry Bely wrote: > > Well, you are right. Looks like the problem is not with authorization > but in the line > > ?Contact: > > that the gateway would like to see as > > ?Contact: > > I've found (almost undocumented) parameter extension-in-contact, but > it still gives > > ?Contact: - Dmitry Bely From brian at freeswitch.org Fri Sep 4 14:08:36 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Sep 2009 16:08:36 -0500 Subject: [Freeswitch-users] Proxy authorization In-Reply-To: <90823c940909041343t724c7804o62fcf2794699e102@mail.gmail.com> References: <90823c940909031326g5e9d5a7bsd2650a59a7a5eaf@mail.gmail.com> <7B180BCD-82F4-43BD-A8C9-48CC2A38D899@freeswitch.org> <90823c940909040143ie0c175fy5702a9c280676d28@mail.gmail.com> <9147D03C-22CC-4990-8DA6-701D28FC49B1@freeswitch.org> <90823c940909041222q3ad30f88iadf9d067502b2076@mail.gmail.com> <533FE634-F83A-402D-87E7-9CC61BAA9D09@freeswitch.org> <90823c940909041343t724c7804o62fcf2794699e102@mail.gmail.com> Message-ID: show me your XML for the gateway please. /b On Sep 4, 2009, at 3:43 PM, Dmitry Bely wrote: > I'm started to suspect another thing.. Successful register (SIP > phone) contains > > REGISTER sip:Domain SIP/2.0 > > while unsuccessful one is > > REGISTER sip:1.2.3.4 SIP/2.0 > > What parameter is responsible for Request-URI? Note that I need both > IP address for proxy and symbolic name for SIP domain (which is not > mapped the resolvable DNS name). -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/1f6214ff/attachment-0001.html From mitul at enterux.com Fri Sep 4 14:21:07 2009 From: mitul at enterux.com (Mitul Limbani) Date: Sat, 5 Sep 2009 02:51:07 +0530 Subject: [Freeswitch-users] Minimum/Recommended Freeswitch System Configuration In-Reply-To: <93C3A49CAB7743DFB845CFA0A8714A4E@greyhawk.tonecommander.com> References: <93C3A49CAB7743DFB845CFA0A8714A4E@greyhawk.tonecommander.com> Message-ID: <8672FA11-0B1C-4967-852F-6B2D82A1B405@enterux.com> Jerry, As far as I understand freeswitch, it using kernel to thread and this operation eats good amount of RAM, but since the internal strructure of fs is to store all these sip details in runtime sqlite db, which is compressed text data earlier written in XML but while fs loads this configs it gets it in sqlite and that's what it used instead of asterisks astdb. Although what you see as recommended config for 500 users is true but it also depends on which processor you are trying this on. Intel or AMD is still ok but if you trying it on arm I don't have any data as such, interestingly if you have some test hardware scenario you can actually test and let us all know about it, it's quite useful bit of info that can be positioned on the FS Wiki, in case you want to take this experiment offlist do write to me, im interested to document :) Look forward to hear from you, Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 05-Sep-2009, at 12:03 AM, "Jerry Richards" wrote: > > Under the Minimum/Recommended System Requirements, what is meant by > "We > recommend you plan for 50% duty cycle"? What is this duty cycle? > > Also, I see that the system requirements indicate Freeswitch > recommends 1GB > RAM and 50MB disk space. I guess I'm wondering how the number of > extensions > and external interfaces drive size of RAM and disk space? For > example, > would these recommendations support 100 extensions and one external > interface? 500 extensions and 10 external interfaces? Etc.? > > Best Regards, > Jerry > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From dmitry.bely at gmail.com Fri Sep 4 14:33:22 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Sat, 5 Sep 2009 01:33:22 +0400 Subject: [Freeswitch-users] Proxy authorization In-Reply-To: References: <90823c940909031326g5e9d5a7bsd2650a59a7a5eaf@mail.gmail.com> <7B180BCD-82F4-43BD-A8C9-48CC2A38D899@freeswitch.org> <90823c940909040143ie0c175fy5702a9c280676d28@mail.gmail.com> <9147D03C-22CC-4990-8DA6-701D28FC49B1@freeswitch.org> <90823c940909041222q3ad30f88iadf9d067502b2076@mail.gmail.com> <533FE634-F83A-402D-87E7-9CC61BAA9D09@freeswitch.org> <90823c940909041343t724c7804o62fcf2794699e102@mail.gmail.com> Message-ID: <90823c940909041433y17738430ucfe45c9b1a5e10ee@mail.gmail.com> On Sat, Sep 5, 2009 at 1:08 AM, Brian West wrote: > show me your XML for the gateway please. > /b It's fairly standard: default_provider_register is set to true. In the meantime I looked into the sources. If I understand them right, proxy address is always used in REGISTER header: sofia.c, line 1471 gateway->register_url = switch_core_sprintf(gateway->pool, "sip:%s", proxy); Probably it's incorrect as RFC 3261 says: Request-URI: The Request-URI names the domain of the location service for which the registration is meant (for example, "sip:chicago.com"). The "userinfo" and "@" components of the SIP URI MUST NOT be present. So the domain name (from-domain?) should be used there, not the proxy address. > On Sep 4, 2009, at 3:43 PM, Dmitry Bely wrote: > > I'm started to suspect another thing.. Successful register (SIP phone) > contains > > REGISTER?sip:Domain?SIP/2.0 > > while unsuccessful one is > > REGISTER?sip:1.2.3.4?SIP/2.0 > > What parameter is responsible for Request-URI? Note that I need both > IP address for proxy and symbolic name for SIP domain ?(which is not > mapped the resolvable DNS name). - Dmitry Bely From brian at freeswitch.org Fri Sep 4 14:38:24 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Sep 2009 16:38:24 -0500 Subject: [Freeswitch-users] Proxy authorization In-Reply-To: <90823c940909041433y17738430ucfe45c9b1a5e10ee@mail.gmail.com> References: <90823c940909031326g5e9d5a7bsd2650a59a7a5eaf@mail.gmail.com> <7B180BCD-82F4-43BD-A8C9-48CC2A38D899@freeswitch.org> <90823c940909040143ie0c175fy5702a9c280676d28@mail.gmail.com> <9147D03C-22CC-4990-8DA6-701D28FC49B1@freeswitch.org> <90823c940909041222q3ad30f88iadf9d067502b2076@mail.gmail.com> <533FE634-F83A-402D-87E7-9CC61BAA9D09@freeswitch.org> <90823c940909041343t724c7804o62fcf2794699e102@mail.gmail.com> <90823c940909041433y17738430ucfe45c9b1a5e10ee@mail.gmail.com> Message-ID: Can you send it to me with the data filled out off list please. /b On Sep 4, 2009, at 4:33 PM, Dmitry Bely wrote: > It's fairly standard: > > From mitul at enterux.com Fri Sep 4 14:39:56 2009 From: mitul at enterux.com (Mitul Limbani) Date: Sat, 5 Sep 2009 03:09:56 +0530 Subject: [Freeswitch-users] Does FreeSWITCH wiki have update notify? Message-ID: <5DBA5DF4-7C42-41C5-AB79-1BC75FAD580E@enterux.com> Hello, It may sound a bit stupid but still wanna ask out here, if there is any way to replicate FreeSWITCH wiki mirror for local reference or mainataining local copy instead of Reading online which costs a lot in developing countries like that of ours and Asia/Africa in general. We tried httrack but that brings in everything back here as static HTML so we can't really search coz every search the static content takes us back online on wiki. Any suggestions in this space would be really helpful, in past I have asked if one can mirror FS wiki but so far it won't work unless FS server allows rsync request as how it works with mirroring php.net with all comments etc. I look forward for more suggestions on the same, Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com From msc at freeswitch.org Fri Sep 4 15:03:40 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 4 Sep 2009 15:03:40 -0700 Subject: [Freeswitch-users] Does FreeSWITCH wiki have update notify? In-Reply-To: <5DBA5DF4-7C42-41C5-AB79-1BC75FAD580E@enterux.com> References: <5DBA5DF4-7C42-41C5-AB79-1BC75FAD580E@enterux.com> Message-ID: <87f2f3b90909041503s46a4af3bg38e7b3e014c00816@mail.gmail.com> I will look into the MediaWiki docs to see what's available. In the meantime you will probable need to use the "recent changes" link on the navigation bar. -MC On Fri, Sep 4, 2009 at 2:39 PM, Mitul Limbani wrote: > Hello, > > It may sound a bit stupid but still wanna ask out here, if there is > any way to replicate FreeSWITCH wiki mirror for local reference or > mainataining local copy instead of Reading online which costs a lot in > developing countries like that of ours and Asia/Africa in general. > > We tried httrack but that brings in everything back here as static > HTML so we can't really search coz every search the static content > takes us back online on wiki. > > Any suggestions in this space would be really helpful, in past I have > asked if one can mirror FS wiki but so far it won't work unless FS > server allows rsync request as how it works with mirroring php.net > with all comments etc. > > I look forward for more suggestions on the same, > > Thanks & Regards, > Mitul Limbani, > Founder & CEO, > Enterux Solutions Pvt. Ltd., > The Enterprise Linux Company (r), > http://www.enterux.com > http://www.entVoice.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/1f7f7941/attachment.html From tina at a2unlimited.com Fri Sep 4 15:40:04 2009 From: tina at a2unlimited.com (Tina Martinez) Date: Fri, 04 Sep 2009 18:40:04 -0400 Subject: [Freeswitch-users] Monitoring FreeSWITCH Events via Telnet socket (Net::Telnet) Message-ID: <34109.1252104004@a2unlimited.com> Hello, I have a question, but I????????m not certain whether this is a FreeSWITCH issue, or something specific to Perl. I have setup a Perl application (???????listener???????) that monitors the events of my FreeSWITCH server via a Telnet socket. So far, the application seems to work very nicely, except that the listener does not consistently capture all of the events that are streaming through the socket. I have been able to work around most of the issues, but one of the more significant pain-points is when a new member is getting added to a conference (Event-Name: CUSTOM, Action: add-member). The log file that I generate from the Telnet socket shows the event details, so I know that the data is coming across the pipe, but I don????????t consistently see the details in my event listener code. Also, the fact that I sometimes do see all of the details confuses me (I don't see a pattern to give any clues towards the cause). BTW - I do have the in the dialplan. So, I wonder, is there something else that I can do in FreeSWITCH to increase the reliability of capturing the event details? or is there something I should be doing in Perl to somehow buffer the data (i.e., why would I see the details in the socket log file, but not in the data stream within the code?). I'm not a socket wizard by any stretch, so I'm hoping that it might be a simple issue related to the Net::Telnet implementation. Any suggestions would be greatly appreciated. Thank you in advance, - Tina From anthony.minessale at gmail.com Fri Sep 4 15:52:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 4 Sep 2009 17:52:35 -0500 Subject: [Freeswitch-users] Monitoring FreeSWITCH Events via Telnet socket (Net::Telnet) In-Reply-To: <34109.1252104004@a2unlimited.com> References: <34109.1252104004@a2unlimited.com> Message-ID: <191c3a030909041552i15b64c64r7964d47bc738e4b3@mail.gmail.com> you should use ESL lib and the supplied perl mod from FS build root cd libs/esl make make perlmod cd perl copy ESL.pm and ESL.so into your INC path see the examples in that same folder. On Fri, Sep 4, 2009 at 5:40 PM, Tina Martinez wrote: > Hello, > > I have a question, but I????????m not certain whether this is a FreeSWITCH > issue, or > something specific to Perl. > > I have setup a Perl application (???????listener?????? ) that monitors the > events of my > FreeSWITCH server via a Telnet socket. So far, the application seems to > work > very nicely, except that the listener does not consistently capture all of > the > events that are streaming through the socket. I have been able to work > around > most of the issues, but one of the more significant pain-points is when a > new > member is getting added to a conference (Event-Name: CUSTOM, Action: > add-member). > > The log file that I generate from the Telnet socket shows the event > details, so I > know that the data is coming across the pipe, but I don????????t > consistently see the > details in my event listener code. Also, the fact that I sometimes do see > all of > the details confuses me (I don't see a pattern to give any clues towards > the cause). > > BTW - I do have the in > the > dialplan. > > So, I wonder, is there something else that I can do in FreeSWITCH to > increase the > reliability of capturing the event details? or is there something I should > be > doing in Perl to somehow buffer the data (i.e., why would I see the details > in > the socket log file, but not in the data stream within the code?). I'm not > a > socket wizard by any stretch, so I'm hoping that it might be a simple issue > related to the Net::Telnet implementation. > > Any suggestions would be greatly appreciated. > > Thank you in advance, > > > - Tina > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/1e253b81/attachment-0001.html From mitul at enterux.com Fri Sep 4 16:26:52 2009 From: mitul at enterux.com (Mitul Limbani) Date: Sat, 5 Sep 2009 04:56:52 +0530 Subject: [Freeswitch-users] Does FreeSWITCH wiki have update notify? In-Reply-To: <87f2f3b90909041503s46a4af3bg38e7b3e014c00816@mail.gmail.com> References: <5DBA5DF4-7C42-41C5-AB79-1BC75FAD580E@enterux.com> <87f2f3b90909041503s46a4af3bg38e7b3e014c00816@mail.gmail.com> Message-ID: <638F6B09-52A9-46B2-A186-D6A4E6CDF732@enterux.com> Mike, I m not sure if we can program httrack to pickup changes automatically (I.e. Looking at recent changes) so this brings us back to square one, instead we can setup media wiki here and do rsync with fs wiki box. Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 05-Sep-2009, at 3:33 AM, Michael Collins wrote: > I will look into the MediaWiki docs to see what's available. In the > meantime you will probable need to use the "recent changes" link on > the navigation bar. > -MC > > On Fri, Sep 4, 2009 at 2:39 PM, Mitul Limbani > wrote: > Hello, > > It may sound a bit stupid but still wanna ask out here, if there is > any way to replicate FreeSWITCH wiki mirror for local reference or > mainataining local copy instead of Reading online which costs a lot in > developing countries like that of ours and Asia/Africa in general. > > We tried httrack but that brings in everything back here as static > HTML so we can't really search coz every search the static content > takes us back online on wiki. > > Any suggestions in this space would be really helpful, in past I have > asked if one can mirror FS wiki but so far it won't work unless FS > server allows rsync request as how it works with mirroring php.net > with all comments etc. > > I look forward for more suggestions on the same, > > Thanks & Regards, > Mitul Limbani, > Founder & CEO, > Enterux Solutions Pvt. Ltd., > The Enterprise Linux Company (r), > http://www.enterux.com > http://www.entVoice.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090905/65f13e8c/attachment.html From rogelio.perez at gmail.com Fri Sep 4 16:28:12 2009 From: rogelio.perez at gmail.com (Rogelio Perez) Date: Fri, 4 Sep 2009 20:28:12 -0300 Subject: [Freeswitch-users] Mod_nibblebill for CDR billing Message-ID: <7F5518A9-FF8A-4E94-9E4A-5FE2CC6D8F50@gmail.com> From the mod_nibblebill documentation: At the end of a call, the module sets a variable named nibble_total_billed. You can use mod_cdr to record this variable to your CDR log. Is it possible to do the same with mod_xml_cdr? I'm looking for a simple way of billing my CDRs and this one looks like a good solution. Has anyone tried doing anything similar? Thanks, Rogelio From brian at freeswitch.org Fri Sep 4 16:32:14 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Sep 2009 18:32:14 -0500 Subject: [Freeswitch-users] Mod_nibblebill for CDR billing In-Reply-To: <7F5518A9-FF8A-4E94-9E4A-5FE2CC6D8F50@gmail.com> References: <7F5518A9-FF8A-4E94-9E4A-5FE2CC6D8F50@gmail.com> Message-ID: <85C17AAF-BDCB-442C-BD49-741084252CDE@freeswitch.org> All the variables are there in XML_CDR too. /b On Sep 4, 2009, at 6:28 PM, Rogelio Perez wrote: > From the mod_nibblebill documentation: > > At the end of a call, the module sets a variable named > nibble_total_billed. You can use mod_cdr to record this variable to > your CDR log. > > Is it possible to do the same with mod_xml_cdr? > I'm looking for a simple way of billing my CDRs and this one looks > like a good solution. > Has anyone tried doing anything similar? > > Thanks, > Rogelio From ujjval at simplesignal.com Fri Sep 4 15:15:17 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Fri, 4 Sep 2009 15:15:17 -0700 Subject: [Freeswitch-users] New install Message-ID: <616E2F0E295A7D4FB44FB9922373715C0582B9BFC6@mbx01.citservers.local> Hi, I just installed freeswitch as a replacement for our Asterisk Server. I want to untimately do Conferencing with it as I have heard is it pretty good at it. I have it compiled and up and running. However, when I provision a Sofphone/Xlite to register with it to run basic tests, it does not seem to register. Looked at freeswitch.log but doesn't have anything related to the REGISTER requests from Xlite. Not too familiar with CLI or configg files yet. Help is appreciated. Also: If there a howto to setup a conferencing Bridge on it. Thx, Ujjval. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/b4a6f629/attachment.html From hjqlopez at hotmail.com Fri Sep 4 09:06:31 2009 From: hjqlopez at hotmail.com (Humberto Quintana) Date: Fri, 4 Sep 2009 12:06:31 -0400 Subject: [Freeswitch-users] 482 Request merged, in serial forking Message-ID: Hello, I'm a new Freeswitch user. After some reading I put Freeswitch (Version 1.0.4) to work as Session Border Controller. I have only one problem that I dont know how to solve it ( or which parameter to set) and I'd appreciate if someone could give me a clue about this. Kamailio is sitting behind FS and it selects the route or routes in case of failure (serial forking) . Freeswitch is configured to use directly the Request-URI sent by Kamailio. So, when the 1st route fails, Kamailio receives the Reply from FS and sends back the ACK to end the transaction. More than 1 second later, a new INVITE from Kamailio with the next route is tried (With the To-header's tag is empty. Same Callid, From and Cseq header but different VIA-header's branch parameter) and FS is answering back 482 Merged Request. It happens the same for the 3rd route. It seems that the transaction is still 'alive' in FS even if the ACK was received ? Thanks, Humberto ===1st route=== U 2009/09/03 17:20:36.069147 kamailio -> freeswitch INVITE sip:5145555555 at gw1 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.0 Call-ID: 1 U 2009/09/03 17:20:36.169147 freeswitch -> gw1 INVITE sip:5145555555 at gw1 SIP/2.0. Call-ID: 2 U 2009/09/03 17:20:36.170158 gw1 -> freeswitch SIP/2.0 100 Trying. Call-ID: 2 U 2009/09/03 17:20:36.190457 gw1 -> freeswitch SIP/2.0 503 Service Unavailable. Call-ID: 2 U 2009/09/03 17:20:36.193296 freeswitch -> gw1 ACK sip:5142776756 at gw1 SIP/2.0. Call-ID: 2 U 2009/09/03 17:20:36.227492 freeswitch -> kamailio SIP/2.0 503 Service Unavailable. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.0 Call-ID: 1 U 2009/09/03 17:20:36.228122 kamailio -> freeswitch ACK sip:5145555555 at gw1 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.0 Call-ID: 1 ===2nd route=== U 2009/09/03 17:20:37.596885 kamailio -> freeswitch INVITE sip:15145555555 at gw2:5061 SIP/2.0 Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.1 Call-ID: 1 U 2009/09/03 17:20:37.597590 freeswitch -> kamailio SIP/2.0 482 Request merged. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.1 Call-ID: 1 U 2009/09/03 17:20:37.598163 kamailio -> freeswitch ACK sip:15145555555 at gw2:5061 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.1 Call-ID: 1 ===3rd route=== U 2009/09/03 17:20:37.642098 kamailio -> freeswitch INVITE sip:5145555555 at gw3 SIP/2.0 Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.2 Call-ID: 1 U 2009/09/03 17:20:37.642634 freeswitch -> kamailio SIP/2.0 482 Request merged. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.2 Call-ID: 1 U 2009/09/03 17:20:37.643139 kamailio -> freeswitch ACK sip:5145555555 at gw3 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKa0fa.0cd8784.2 Call-ID: 1 _________________________________________________________________ Click less, chat more: Messenger on MSN.ca http://go.microsoft.com/?linkid=9677404 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/0b2d2471/attachment-0001.html From brian at freeswitch.org Fri Sep 4 16:34:40 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Sep 2009 18:34:40 -0500 Subject: [Freeswitch-users] 482 Request merged, in serial forking In-Reply-To: References: Message-ID: I'm going to gess the call-id is the same for the second transaction... can you provide a more detailed trace? /b On Sep 4, 2009, at 11:06 AM, Humberto Quintana wrote: > Hello, > > I'm a new Freeswitch user. After some reading I put Freeswitch > (Version 1.0.4) to work as Session Border Controller. I have only > one problem that I dont know how to solve it ( or which parameter to > set) and I'd appreciate if someone could give me a clue about this. > > Kamailio is sitting behind FS and it selects the route or routes in > case of failure (serial forking) . Freeswitch is configured to use > directly the Request-URI sent by Kamailio. > > So, when the 1st route fails, Kamailio receives the Reply from FS > and sends back the ACK to end the transaction. More than 1 second > later, a new INVITE from Kamailio with the next route is tried (With > the To-header's tag is empty. Same Callid, From and Cseq header but > different VIA-header's branch parameter) and FS is answering back > 482 Merged Request. It happens the same for the 3rd route. > > It seems that the transaction is still 'alive' in FS even if the ACK > was received ? > > > Thanks, > > Humberto -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/2d355dc4/attachment.html From brian at freeswitch.org Fri Sep 4 16:35:09 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 4 Sep 2009 18:35:09 -0500 Subject: [Freeswitch-users] New install In-Reply-To: <616E2F0E295A7D4FB44FB9922373715C0582B9BFC6@mbx01.citservers.local> References: <616E2F0E295A7D4FB44FB9922373715C0582B9BFC6@mbx01.citservers.local> Message-ID: make sure your firewall is not up.... /b On Sep 4, 2009, at 5:15 PM, Ujjval Karihaloo wrote: > > Hi, > > I just installed freeswitch as a replacement for our Asterisk > Server. I want to untimately do Conferencing with it as I have heard > is it pretty good at it. > > I have it compiled and up and running. However, when I provision a > Sofphone/Xlite to register with it to run basic tests, it does not > seem to register. Looked at freeswitch.log but doesn?t have anything > related to the REGISTER requests from Xlite. Not too familiar with > CLI or configg files yet. > > Help is appreciated. > > Also: If there a howto to setup a conferencing Bridge on it. > > Thx, > Ujjval. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/d1ba76a4/attachment.html From diego.viola at gmail.com Fri Sep 4 18:28:54 2009 From: diego.viola at gmail.com (Diego Viola) Date: Sat, 5 Sep 2009 01:28:54 +0000 Subject: [Freeswitch-users] Mod_nibblebill for CDR billing In-Reply-To: <7F5518A9-FF8A-4E94-9E4A-5FE2CC6D8F50@gmail.com> References: <7F5518A9-FF8A-4E94-9E4A-5FE2CC6D8F50@gmail.com> Message-ID: <86a32abc0909041828n1fc4b55es47898f408c5fc0d7@mail.gmail.com> If you do "event plain all" from the FS CLI you should see the variable exported on the CHANNEL_HANGUP_COMPLETE event, with the other CDR variables as well. These information should be available on mod_xml_cdr and mod_cdr_csv as well. Diego On Fri, Sep 4, 2009 at 11:28 PM, Rogelio Perez wrote: > From the mod_nibblebill documentation: > > At the end of a call, the module sets a variable named > nibble_total_billed. You can use mod_cdr to record this variable to > your CDR log. > > Is it possible to do the same with mod_xml_cdr? > I'm looking for a simple way of billing my CDRs and this one looks > like a good solution. > Has anyone tried doing anything similar? > > Thanks, > Rogelio > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090905/796619d9/attachment.html From djbinter at yahoo.com Fri Sep 4 16:54:58 2009 From: djbinter at yahoo.com (DJB) Date: Fri, 4 Sep 2009 16:54:58 -0700 (PDT) Subject: [Freeswitch-users] Call Transfer Problem Message-ID: <504742.69041.qm@web37507.mail.mud.yahoo.com> I have a call transfer problem with Freeswitch Here is the call flow: I call from the PSTN ?(A party) into my Polycom phone (B-party) which is registered to FreeSwtich. The Freeswtich is setup not to route media as I have an SBC acting as a mirror proxy that will do all the NAT and media routing. The inbound call is setup fine and there is two way voice. I then blind transfer from the Polycom to my Cell phone. I see the polycom send a SIP refer to Freeswitch and it sends a 202 accepted fine and that leg between the Polycom (B party) and the A party is torn down fine like its supposed to be. The Freeswitch places the outbound call (the number the call is transferring to C-party) and that call completes. However now there is one way audio between the A party and C party . I see RTP streaming back from the egress carrier where the call was transfered to so the A party can hear the C party but the C party cannot hear the A party . When I look at the SIP traces of the original inbound call from the A-party I see a SIP re-invite from free switch to place the call on hold (contains Freeswitch RTP address to I can hear hold music) while it is transferring the call and the A-party does hear on hold music from Freeswitch while the call is being transferred. However I do not see a second re-invite from freeswitch to pass the media IP it got from the egress leg back to the original inbound leg. If my inbound gateway does not get a re-invite from Freeswitch to redirect its media to the new RTP address of of the egress carrier it will not do so hence the one way voice. How do I get the Freeswitch to re-invite the original ingress leg once it gets the SIP 183 from the egress with the new RTP info ? Free switch is sending the first SIP re-invite to my inbound gateway with new media IP (IP of itself) so the A-party can hear on hold music , but does not send a second re-invite to my inbound gateway after it receives the new RTP address from the egress carrier for the call that was transferred back out. ? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/d736afba/attachment.html From ujjval at simplesignal.com Fri Sep 4 22:03:14 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Fri, 4 Sep 2009 22:03:14 -0700 Subject: [Freeswitch-users] New install In-Reply-To: References: <616E2F0E295A7D4FB44FB9922373715C0582B9BFC6@mbx01.citservers.local> Message-ID: <616E2F0E295A7D4FB44FB9922373715C0582B9BFE8@mbx01.citservers.local> Would that be firewall on the CentOS machine that FS is installed on? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, September 04, 2009 5:35 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] New install make sure your firewall is not up.... /b On Sep 4, 2009, at 5:15 PM, Ujjval Karihaloo wrote: Hi, I just installed freeswitch as a replacement for our Asterisk Server. I want to untimately do Conferencing with it as I have heard is it pretty good at it. I have it compiled and up and running. However, when I provision a Sofphone/Xlite to register with it to run basic tests, it does not seem to register. Looked at freeswitch.log but doesn't have anything related to the REGISTER requests from Xlite. Not too familiar with CLI or configg files yet. Help is appreciated. Also: If there a howto to setup a conferencing Bridge on it. Thx, Ujjval. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/6a275391/attachment-0001.html From msc at freeswitch.org Fri Sep 4 23:07:21 2009 From: msc at freeswitch.org (Michael S Collins) Date: Fri, 4 Sep 2009 23:07:21 -0700 Subject: [Freeswitch-users] New install In-Reply-To: <616E2F0E295A7D4FB44FB9922373715C0582B9BFE8@mbx01.citservers.local> References: <616E2F0E295A7D4FB44FB9922373715C0582B9BFC6@mbx01.citservers.local> <616E2F0E295A7D4FB44FB9922373715C0582B9BFE8@mbx01.citservers.local> Message-ID: <40448827-5C9E-499D-B2D8-8629B9E1667C@freeswitch.org> Sent from my iPhone On Sep 4, 2009, at 10:03 PM, Ujjval Karihaloo wrote: > Would that be firewall on the CentOS machine that FS is installed on? > > Correct > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Brian West > Sent: Friday, September 04, 2009 5:35 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] New install > > > > make sure your firewall is not up.... > > > > /b > > > > On Sep 4, 2009, at 5:15 PM, Ujjval Karihaloo wrote: > > > > > > > Hi, > > > > I just installed freeswitch as a replacement for our Asterisk > Server. I want to untimately do Conferencing with it as I have heard > is it pretty good at it. > > > > I have it compiled and up and running. However, when I provision a > Sofphone/Xlite to register with it to run basic tests, it does not > seem to register. Looked at freeswitch.log but doesn?t have anything > related to the REGISTER requests from Xlite. Not too familiar with > CLI or configg files yet. > > > > Help is appreciated. > > > > Also: If there a howto to setup a conferencing Bridge on it. > > > > Thx, > > Ujjval. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090904/a28d078f/attachment.html From mayamatakeshi at gmail.com Fri Sep 4 23:40:01 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Sat, 5 Sep 2009 15:40:01 +0900 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> Message-ID: <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> On Fri, Sep 4, 2009 at 7:28 PM, mayamatakeshi wrote: > > On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi wrote: > >> Hello, >> I'm testing FS support for the header Path (FS is behind opensips). >> It pretty much works: I tested calling from one user to the other and >> calls work perfectly. >> However, I've noticed that when I register my terminal directly with FS >> without going thru the proxy, I receive an unsolicited NOTIFY containing >> Message-Waiting information. But when I register via proxy, FS doesn't send >> this NOTIFY. >> What could be causing this difference of behavior? (enabling debug (F8) >> doesn't show anything for registration handling). >> > > I have enabled Sofia debug and I can see NTA is complaining about invalid > URI when building the NOTIFY: > > nua: nua_notify: entering > nua(0x9b3c1e8): sent signal r_notify > nua(0x9b3c1e8): recv signal r_notify > nua: nua_stack_set_params: entering > nua(0x9b3c1e8): adding notify usage with event message-summary > nta_leg_tcreate(0x9b74c68) > nta outgoing create: invalid URI > nta: outgoing_free(0x9b74928) > nua(0x9b3c1e8): event r_notify 900 Internal error at nua_client.c:711 > nua(0x9b3c1e8): removing notify usage with event message-summary > > My REGISTER relayed by opensips is this: > > REGISTER sip:test.com SIP/2.0 > Record-Route: > > Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKe62c.49e9f6c1.0 > Via: SIP/2.0/UDP 192.168.2.121:5060 > ;received=192.168.2.121;rport=5060;branch=z9hG4bKPj4uAYgDuRbilYy4lCWcjlDKIDAtf-9RdS > Max-Forwards: 69 > From: > >;tag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5 > To: > > Call-ID: JvQ.apMLiJtfHa7z4ShIfgBPi5jIbtBv > CSeq: 14872 REGISTER > Contact: > Expires: 60 > Authorization: Digest username="user1", realm="test.com", > nonce="7d911eef-2c16-4deb-99f6-afcff9968a19", uri="sip:192.168.2.100", > response="df29caeb78790b4527f1176622cbf192", algorithm=MD5, > cnonce="5.EXCbM3RZTx6iOh1cvUzUvEZTs2eheG", qop=auth, nc=00000001 > Content-Length: 0 > Path: > ;lr;received=sip:192.168.2.121:5060> > > I hope someone can point out a problem. > I'm looking at NTA with gdb but I'm slow on this. The invalid URI nta is complaining about is the route_uri extracted from the Contact stored upon registration. The difference of behavior between INVITE (works) and NOTIFY (doesn't work) via proxy, seems to be because for INVITE, mod_sofia code (function sofia_glue_do_invite in sofia_glue.c) calls sofia_overcome_sip_uri_weakness to adjust the route_uri. But for a NOTIFY, this function is not called (and it cannot be called, as there's no session which is required as a parameter). In my case I can see that basically what sofia_overcome_sip_uri_weakness does is to remove the "<" , ">" around the route_uri. I messed with the code in sofia_glue_send_notify to just remove "<" and ">" and after that I was able to receive the NOTIFY. So I believe there is some code lacking in FS to properly permit UAs registering via proxy to receive NOTIFY. I might be wrong: if there is anyone using this scenario successfully, please let me know. Otherwise, I'll open a ticket on JIRA. regards, takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090905/b759d268/attachment.html From juanbackson at gmail.com Sat Sep 5 01:36:13 2009 From: juanbackson at gmail.com (Juan Backson) Date: Sat, 5 Sep 2009 16:36:13 +0800 Subject: [Freeswitch-users] No dial-string available error Message-ID: <27c25bc40909050136n4ea87c08xdc541874e979d4f7@mail.gmail.com> Hi, I am getting no dial-string available error when using xml_odbc module to bridge a call. How can I resolve this problem? EXECUTE sofia/internal/180001 at 192.168.1.130 bridge(user/180001) 2009-09-05 16:31:29.853456 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [key_value]=[192.168.1.134] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [key_name]=[name] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [tag_name]=[domain] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [section]=[directory] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Name]=[REQUEST_PARAMS] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Core-UUID]=[f649cdfd-3715-4d18-94d9-417aa7e26873] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [FreeSWITCH-Hostname]=[server] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [FreeSWITCH-IPv4]=[192.168.1.134] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [FreeSWITCH-IPv6]=[::1] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Date-Local]=[2009-09-05 16:31:29] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Date-GMT]=[Sat, 05 Sep 2009 08:31:29 GMT] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Date-Timestamp]=[1252139489853456] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Calling-File]=[mod_dptools.c] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Calling-Function]=[user_outgoing_channel] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [Event-Calling-Line-Number]=[2365] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [as_channel]=[true] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [key]=[id] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [user]=[180001] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in xml_odbc_search, header [domain]=[192.168.1.134] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:319 DEBUG GOING TO RENDER TEMPLATE [default] 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:214 DEBUG Performing Query: SELECT enabled, sip_password, account_id FROM agent WHERE sip_user = 180001 2009-09-05 16:31:29.855416 [INFO] mod_xml_odbc.c:214 DEBUG Performing Query: SELECT sip_password FROM agent WHERE sip_user = 180001 2009-09-05 16:31:29.855416 [INFO] mod_xml_odbc.c:417 Debug dump of generated XML:
2009-09-05 16:31:29.856442 [CONSOLE] mod_xml_odbc.c:457 Generated XML is in [/tmp/16d2f01a-4332-43ff-9535-5615a192b40e.tmp.xml] 2009-09-05 16:31:29.856442 [ERR] mod_dptools.c:2430 No dial-string available, please check your user directory. 2009-09-05 16:31:29.856442 [ERR] switch_ivr_originate.c:1527 Cannot create outgoing channel of type [user] cause: [MANDATORY_IE_MISSING] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090905/4b54db41/attachment.html From tculjaga at gmail.com Sat Sep 5 01:40:25 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sat, 5 Sep 2009 10:40:25 +0200 Subject: [Freeswitch-users] New install In-Reply-To: <616E2F0E295A7D4FB44FB9922373715C0582B9BFC6@mbx01.citservers.local> References: <616E2F0E295A7D4FB44FB9922373715C0582B9BFC6@mbx01.citservers.local> Message-ID: <65d96fc80909050140o51ee814oe1b12921498cb162@mail.gmail.com> it may be that you have 2 configured nics and FS hooks up to your 1st one only ignoring the 2nd one. This happend to me. You need to configure the 2nd IP address explicitly. T. On Sat, Sep 5, 2009 at 12:15 AM, Ujjval Karihaloo wrote: > > > Hi, > > > > I just installed freeswitch as a replacement for our Asterisk Server. I > want to untimately do Conferencing with it as I have heard is it pretty good > at it. > > > > I have it compiled and up and running. However, when I provision a > Sofphone/Xlite to register with it to run basic tests, it does not seem to > register. Looked at freeswitch.log but doesn?t have anything related to the > REGISTER requests from Xlite. Not too familiar with CLI or configg files > yet. > > > > Help is appreciated. > > > > Also: If there a howto to setup a conferencing Bridge on it. > > > > Thx, > > Ujjval. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090905/8d7d3f59/attachment-0001.html From gmaruzz at celliax.org Sat Sep 5 01:49:53 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 5 Sep 2009 10:49:53 +0200 Subject: [Freeswitch-users] mod_skypiax for OSX????? Message-ID: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> Seeeeeeeven! I saw the modification you made on the wiki page... You made it, mod_skypiax runs on OSX!!!! Let's merge your mods on the mainline, pleaaaase ;-))) -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From dujinfang at gmail.com Sat Sep 5 04:13:03 2009 From: dujinfang at gmail.com (Seven Du) Date: Sat, 5 Sep 2009 19:13:03 +0800 Subject: [Freeswitch-users] mod_skypiax for OSX????? In-Reply-To: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> References: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> Message-ID: <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> gm, Thanks a lot you can merge into the mainline. I check into my branch because it's currently not as useful as on Linux and Windows and the solution is not good. But it works and it is a good start that mod_skypiax runs on OSX. Sure it would be easier for people want to test and improve it if it been merged into trunk. I think you can make a diff by svn diff -r 14472:14772 http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax FYI for personal reason I won't have much time put on this in the coming month. Actually the code was done a few weeks ago but i only got a chance to commit it yesterday. Sure that is not to say I cannot do but fixes. But can you please make sure it won't break Linux/ windows build when you merge the code? I haven't have a chance to test all of them yet. -7- On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: > Seeeeeeeven! > > I saw the modification you made on the wiki page... > > You made it, mod_skypiax runs on OSX!!!! > > Let's merge your mods on the mainline, pleaaaase ;-))) > > -giovanni > > > > > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at celliax.org Sat Sep 5 04:41:43 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sat, 5 Sep 2009 13:41:43 +0200 Subject: [Freeswitch-users] mod_skypiax for OSX????? In-Reply-To: <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> References: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> Message-ID: <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> Seven, thanks a lot for your efforts. I will merge it in the next days, and I will take care that it will not breaks Windows or Linux. If I find problems I will wait for you having more time in the future. I send you my super best wishes for your personal things to go well and solves in the best of the possible ways. ciao for now, -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 On Sat, Sep 5, 2009 at 1:13 PM, Seven Du wrote: > gm, > > Thanks a lot you can merge into the mainline. I check into my branch > because it's currently not as useful as on Linux and Windows and the > solution is not good. But it works and it is a good start that > mod_skypiax runs on OSX. Sure it would be easier for people want to > test and improve it if it been merged into trunk. I think you can make > a diff by > > svn diff -r 14472:14772 http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax > > FYI for personal reason I won't have much time put on this in the > coming month. Actually the code was done a few weeks ago but i only > got a chance to commit it yesterday. Sure that is not to say I cannot > do but fixes. But can you please make sure it won't break Linux/ > windows build when you merge the code? I haven't have a chance to test > all of them yet. > > -7- > > On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: >> Seeeeeeeven! >> >> I saw the modification you made on the wiki page... >> >> You made it, mod_skypiax runs on OSX!!!! >> >> Let's merge your mods on the mainline, pleaaaase ;-))) >> >> -giovanni >> >> >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From tina at a2unlimited.com Sat Sep 5 05:11:00 2009 From: tina at a2unlimited.com (Tina Martinez) Date: Sat, 05 Sep 2009 08:11:00 -0400 Subject: [Freeswitch-users] Monitoring FreeSWITCH Events via Telnet socket Message-ID: <53595.1252152660@a2unlimited.com> Thank you for the guidance towards the ESL lib, but I'm encountering some problems. ----> make perlmod When I attempt to execute the make perlmod, I currently get the following error: /usr/bin/ld: cannot find -ldb collect2: ld returned 1 exit status make[1]: *** [ESL.so] Error 1 Prior to this, I encountered a similar error (cannot find -lgdbm) where I had to create a link to fix: libgdbm.so -> /usr/lib64/libgdbm.so.2.0.0 I'm suspecting that my installation of Perl is not looking in the lib64 directory as it should? Unfortunately my Perl skills are not sufficient to be hacking around with this stuff, anyone with thoughts or suggestions? From brian at freeswitch.org Sat Sep 5 07:44:21 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 5 Sep 2009 09:44:21 -0500 Subject: [Freeswitch-users] Monitoring FreeSWITCH Events via Telnet socket In-Reply-To: <53595.1252152660@a2unlimited.com> References: <53595.1252152660@a2unlimited.com> Message-ID: <839BB258-7E27-4DDC-AF45-89E277CF6044@freeswitch.org> You need to install db4-devel. /b On Sep 5, 2009, at 7:11 AM, Tina Martinez wrote: > When I attempt to execute the make perlmod, I currently get the > following error: > > /usr/bin/ld: cannot find -ldb > collect2: ld returned 1 exit status > make[1]: *** [ESL.so] Error 1 From brian at freeswitch.org Sat Sep 5 07:44:42 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 5 Sep 2009 09:44:42 -0500 Subject: [Freeswitch-users] Monitoring FreeSWITCH Events via Telnet socket In-Reply-To: <53595.1252152660@a2unlimited.com> References: <53595.1252152660@a2unlimited.com> Message-ID: <5BF8E289-5083-4AD2-9561-B6DE0EAC8FB6@freeswitch.org> No you should have installed gdbm-devel. /b On Sep 5, 2009, at 7:11 AM, Tina Martinez wrote: > Prior to this, I encountered a similar error (cannot find -lgdbm) > where I had to > create a link to fix: From woodydickson at gmail.com Sat Sep 5 08:13:49 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sat, 5 Sep 2009 23:13:49 +0800 Subject: [Freeswitch-users] auto expiration Message-ID: Hi, I would like to set up freeswitch to automatically expire a user registration if either NOTIFY or REGISTER is not received within certain time frame. Does anyone know how to do that? Thanks, Woody -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090905/3758dd4a/attachment.html From mayamatakeshi at gmail.com Sat Sep 5 08:23:20 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Sun, 6 Sep 2009 00:23:20 +0900 Subject: [Freeswitch-users] auto expiration In-Reply-To: References: Message-ID: <15b9404e0909050823i5b936eadka3dbb7ce31e70f81@mail.gmail.com> On Sun, Sep 6, 2009 at 12:13 AM, Woody Dickson wrote: > Hi, > > I would like to set up freeswitch to automatically expire a user > registration if either NOTIFY or REGISTER is not received within certain > time frame. > > Does anyone know how to do that? > I don't know about NOTIFY. But for REGISTER: http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Force_Registrations_to_Expire regards, takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090906/9751e10a/attachment.html From juanbackson at gmail.com Sat Sep 5 09:19:05 2009 From: juanbackson at gmail.com (Juan Backson) Date: Sun, 6 Sep 2009 00:19:05 +0800 Subject: [Freeswitch-users] chat api problem Message-ID: <27c25bc40909050919x48fe3094x768bc90ea57003a4@mail.gmail.com> Hi, I am not able to get the chat api to work. Any idea why the following api command is not working? freeswitch at localhost.localdomain> chat sip|180001 at 192.168.1.102|testing API CALL [chat(sip|180001 at 192.168.1.102|testing)] output: Invalid freeswitch at localhost.localdomain> thx, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090906/5faf7681/attachment.html From larclap at yahoo.com Sat Sep 5 10:13:13 2009 From: larclap at yahoo.com (Lars Zeb) Date: Sat, 5 Sep 2009 10:13:13 -0700 Subject: [Freeswitch-users] restart when convenient In-Reply-To: <4AA0ADAE.2090709@xpirio.com> References: <4AA0ADAE.2090709@xpirio.com> Message-ID: <002f01ca2e4c$27c2f200$7748d600$@com> Try: /usr/local/freeswitch/bin/fs_cli -x 'fsctl shutdown elegant restart' -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Christian L?schenkohl Sent: Thursday, September 03, 2009 11:03 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] restart when convenient hello i'm looking for a possibility to restart freeswitch like it is possible with asterisk. for me i tried to created a script that looks for open channels and if no channel is open it restarts freeswitch with the init script (not the most efficient way). i think i would be great if we would have a buildin function for this, i think such command would help with maintenance and not only for me. br -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From juanbackson at gmail.com Sat Sep 5 11:26:08 2009 From: juanbackson at gmail.com (Juan Backson) Date: Sun, 6 Sep 2009 02:26:08 +0800 Subject: [Freeswitch-users] Chat redirect Message-ID: <27c25bc40909051126t619e34e0q85fa8b097a61240f@mail.gmail.com> Hi Is there anyway to use freeswitch to redirect chat message the same that it redirects SIP message? If so, how can it be done? thx, jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090906/f9943b7a/attachment.html From brian at freeswitch.org Sat Sep 5 11:36:01 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 5 Sep 2009 13:36:01 -0500 Subject: [Freeswitch-users] Chat redirect In-Reply-To: <27c25bc40909051126t619e34e0q85fa8b097a61240f@mail.gmail.com> References: <27c25bc40909051126t619e34e0q85fa8b097a61240f@mail.gmail.com> Message-ID: Not automatically. But you could use the event socket to get the message and forward it via ESL. /b On Sep 5, 2009, at 1:26 PM, Juan Backson wrote: > > If so, how can it be done? From brian at freeswitch.org Sat Sep 5 11:37:31 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 5 Sep 2009 13:37:31 -0500 Subject: [Freeswitch-users] chat api problem In-Reply-To: <27c25bc40909050919x48fe3094x768bc90ea57003a4@mail.gmail.com> References: <27c25bc40909050919x48fe3094x768bc90ea57003a4@mail.gmail.com> Message-ID: <94465B0F-4190-4900-94C8-F5D03B92F687@freeswitch.org> You're missing some required args there. chat ||||[] /b On Sep 5, 2009, at 11:19 AM, Juan Backson wrote: > freeswitch at localhost.localdomain> chat sip|180001 at 192.168.1.102| > testing > API CALL [chat(sip|180001 at 192.168.1.102|testing)] output: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090905/ce8c0db2/attachment.html From tparikh at gmail.com Sat Sep 5 16:25:49 2009 From: tparikh at gmail.com (Tapan Parikh) Date: Sat, 5 Sep 2009 16:25:49 -0700 Subject: [Freeswitch-users] mod_skypiax for OSX????? In-Reply-To: <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> References: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> Message-ID: <1ecdcb6a0909051625ud2f8dbbjf327e6e1ac8a21d1@mail.gmail.com> Yes, thanks a lot! Im still having a bit of trouble getting it working though. When skypiax_proxy starts, I get the following errors: 2009-09-05 15:39:40.631 skypiax_proxy[77842:10b] Failed to init theDOProxy 2009-09-05 15:39:42.139 skypiax_proxy[77842:10b] Failed to init theDOProxy Then, when an incoming call comes in, I see the messages: Message received CALL 683 STATUS RINGING Sent 23 bytes to FreeSWITCH Sending to skype: GET CALL 683 PARTNER_HANDLE Send to skype: GET CALL 683 PARTNER_HANDLE Message received CALL 683 CONF_ID 0 Sent 18 bytes to FreeSWITCH Message received CALL 683 FAILUREREASON 14 Sent 25 bytes to FreeSWITCH Message received CALL 683 STATUS MISSED Sent 22 bytes to FreeSWITCH Message received CALL 683 SEEN FALSE Sent 19 bytes to FreeSWITCH But the call is not actually picked up by the proxy. Any ideas? Im on Mac OS X 10.5, w/ freeswitch from SVN. Thanks! On Sat, Sep 5, 2009 at 4:41 AM, Giovanni Maruzzelli wrote: > Seven, > > thanks a lot for your efforts. > > I will merge it in the next days, and I will take care that it will > not breaks Windows or Linux. > > If I find problems I will wait for you having more time in the future. > > I send you my super best wishes for your personal things to go well > and solves in the best of the possible ways. > > ciao for now, > > -giovanni > > > > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > > > On Sat, Sep 5, 2009 at 1:13 PM, Seven Du wrote: > > gm, > > > > Thanks a lot you can merge into the mainline. I check into my branch > > because it's currently not as useful as on Linux and Windows and the > > solution is not good. But it works and it is a good start that > > mod_skypiax runs on OSX. Sure it would be easier for people want to > > test and improve it if it been merged into trunk. I think you can make > > a diff by > > > > svn diff -r 14472:14772 > http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax > > > > FYI for personal reason I won't have much time put on this in the > > coming month. Actually the code was done a few weeks ago but i only > > got a chance to commit it yesterday. Sure that is not to say I cannot > > do but fixes. But can you please make sure it won't break Linux/ > > windows build when you merge the code? I haven't have a chance to test > > all of them yet. > > > > -7- > > > > On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: > >> Seeeeeeeven! > >> > >> I saw the modification you made on the wiki page... > >> > >> You made it, mod_skypiax runs on OSX!!!! > >> > >> Let's merge your mods on the mainline, pleaaaase ;-))) > >> > >> -giovanni > >> > >> > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090905/f42a7140/attachment.html From wiltingtree at gmail.com Sat Sep 5 20:25:07 2009 From: wiltingtree at gmail.com (Adam Wilt) Date: Sat, 5 Sep 2009 23:25:07 -0400 Subject: [Freeswitch-users] how to call mod_commands from within a lua script Message-ID: Hi, the documentation says that mod_commands is available from within mod_lua. But when I try to access it like this: session:execute("uuid_broadcast",session_id .. " " .. filename .. " both") I get: Invalid Application uuid_broadcast or session:execute("bgapi","uuid_broadcast " .. session_id .. " " .. filename .. " both") I get: Invalid Application bgapi or session:uuid_broadcast(session_id .. " " .. filename .. " both") I get: attempt to call field 'uuid_broadcast' (a nil value) Can somebody please explain how to do this? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090905/390c7d23/attachment.html From mayamatakeshi at gmail.com Sat Sep 5 21:08:00 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Sun, 6 Sep 2009 13:08:00 +0900 Subject: [Freeswitch-users] No dial-string available error In-Reply-To: <27c25bc40909050136n4ea87c08xdc541874e979d4f7@mail.gmail.com> References: <27c25bc40909050136n4ea87c08xdc541874e979d4f7@mail.gmail.com> Message-ID: <15b9404e0909052108j3842ef91ua13c54895a3959a5@mail.gmail.com> On Sat, Sep 5, 2009 at 5:36 PM, Juan Backson wrote: > Hi, > > I am getting no dial-string available error when using xml_odbc module to > bridge a call. How can I resolve this problem? > Hello, I never tried the mod_xml_odbc. But as the message says, you are not providing a dial-string. I believe your template is incomplete. Read about the dial-string here: http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Dial_String > > EXECUTE sofia/internal/180001 at 192.168.1.130 bridge(user/180001) > 2009-09-05 16:31:29.853456 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [key_value]=[192.168.1.134] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [key_name]=[name] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [tag_name]=[domain] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [section]=[directory] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Name]=[REQUEST_PARAMS] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Core-UUID]=[f649cdfd-3715-4d18-94d9-417aa7e26873] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [FreeSWITCH-Hostname]=[server] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [FreeSWITCH-IPv4]=[192.168.1.134] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [FreeSWITCH-IPv6]=[::1] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Date-Local]=[2009-09-05 16:31:29] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Date-GMT]=[Sat, 05 Sep 2009 08:31:29 GMT] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Date-Timestamp]=[1252139489853456] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Calling-File]=[mod_dptools.c] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Calling-Function]=[user_outgoing_channel] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Calling-Line-Number]=[2365] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [as_channel]=[true] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [key]=[id] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [user]=[180001] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [domain]=[192.168.1.134] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:319 DEBUG GOING TO RENDER > TEMPLATE [default] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:214 DEBUG Performing > Query: > SELECT enabled, sip_password, account_id FROM agent WHERE sip_user = > 180001 > 2009-09-05 16:31:29.855416 [INFO] mod_xml_odbc.c:214 DEBUG Performing > Query: > SELECT sip_password FROM agent WHERE sip_user = 180001 > 2009-09-05 16:31:29.855416 [INFO] mod_xml_odbc.c:417 Debug dump of > generated XML: > >
> > > > > > > > > > >
> > 2009-09-05 16:31:29.856442 [CONSOLE] mod_xml_odbc.c:457 Generated XML is in > [/tmp/16d2f01a-4332-43ff-9535-5615a192b40e.tmp.xml] > 2009-09-05 16:31:29.856442 [ERR] mod_dptools.c:2430 No dial-string > available, please check your user directory. > 2009-09-05 16:31:29.856442 [ERR] switch_ivr_originate.c:1527 Cannot create > outgoing channel of type [user] cause: [MANDATORY_IE_MISSING] > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090906/81be5e3e/attachment-0001.html From ujjval at simplesignal.com Sat Sep 5 22:57:32 2009 From: ujjval at simplesignal.com (Ujjval Karihaloo) Date: Sat, 5 Sep 2009 22:57:32 -0700 Subject: [Freeswitch-users] Conferencing setup with FS Message-ID: <616E2F0E295A7D4FB44FB9922373715C0582B9BFF2@mbx01.citservers.local> Hi , I cannot seem to find a Document online for setting up conferencingon FreeSwitch. Can someone point me to one? Thx. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090905/e1adb7dd/attachment.html From jason at jasonjgw.net Sat Sep 5 23:16:38 2009 From: jason at jasonjgw.net (Jason White) Date: Sun, 6 Sep 2009 16:16:38 +1000 Subject: [Freeswitch-users] Conferencing setup with FS In-Reply-To: <616E2F0E295A7D4FB44FB9922373715C0582B9BFF2@mbx01.citservers.local> References: <616E2F0E295A7D4FB44FB9922373715C0582B9BFF2@mbx01.citservers.local> Message-ID: <20090906061638.GA15591@jdc.jasonjgw.net> Ujjval Karihaloo wrote: > > I cannot seem to find a Document online for setting up conferencingon > FreeSwitch. Can someone point me to one? Have a look at http://wiki.freeswitch.org/ and search for "conference". There's a document describing mod_conference there. Also look at the default conference configuration supplied with FreeSWITCH. From ivan at myrvold.org Sun Sep 6 00:19:58 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Sun, 6 Sep 2009 09:19:58 +0200 Subject: [Freeswitch-users] mod_skypiax for OSX????? In-Reply-To: <1ecdcb6a0909051625ud2f8dbbjf327e6e1ac8a21d1@mail.gmail.com> References: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> <1ecdcb6a0909051625ud2f8dbbjf327e6e1ac8a21d1@mail.gmail.com> Message-ID: I have got outgoing call to Skype to work, and the audio quality is excellent. But I also have problems with incoming call, looks like it doesn't get bridged by FreeSWITCH. I put sofia on debug, but couldn't see any Sofia messages at all. Only message I saw in the console was this: 2009-09-05 20:44:39.720943 [WARNING] skypiax_protocol.c:375 rev 14772[0x0|37 ][WARNINGA 375 ][interface1][-1, 0, 0] skype_call: 332, STATUS: RINGINGCALL is not recognized But I am thrilled that Seven have got skypiax to work this far, and am confident that we will have it worked both incoming and outgoing soon. Ivan Den 6. sep.. 2009 kl. 01:25 skrev Tapan Parikh: > > Yes, thanks a lot! > > Im still having a bit of trouble getting it working though. When > skypiax_proxy starts, I get the following errors: > > 2009-09-05 15:39:40.631 skypiax_proxy[77842:10b] Failed to init > theDOProxy > 2009-09-05 15:39:42.139 skypiax_proxy[77842:10b] Failed to init > theDOProxy > > Then, when an incoming call comes in, I see the messages: > > Message received CALL 683 STATUS RINGING > Sent 23 bytes to FreeSWITCH > Sending to skype: GET CALL 683 PARTNER_HANDLE > Send to skype: GET CALL 683 PARTNER_HANDLE > Message received CALL 683 CONF_ID 0 > Sent 18 bytes to FreeSWITCH > Message received CALL 683 FAILUREREASON 14 > Sent 25 bytes to FreeSWITCH > Message received CALL 683 STATUS MISSED > Sent 22 bytes to FreeSWITCH > Message received CALL 683 SEEN FALSE > Sent 19 bytes to FreeSWITCH > > But the call is not actually picked up by the proxy. Any ideas? Im > on Mac OS X 10.5, w/ freeswitch from SVN. Thanks! > > > On Sat, Sep 5, 2009 at 4:41 AM, Giovanni Maruzzelli > wrote: > Seven, > > thanks a lot for your efforts. > > I will merge it in the next days, and I will take care that it will > not breaks Windows or Linux. > > If I find problems I will wait for you having more time in the future. > > I send you my super best wishes for your personal things to go well > and solves in the best of the possible ways. > > ciao for now, > > -giovanni > > > > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > > > On Sat, Sep 5, 2009 at 1:13 PM, Seven Du wrote: > > gm, > > > > Thanks a lot you can merge into the mainline. I check into my branch > > because it's currently not as useful as on Linux and Windows and the > > solution is not good. But it works and it is a good start that > > mod_skypiax runs on OSX. Sure it would be easier for people want to > > test and improve it if it been merged into trunk. I think you can > make > > a diff by > > > > svn diff -r 14472:14772 http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax > > > > FYI for personal reason I won't have much time put on this in the > > coming month. Actually the code was done a few weeks ago but i only > > got a chance to commit it yesterday. Sure that is not to say I > cannot > > do but fixes. But can you please make sure it won't break Linux/ > > windows build when you merge the code? I haven't have a chance to > test > > all of them yet. > > > > -7- > > > > On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: > >> Seeeeeeeven! > >> > >> I saw the modification you made on the wiki page... > >> > >> You made it, mod_skypiax runs on OSX!!!! > >> > >> Let's merge your mods on the mainline, pleaaaase ;-))) > >> > >> -giovanni > >> > >> > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090906/681c93f8/attachment.html From msc at freeswitch.org Sun Sep 6 00:36:35 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sun, 6 Sep 2009 00:36:35 -0700 Subject: [Freeswitch-users] how to call mod_commands from within a lua script Message-ID: <3C02E6BE-496A-4AFA-AD2B-7AA050C008E6@freeswitch.org> On Sep 5, 2009, at 8:25 PM, Adam Wilt wrote: > Hi, the documentation says that mod_commands is available from > within mod_lua. But when I try to access it like this: > > session:execute("uuid_broadcast",session_id .. " " .. filename .. " > both") > > I get: Invalid Application uuid_broadcast > > or > > session:execute("bgapi","uuid_broadcast " .. session_id .. " " .. > filename .. " both") > > I get: Invalid Application bgapi > > or > > session:uuid_broadcast(session_id .. " " .. filename .. " both") > > I get: attempt to call field 'uuid_broadcast' (a nil value) > > > > Can somebody please explain how to do this? > > Thanks! > Absolutely. Using session:execute is only for dialplan applications. To do API commands, including all the stuff in mod_commands, you need to create an API object: api = freeswitch.API() Then send a command and the variable will receive the results: reply = api:executeString("version") Check out this page for more examples: http://wiki.freeswitch.org/wiki/Make_API_calls_directly_from_Lua_code -MC > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dujinfang at gmail.com Sun Sep 6 01:09:45 2009 From: dujinfang at gmail.com (Seven Du) Date: Sun, 6 Sep 2009 16:09:45 +0800 Subject: [Freeswitch-users] mod_skypiax for OSX????? In-Reply-To: References: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> <1ecdcb6a0909051625ud2f8dbbjf327e6e1ac8a21d1@mail.gmail.com> Message-ID: I'm not sure this and don't have time to debug. But last time I tried incoming calls worked by setting skype to Auto-answer calls. Can you try that? On Sep 6, 2009, at 3:19 PM, Ivan C Myrvold wrote: > I have got outgoing call to Skype to work, and the audio quality is > excellent. But I also have problems with incoming call, looks like > it doesn't get bridged by FreeSWITCH. I put sofia on debug, but > couldn't see any Sofia messages at all. Only message I saw in the > console was this: > > 2009-09-05 20:44:39.720943 [WARNING] skypiax_protocol.c:375 rev > 14772[0x0|37 ][WARNINGA 375 ][interface1][-1, 0, 0] > skype_call: 332, STATUS: RINGINGCALL is not recognized > > But I am thrilled that Seven have got skypiax to work this far, and > am confident that we will have it worked both incoming and outgoing > soon. > > Ivan > > Den 6. sep.. 2009 kl. 01:25 skrev Tapan Parikh: > >> >> Yes, thanks a lot! >> >> Im still having a bit of trouble getting it working though. When >> skypiax_proxy starts, I get the following errors: >> >> 2009-09-05 15:39:40.631 skypiax_proxy[77842:10b] Failed to init >> theDOProxy >> 2009-09-05 15:39:42.139 skypiax_proxy[77842:10b] Failed to init >> theDOProxy >> >> Then, when an incoming call comes in, I see the messages: >> >> Message received CALL 683 STATUS RINGING >> Sent 23 bytes to FreeSWITCH >> Sending to skype: GET CALL 683 PARTNER_HANDLE >> Send to skype: GET CALL 683 PARTNER_HANDLE >> Message received CALL 683 CONF_ID 0 >> Sent 18 bytes to FreeSWITCH >> Message received CALL 683 FAILUREREASON 14 >> Sent 25 bytes to FreeSWITCH >> Message received CALL 683 STATUS MISSED >> Sent 22 bytes to FreeSWITCH >> Message received CALL 683 SEEN FALSE >> Sent 19 bytes to FreeSWITCH >> >> But the call is not actually picked up by the proxy. Any ideas? >> Im on Mac OS X 10.5, w/ freeswitch from SVN. Thanks! >> >> >> On Sat, Sep 5, 2009 at 4:41 AM, Giovanni Maruzzelli > > wrote: >> Seven, >> >> thanks a lot for your efforts. >> >> I will merge it in the next days, and I will take care that it will >> not breaks Windows or Linux. >> >> If I find problems I will wait for you having more time in the >> future. >> >> I send you my super best wishes for your personal things to go well >> and solves in the best of the possible ways. >> >> ciao for now, >> >> -giovanni >> >> >> >> Sincerely, >> >> Giovanni Maruzzelli >> Cell : +39-347-2665618 >> >> >> >> >> On Sat, Sep 5, 2009 at 1:13 PM, Seven Du wrote: >> > gm, >> > >> > Thanks a lot you can merge into the mainline. I check into my >> branch >> > because it's currently not as useful as on Linux and Windows and >> the >> > solution is not good. But it works and it is a good start that >> > mod_skypiax runs on OSX. Sure it would be easier for people want to >> > test and improve it if it been merged into trunk. I think you can >> make >> > a diff by >> > >> > svn diff -r 14472:14772 http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax >> > >> > FYI for personal reason I won't have much time put on this in the >> > coming month. Actually the code was done a few weeks ago but i only >> > got a chance to commit it yesterday. Sure that is not to say I >> cannot >> > do but fixes. But can you please make sure it won't break Linux/ >> > windows build when you merge the code? I haven't have a chance to >> test >> > all of them yet. >> > >> > -7- >> > >> > On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: >> >> Seeeeeeeven! >> >> >> >> I saw the modification you made on the wiki page... >> >> >> >> You made it, mod_skypiax runs on OSX!!!! >> >> >> >> Let's merge your mods on the mainline, pleaaaase ;-))) >> >> >> >> -giovanni >> >> >> >> >> >> >> >> >> >> Sincerely, >> >> >> >> Giovanni Maruzzelli >> >> Cell : +39-347-2665618 >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dujinfang at gmail.com Sun Sep 6 01:28:02 2009 From: dujinfang at gmail.com (Seven Du) Date: Sun, 6 Sep 2009 16:28:02 +0800 Subject: [Freeswitch-users] mod_skypiax for OSX????? In-Reply-To: <1ecdcb6a0909051625ud2f8dbbjf327e6e1ac8a21d1@mail.gmail.com> References: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> <1ecdcb6a0909051625ud2f8dbbjf327e6e1ac8a21d1@mail.gmail.com> Message-ID: <20A29ACA-25E5-401D-A213-B66D8942DD78@gmail.com> On Sep 6, 2009, at 7:25 AM, Tapan Parikh wrote: > > Yes, thanks a lot! > > Im still having a bit of trouble getting it working though. When > skypiax_proxy starts, I get the following errors: > > 2009-09-05 15:39:40.631 skypiax_proxy[77842:10b] Failed to init > theDOProxy > 2009-09-05 15:39:42.139 skypiax_proxy[77842:10b] Failed to init > theDOProxy > I don't have that problem. google found this: https://developer.skype.com/jira/browse/SPA-418 > Then, when an incoming call comes in, I see the messages: > Please try set Auto-Answer on Skype Preference and retry. > Message received CALL 683 STATUS RINGING > Sent 23 bytes to FreeSWITCH > Sending to skype: GET CALL 683 PARTNER_HANDLE > Send to skype: GET CALL 683 PARTNER_HANDLE > Message received CALL 683 CONF_ID 0 > Sent 18 bytes to FreeSWITCH > Message received CALL 683 FAILUREREASON 14 > Sent 25 bytes to FreeSWITCH > Message received CALL 683 STATUS MISSED > Sent 22 bytes to FreeSWITCH > Message received CALL 683 SEEN FALSE > Sent 19 bytes to FreeSWITCH > > But the call is not actually picked up by the proxy. Any ideas? Im > on Mac OS X 10.5, w/ freeswitch from SVN. Thanks! > > > On Sat, Sep 5, 2009 at 4:41 AM, Giovanni Maruzzelli > wrote: > Seven, > > thanks a lot for your efforts. > > I will merge it in the next days, and I will take care that it will > not breaks Windows or Linux. > > If I find problems I will wait for you having more time in the future. > > I send you my super best wishes for your personal things to go well > and solves in the best of the possible ways. > > ciao for now, > > -giovanni > > > > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > > > On Sat, Sep 5, 2009 at 1:13 PM, Seven Du wrote: > > gm, > > > > Thanks a lot you can merge into the mainline. I check into my branch > > because it's currently not as useful as on Linux and Windows and the > > solution is not good. But it works and it is a good start that > > mod_skypiax runs on OSX. Sure it would be easier for people want to > > test and improve it if it been merged into trunk. I think you can > make > > a diff by > > > > svn diff -r 14472:14772 http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax > > > > FYI for personal reason I won't have much time put on this in the > > coming month. Actually the code was done a few weeks ago but i only > > got a chance to commit it yesterday. Sure that is not to say I > cannot > > do but fixes. But can you please make sure it won't break Linux/ > > windows build when you merge the code? I haven't have a chance to > test > > all of them yet. > > > > -7- > > > > On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: > >> Seeeeeeeven! > >> > >> I saw the modification you made on the wiki page... > >> > >> You made it, mod_skypiax runs on OSX!!!! > >> > >> Let's merge your mods on the mainline, pleaaaase ;-))) > >> > >> -giovanni > >> > >> > >> > >> > >> Sincerely, > >> > >> Giovanni Maruzzelli > >> Cell : +39-347-2665618 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From juanbackson at gmail.com Sun Sep 6 03:32:01 2009 From: juanbackson at gmail.com (Juan Backson) Date: Sun, 6 Sep 2009 18:32:01 +0800 Subject: [Freeswitch-users] Chat redirect In-Reply-To: References: <27c25bc40909051126t619e34e0q85fa8b097a61240f@mail.gmail.com> Message-ID: <27c25bc40909060332i29a8139eh63adbbe98845a8c1@mail.gmail.com> Hi Brian, >From the event socket, there is no message received when a MESSAGE is sent from one sip user to another. If both users are registered, I can send message between them. But if the receiving party is not registered, I want to be able to store it. However, there is no way to intercept this MESSAGE. Is there anyway to solve this problem. thx, jb On Sun, Sep 6, 2009 at 2:36 AM, Brian West wrote: > Not automatically. But you could use the event socket to get the > message and forward it via ESL. > /b > > On Sep 5, 2009, at 1:26 PM, Juan Backson wrote: > > > > > If so, how can it be done? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090906/4d230bb4/attachment.html From n.geordzhev at gmail.com Sun Sep 6 03:40:19 2009 From: n.geordzhev at gmail.com (Nikolai Geordzhev) Date: Sun, 6 Sep 2009 13:40:19 +0300 Subject: [Freeswitch-users] Call Forwarding Question Message-ID: Hi, I`m trying to implement Call Forwarding in my FS setup. I set a user variable managing the type of forwarding (busy,no answer,unconditional) and the destination the phone is forwarded to: > > > > > > > Can somebody please explain how to do this? > > > > Thanks! > > > > Absolutely. Using session:execute is only for dialplan applications. > To do API commands, including all the stuff in mod_commands, you need > to create an API object: > > api = freeswitch.API() > > Then send a command and the variable will receive the results: > > reply = api:executeString("version") > > Check out this page for more examples: > http://wiki.freeswitch.org/wiki/Make_API_calls_directly_from_Lua_code > > -MC > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090906/b95fd6aa/attachment.html From gservat at gmail.com Sun Sep 6 06:24:47 2009 From: gservat at gmail.com (Gonzalo Servat) Date: Sun, 6 Sep 2009 23:24:47 +1000 Subject: [Freeswitch-users] Skypiax working but laggy Message-ID: Hi All, I'm just testing out mod_skypiax (great work Giovanni & co!) and while it's working and all, I find that when I call in from a Skype contact, it's /very/ laggy. I would say something on the skype end and I would hear it on the FS end quite a bit later. Funny thing is the audio going the other way has much faster response. I'm running FreeSWITCH Version 1.0.trunk (14772) on Ubuntu 9.04 Jaunty. Do you need more info? Thanks, Gonzalo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090906/9859a130/attachment.html From tomabroad at gmail.com Sun Sep 6 06:43:54 2009 From: tomabroad at gmail.com (tom) Date: Sun, 6 Sep 2009 09:43:54 -0400 Subject: [Freeswitch-users] freeswitch - q: originate calls from database Message-ID: <6f7c60c40909060643t24f46400ic40451e2796db861@mail.gmail.com> hi, is this scenario doable? let the system call people , they talk do an ivr, and can dependend on their selection end up in a real call-queue. 1) how would i tell FS to call xyz-people from the databse? thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090906/44733ab7/attachment.html From gmaruzz at celliax.org Sun Sep 6 07:07:43 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Sun, 6 Sep 2009 16:07:43 +0200 Subject: [Freeswitch-users] Skypiax working but laggy In-Reply-To: References: Message-ID: <7b197bef0909060707k64d91aa1lddbd4b60b7edeb8a@mail.gmail.com> Ubuntu 9.04 is explicitly discouraged, for heavy duty, if you like Ubuntu, use 8.04. That said, how is your call flow? I mean: Skypeclient->FS->SIP is laggy? How much? (1 sec, 10 sec, ...) SIP->FS->Skypeclient is not laggy? Or you mean that one side hear the other in real time, while the other side hear the other with a lag? Can you describe the problem with full informations? (what kind of protocols, clients, how the calls are originated, how are answered, etc etc etc etc etc :-) ) -giovanni Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 On Sun, Sep 6, 2009 at 3:24 PM, Gonzalo Servat wrote: > Hi All, > I'm just testing out mod_skypiax (great work Giovanni & co!) and while it's > working and all, I find that when I call in from a Skype contact, it's > /very/ laggy. I would say something on the skype end and I would hear it on > the FS end quite a bit later. Funny thing is the audio going the other way > has much faster response. > I'm running?FreeSWITCH Version 1.0.trunk (14772) on Ubuntu 9.04 Jaunty. Do > you need more info? > Thanks, > Gonzalo > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From ivan at myrvold.org Sun Sep 6 08:11:10 2009 From: ivan at myrvold.org (Ivan C Myrvold) Date: Sun, 6 Sep 2009 17:11:10 +0200 Subject: [Freeswitch-users] mod_skypiax for OSX????? In-Reply-To: References: <7b197bef0909050149n7354e6abva3061a8833b37a5e@mail.gmail.com> <06F4A075-A66F-40EA-8780-980425276F20@gmail.com> <7b197bef0909050441j7fd8fa74m986a8f0992251761@mail.gmail.com> <1ecdcb6a0909051625ud2f8dbbjf327e6e1ac8a21d1@mail.gmail.com> Message-ID: Yes, that worked! Now skypiax is working for me on OS X. This is great! Ivan Den 6. sep.. 2009 kl. 10:09 skrev Seven Du: > I'm not sure this and don't have time to debug. But last time I tried > incoming calls worked by setting skype to Auto-answer calls. Can you > try that? > > On Sep 6, 2009, at 3:19 PM, Ivan C Myrvold wrote: >> I have got outgoing call to Skype to work, and the audio quality is >> excellent. But I also have problems with incoming call, looks like >> it doesn't get bridged by FreeSWITCH. I put sofia on debug, but >> couldn't see any Sofia messages at all. Only message I saw in the >> console was this: >> >> 2009-09-05 20:44:39.720943 [WARNING] skypiax_protocol.c:375 rev >> 14772[0x0|37 ][WARNINGA 375 ][interface1][-1, 0, 0] >> skype_call: 332, STATUS: RINGINGCALL is not recognized >> >> But I am thrilled that Seven have got skypiax to work this far, and >> am confident that we will have it worked both incoming and outgoing >> soon. >> >> Ivan >> >> Den 6. sep.. 2009 kl. 01:25 skrev Tapan Parikh: >> >>> >>> Yes, thanks a lot! >>> >>> Im still having a bit of trouble getting it working though. When >>> skypiax_proxy starts, I get the following errors: >>> >>> 2009-09-05 15:39:40.631 skypiax_proxy[77842:10b] Failed to init >>> theDOProxy >>> 2009-09-05 15:39:42.139 skypiax_proxy[77842:10b] Failed to init >>> theDOProxy >>> >>> Then, when an incoming call comes in, I see the messages: >>> >>> Message received CALL 683 STATUS RINGING >>> Sent 23 bytes to FreeSWITCH >>> Sending to skype: GET CALL 683 PARTNER_HANDLE >>> Send to skype: GET CALL 683 PARTNER_HANDLE >>> Message received CALL 683 CONF_ID 0 >>> Sent 18 bytes to FreeSWITCH >>> Message received CALL 683 FAILUREREASON 14 >>> Sent 25 bytes to FreeSWITCH >>> Message received CALL 683 STATUS MISSED >>> Sent 22 bytes to FreeSWITCH >>> Message received CALL 683 SEEN FALSE >>> Sent 19 bytes to FreeSWITCH >>> >>> But the call is not actually picked up by the proxy. Any ideas? >>> Im on Mac OS X 10.5, w/ freeswitch from SVN. Thanks! >>> >>> >>> On Sat, Sep 5, 2009 at 4:41 AM, Giovanni Maruzzelli >>> wrote: >>> Seven, >>> >>> thanks a lot for your efforts. >>> >>> I will merge it in the next days, and I will take care that it will >>> not breaks Windows or Linux. >>> >>> If I find problems I will wait for you having more time in the >>> future. >>> >>> I send you my super best wishes for your personal things to go well >>> and solves in the best of the possible ways. >>> >>> ciao for now, >>> >>> -giovanni >>> >>> >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> Cell : +39-347-2665618 >>> >>> >>> >>> >>> On Sat, Sep 5, 2009 at 1:13 PM, Seven Du wrote: >>>> gm, >>>> >>>> Thanks a lot you can merge into the mainline. I check into my >>> branch >>>> because it's currently not as useful as on Linux and Windows and >>> the >>>> solution is not good. But it works and it is a good start that >>>> mod_skypiax runs on OSX. Sure it would be easier for people want to >>>> test and improve it if it been merged into trunk. I think you can >>> make >>>> a diff by >>>> >>>> svn diff -r 14472:14772 http://svn.freeswitch.org/svn/freeswitch/branches/seven/src/mod/endpoints/mod_skypiax >>>> >>>> FYI for personal reason I won't have much time put on this in the >>>> coming month. Actually the code was done a few weeks ago but i only >>>> got a chance to commit it yesterday. Sure that is not to say I >>> cannot >>>> do but fixes. But can you please make sure it won't break Linux/ >>>> windows build when you merge the code? I haven't have a chance to >>> test >>>> all of them yet. >>>> >>>> -7- >>>> >>>> On Sep 5, 2009, at 4:49 PM, Giovanni Maruzzelli wrote: >>>>> Seeeeeeeven! >>>>> >>>>> I saw the modification you made on the wiki page... >>>>> >>>>> You made it, mod_skypiax runs on OSX!!!! >>>>> >>>>> Let's merge your mods on the mainline, pleaaaase ;-))) >>>>> >>>>> -giovanni >>>>> >>>>> >>>>> >>>>> >>>>> Sincerely, >>>>> >>>>> Giovanni Maruzzelli >>>>> Cell : +39-347-2665618 >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From frank at impactfax.com Sun Sep 6 08:43:30 2009 From: frank at impactfax.com (Frank @ Impact) Date: Sun, 6 Sep 2009 11:43:30 -0400 Subject: [Freeswitch-users] Bind extention to a different Dialplan and cdr php? Message-ID: Is there a way to bind a particular extension to a different dialplan php and a different cdr php script than the default one? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090906/172945ab/attachment.html From n.geordzhev at gmail.com Sun Sep 6 11:22:13 2009 From: n.geordzhev at gmail.com (Nikolai Geordzhev) Date: Sun, 6 Sep 2009 21:22:13 +0300 Subject: [Freeswitch-users] Call Forwarding Question Message-ID: Hi, I`m trying to implement Call Forwarding in my FS setup. I set a user variable managing the type of forwarding (busy,no answer,unconditional) and the destination the phone is forwarded to: > > > References: <86a32abc0909061836j4a4d756bnf5f1e675e1607cd6@mail.gmail.com> Message-ID: <86a32abc0909061857u3bff1a3bkb16400f03f9b980c@mail.gmail.com> I also have plans to add a GUI later, maybe I will merge my code and turn it into a ramaze app, but it should be usable right now. Regards, Diego On Mon, Sep 7, 2009 at 1:36 AM, Diego Viola wrote: > Hello, > > I'm currently working on a calling card application written in Ruby, just a > hobby, I currently have it on a usable state and I thought I would post it > here in case if there is someone interested. > > It uses mod_nibblebill as the billing/rate engine and FSR (FreeSWITCHeR) as > the event socket library. > > Here is the url of the project: > > http://github.com/diego/freeswitch-card/tree/master > > You can find a clone of the project on my FreeSWITCH contrib directory too. > > http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ > > Regards, > > Diego > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/aaf37d2c/attachment.html From djbinter at yahoo.com Sun Sep 6 19:20:35 2009 From: djbinter at yahoo.com (DJB) Date: Sun, 6 Sep 2009 19:20:35 -0700 (PDT) Subject: [Freeswitch-users] Call Transfer Question Message-ID: <638470.52562.qm@web37501.mail.mud.yahoo.com> I am not sure if my question did not get post correctly earlier. I wonder whether anyone can give me any recommendations. Here is the call flow: I call from the PSTN (A party) into my Polycom phone (B-party) which is registered to FreeSwtich. The Freeswtich is setup not to route media as I have an SBC acting as a mirror proxy that will do all the NAT and media routing. The inbound call is setup fine and there is two way voice. I then blind transfer from the Polycom to my Cell phone. I see the polycom send a SIP refer to Freeswitch and it sends a 202 accepted fine and that leg between the Polycom (B party) and the A party is torn down fine like its supposed to be. The Freeswitch places the outbound call (the number the call is transferring to C-party) and that call completes. However now there is one way audio between the A party and C party . I see RTP streaming back from the egress carrier where the call was transfered to so the A party can hear the C party but the C party cannot hear the A party . When I look at the SIP traces of the original inbound call from the A-party I see a SIP re-invite from free switch to place the call on hold (contains Freeswitch RTP address to I can hear hold music) while it is transferring the call and the A-party does hear on hold music from Freeswitch while the call is being transferred. However I do not see a second re-invite from freeswitch to pass the media IP it got from the egress leg back to the original inbound leg. If my inbound gateway does not get a re-invite from Freeswitch to redirect its media to the new RTP address of of the egress carrier it will not do so hence the one way voice. How do I get the Freeswitch to re-invite the original ingress leg once it gets the SIP 183 from the egress with the new RTP info ? Free switch is sending the first SIP re-invite to my inbound gateway with new media IP (IP of itself) so the A-party can hear on hold music , but does not send a second re-invite to my inbound gateway after it receives the new RTP address from the egress carrier for the call that was transferred back out. Thank you. From ahmedmunir007 at gmail.com Sun Sep 6 21:58:14 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Mon, 7 Sep 2009 10:58:14 +0600 Subject: [Freeswitch-users] Checking Busy Status Message-ID: Hi, How can I check the busy status in FS? I've searched all the wiki pages i.e. dptools, dialplans, dialplanxml and even mod_perl portion as well, but couldn't find checking busy status. I've written a perl script but couldn't complete it because theres no any function or class regrding busy status. Kindly let me know how can I check the busy status in mod_perl and also in dialplan tools as well. -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/9795c554/attachment.html From mrene_lists at avgs.ca Sun Sep 6 22:01:58 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Sun, 6 Sep 2009 22:01:58 -0700 Subject: [Freeswitch-users] Checking Busy Status In-Reply-To: References: Message-ID: <120649F5-CC54-4CFB-9F0D-0FB199CBB073@avgs.ca> The hangup cause will be USER_BUSY. You can hop on #freeswitch if you need some explanations. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 6-Sep-09, at 9:58 PM, Ahmed Munir wrote: > Hi, > > How can I check the busy status in FS? I've searched all the wiki > pages i.e. dptools, dialplans, dialplanxml and even mod_perl portion > as well, but couldn't find checking busy status. > > I've written a perl script but couldn't complete it because theres > no any function or class regrding busy status. Kindly let me know > how can I check the busy status in mod_perl and also in dialplan > tools as well. > > > -- > Regards, > > Ahmed Munir > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ahmedmunir007 at gmail.com Sun Sep 6 22:47:33 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Mon, 7 Sep 2009 11:47:33 +0600 Subject: [Freeswitch-users] Checking Busy Status Message-ID: Hi, Thanks for quick reply. I want to know how can I apply USER_BUSY in perl? Like for hangup I'm calling it from function Freeswitch::CoreSession i.e. $session->hangup(); Do I have to call it as listed below; $session->USER_BUSY(); or there other way around in perl? Kindly do let me know. > ---------- Forwarded message ---------- > From: Mathieu Rene > To: freeswitch-users at lists.freeswitch.org > Date: Sun, 6 Sep 2009 22:01:58 -0700 > Subject: Re: [Freeswitch-users] Checking Busy Status > The hangup cause will be USER_BUSY. > > You can hop on #freeswitch if you need some explanations. > > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 6-Sep-09, at 9:58 PM, Ahmed Munir wrote: > > Hi, >> >> How can I check the busy status in FS? I've searched all the wiki pages >> i.e. dptools, dialplans, dialplanxml and even mod_perl portion as well, but >> couldn't find checking busy status. >> >> I've written a perl script but couldn't complete it because theres no any >> function or class regrding busy status. Kindly let me know how can I check >> the busy status in mod_perl and also in dialplan tools as well. >> >> >> -- >> Regards, >> >> Ahmed Munir >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/19ee577d/attachment.html From kawarod at laposte.net Sun Sep 6 23:06:24 2009 From: kawarod at laposte.net (rod) Date: Mon, 07 Sep 2009 10:06:24 +0400 Subject: [Freeswitch-users] Set disable-transcoding in dialplan In-Reply-To: <90823c940909040426s23c36c61vf4d14da939247ff@mail.gmail.com> References: <4A9BE5AB.4030304@laposte.net> <87f2f3b90908310938oa7e838dkd6e07579f224de37@mail.gmail.com> <4A9F6745.6030607@laposte.net> <7d0bfd8c0909031726t53f9900dldf7f2c0eab97305@mail.gmail.com> <4AA0ABA0.5020901@laposte.net> <90823c940909040426s23c36c61vf4d14da939247ff@mail.gmail.com> Message-ID: <4AA4A2E0.1010008@laposte.net> Hi Dmitry, thanks for your help, cause I've been able to set G729 when needed. What did the trick is the use of 'absolute_codec_string' defined using application set. I already tried to use this variable but using it like this: But I've never been successful using it this way. When I tried: Everything went fine. If others could check that absolute_codec_string doesn't work as expected when used with application bridge and that it's not related to my setup (I don't think so, but...), I'll open a jira ticket for the devs. regards, rod Dmitry Bely a ?crit : > I had a similar problem when I needed to talk to a gateway using g729 > while g711 was used by default. The following works for me: > > vars.xml > (...) > data="global_codec_prefs=PCMU,PCMA,G7221 at 32000h,G7221 at 16000h,G722,GSM,G729,G723"/> > > > sip_profiles/internal.xml > (...) > > > dialplan/default/01_example.com.xml > (...) > > data="{absolute_codec_string='G729'}sofia/gateway/${default_gateway}/$1"/> > > On Fri, Sep 4, 2009 at 9:54 AM, rod wrote: > >> Hi Nandy, >> >> yes already tried this, but if I use proxy_media=true, FS makes no >> control on the content of the RTP stream. But the pbm is that I need to >> use this: >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_start_dtmf_generate >> This function enables transcoding of SIP_INFO or RFC2833 to Inband DTMF >> in G711 >> >> But this feature doesn't work if I'm using proxy_media=true. >> >> In fact my setup is the following: >> >> CPE using G711A, G729 and SIP INFO for DTMF >> PEER_A using G729 only and RFC_2833 >> PEER_B using G711 and SIP INFO >> >> I have been able to make this works, with proxy_media=true for PEER_B >> cause I don't need transcoding of DTMF (SIP INFO to SIP INFO). >> For PEER_A, proxy_media is set to false (default) cause I need >> transcoding SIP INFO to RFC2833. I'm able to use G729 using >> codec_negotiation=greedy and setting G729 with highest priority on my >> internal profile. >> >> But the pbm is that I need to add PEER_C. >> PEER_C needs G711 with transcoding DTMF from SIP_INFO to Inband. >> >> And this is where I'm stuck, cause using "greedy settings and G729 with >> priority 1 in my codec list and proxy_media=false" force FS to negotiate >> G729 on leg A. But Leg B is willing to use G711 and FS is unable to >> transcode G729 <---> G711. >> >> I was wondering if there is a way for FS to force the codec order on Leg >> A with some knowledge of the preferred codec on Leg B, ie I know that >> Leg B will always use G711 so that I want to biase the SDP answer on Leg >> A based on this fact. >> >> regards, >> rod >> >> Nandy Dagondon a ?crit : >> >>> rod, >>> >>> have you tried this? >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2008-March/002199.html >>> >>> /nandy >>> >>> >>> On Thu, Sep 3, 2009 at 2:50 PM, rod >> > wrote: >>> >>> Hi Michael, >>> >>> I did some tests but I haven't been successful, so there is what I'm >>> trying to achieve: >>> >>> On A leg, my phone is using: PCMA and G729 (in this priority order) >>> >>> With PEER A, I want to use only G729 (thats is the only codec that >>> this >>> PEER support), so that the RTP flow will be: >>> Phone-----G729----FS-----G729-----PEER_A >>> >>> With PEER B, I want to use only G711, so: >>> Phone-----G711----FS-----G711-----PEER_B >>> >>> In fact, I'd like to force FS announcing the codec list priority based >>> on the priority of the codec announced by the PEER, cause FS is unable >>> to transcode G729 <--> G711. >>> >>> Tried a lot of things (greedy for codec-negociation, late_codec, >>> disable_transcoding, codec-prefs) without success. >>> >>> If you have some clue. >>> >>> regards, >>> rod >>> >>> Michael Collins a ?crit : >>> > Check out this page: >>> > http://wiki.freeswitch.org/wiki/Codec_negotiation >>> > >>> > Late negotiation will probably let you handle all the cases you >>> need. >>> > -MC >>> > >>> > On Mon, Aug 31, 2009 at 8:00 AM, rod >> >>> > >> wrote: >>> > >>> > Hi all, >>> > >>> > I'm wondering if I can do something like this: >>> > - in my internal profile, I have this because of some PEER >>> > using G729: >>> > - >>> > >>> > But for a specific PEER, I'd like to activate transcoding: >>> > - for this PEER, only G711 is used >>> > - I'd like to transcode DTMF SIP INFO or RFC2833 to >>> INBAND >>> > >>> > So in my dialplan, I tried before bridging: >>> > >>> > - >> data="disable-transcoding=false"/> >>> > - >>> > >>> > But I still see RFC2833 events between my FS and PEER and >>> the DTMF are >>> > not working. >>> > >>> > So 2 questions: >>> > - does application "start_dtmf_generate" requires transcoding >>> > - if yes, can I set the variable disable-transcoding in >>> my dialplan >>> > >>> > regards, >>> > rod >>> > > - Dmitry Bely > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > From jmesquita at freeswitch.org Sun Sep 6 23:14:06 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 7 Sep 2009 03:14:06 -0300 Subject: [Freeswitch-users] Checking Busy Status In-Reply-To: References: Message-ID: No expert in perl but you are looking for hangup_cause variable. Check how to get channel variables from a session in perl and you are set. jmesquita On Mon, Sep 7, 2009 at 2:47 AM, Ahmed Munir wrote: > > Hi, > > Thanks for quick reply. I want to know how can I apply USER_BUSY in perl? > Like for hangup I'm calling it from function Freeswitch::CoreSession i.e. > > $session->hangup(); > > Do I have to call it as listed below; > > $session->USER_BUSY(); > > or there other way around in perl? > > Kindly do let me know. > > > > >> ---------- Forwarded message ---------- >> From: Mathieu Rene >> To: freeswitch-users at lists.freeswitch.org >> Date: Sun, 6 Sep 2009 22:01:58 -0700 >> Subject: Re: [Freeswitch-users] Checking Busy Status >> The hangup cause will be USER_BUSY. >> >> You can hop on #freeswitch if you need some explanations. >> >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 6-Sep-09, at 9:58 PM, Ahmed Munir wrote: >> >> Hi, >>> >>> How can I check the busy status in FS? I've searched all the wiki pages >>> i.e. dptools, dialplans, dialplanxml and even mod_perl portion as well, but >>> couldn't find checking busy status. >>> >>> I've written a perl script but couldn't complete it because theres no any >>> function or class regrding busy status. Kindly let me know how can I check >>> the busy status in mod_perl and also in dialplan tools as well. >>> >>> >>> -- >>> Regards, >>> >>> Ahmed Munir >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Regards, > > Ahmed Munir > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/2d7687f1/attachment.html From n.geordzhev at gmail.com Sun Sep 6 23:51:46 2009 From: n.geordzhev at gmail.com (Nikolai Geordzhev) Date: Mon, 7 Sep 2009 09:51:46 +0300 Subject: [Freeswitch-users] Checking Busy Status In-Reply-To: References: Message-ID: I think you have to use: and after the bridge application you need to get the variable ${originate_disposition}. 2009/9/7 Jo?o Mesquita > No expert in perl but you are looking for hangup_cause variable. Check how > to get channel variables from a session in perl and you are set. > > jmesquita > > > On Mon, Sep 7, 2009 at 2:47 AM, Ahmed Munir wrote: > >> >> Hi, >> >> Thanks for quick reply. I want to know how can I apply USER_BUSY in perl? >> Like for hangup I'm calling it from function Freeswitch::CoreSession i.e. >> >> $session->hangup(); >> >> Do I have to call it as listed below; >> >> $session->USER_BUSY(); >> >> or there other way around in perl? >> >> Kindly do let me know. >> >> >> >> >>> ---------- Forwarded message ---------- >>> From: Mathieu Rene >>> To: freeswitch-users at lists.freeswitch.org >>> Date: Sun, 6 Sep 2009 22:01:58 -0700 >>> Subject: Re: [Freeswitch-users] Checking Busy Status >>> The hangup cause will be USER_BUSY. >>> >>> You can hop on #freeswitch if you need some explanations. >>> >>> Mathieu Rene >>> Avant-Garde Solutions Inc >>> Office: + 1 (514) 664-1044 x100 >>> Cell: +1 (514) 664-1044 x200 >>> mrene at avgs.ca >>> >>> >>> >>> >>> On 6-Sep-09, at 9:58 PM, Ahmed Munir wrote: >>> >>> Hi, >>>> >>>> How can I check the busy status in FS? I've searched all the wiki pages >>>> i.e. dptools, dialplans, dialplanxml and even mod_perl portion as well, but >>>> couldn't find checking busy status. >>>> >>>> I've written a perl script but couldn't complete it because theres no >>>> any function or class regrding busy status. Kindly let me know how can I >>>> check the busy status in mod_perl and also in dialplan tools as well. >>>> >>>> >>>> -- >>>> Regards, >>>> >>>> Ahmed Munir >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Regards, >> >> Ahmed Munir >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/8de10eed/attachment-0001.html From nagalenoj at gmail.com Mon Sep 7 00:47:04 2009 From: nagalenoj at gmail.com (Nagalenoj) Date: Mon, 7 Sep 2009 00:47:04 -0700 (PDT) Subject: [Freeswitch-users] ESL: DTMF event is not coming Message-ID: <25326328.post@talk.nabble.com> Dear friends, I am using freeswitch-1.0.4. When I execute the sample script(/libs/esl/perl/server2.pl), it is not receiving the DTMF events. When I execute the same program in freeswitch-1.0.3, it's receiving the event. Do I miss something to configure/upgrade.? -- View this message in context: http://www.nabble.com/ESL%3A-DTMF-event-is-not-coming-tp25326328p25326328.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From ahmedmunir007 at gmail.com Mon Sep 7 01:51:54 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Mon, 7 Sep 2009 14:51:54 +0600 Subject: [Freeswitch-users] Passing custom Dial Plans in perl Message-ID: Hi, I've set some variables in dialplan XML and I want to call these variables in perl, i.e. dialplan.xml Also after executing in perl the value of this variable pass to dialplan.xml Kindly tell me how to do this. -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/3a9d3682/attachment.html From leon at scarlet-internet.nl Mon Sep 7 01:54:37 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Mon, 7 Sep 2009 10:54:37 +0200 Subject: [Freeswitch-users] No dial-string available error In-Reply-To: <15b9404e0909052108j3842ef91ua13c54895a3959a5@mail.gmail.com> References: <27c25bc40909050136n4ea87c08xdc541874e979d4f7@mail.gmail.com> <15b9404e0909052108j3842ef91ua13c54895a3959a5@mail.gmail.com> Message-ID: Hi, Well, that's a coincidence.. I've been busy trying to figure out that same problem the entire weekend :-) FS needs to know where to lookup the registered user, by indeed setting a dialstring. The dialstring is described in the wiki page that's mentioned, but there's something more: It says on that page that you should set a param in your user like this: You can also test it in your cli like this: expand echo ${sofia_contact(username at domainname)} This will reply with: error/facility_not_subscribed sofia_contact will search your db in sip_registrations for all registered users. The actual arguments that can be given to the function are: profilename/username at domainname/arguments If you leave out the profilename, it will put your domainname there. The function then checks whether a *profile* exists with that name, and then lookup the entries in the db and then returns them. This can be fixed in two ways: either you specify the correct profilename (where the users are registered) with the sofia_contact function, or you alias the domainname to the profile where the user is registered in your sip profile. The alias can be set in your profile like this: or automatically like this: (where domainname can also be "all" to alias all domain names) But this presents another problem: The domain names that are aliased in the last way are retrieved from the database at startup of mod_sofia, and it is done through a directory lookup with purpose=gateways, even though the parse attribute - which is for getting gateways - is set to false !! (took me some time to figure that one out) So, I rewrote the mod_xml_odbc a bit so the set-channel-variables function does expansion on all attributes (from-name, from-value, to- name, to-value), and wrote the following template: This will simply return a list of all domains if no key_name and key_value were specified in the lookup, or list one domain if that is requested. The updated mod_xml_odbc and the directory-gateways.conf.xml template are in svn trunk contrib. This does work for me now but still some things feel a bit strange to me (perhaps someone can explain?): For examplle, right now, when I have two profiles where users can register with the same domain (for example one with a v4 and one with a v6 address), what should I do ? I can't alias the same domain to two profiles, so that means I have to call sofia_contact twice, once for each profile ? regards, Leon On Sep 6, 2009, at 6:08 AM, mayamatakeshi wrote: > > On Sat, Sep 5, 2009 at 5:36 PM, Juan Backson > wrote: > Hi, > > I am getting no dial-string available error when using xml_odbc > module to bridge a call. How can I resolve this problem? > > Hello, > I never tried the mod_xml_odbc. > But as the message says, you are not providing a dial-string. > I believe your template is incomplete. > Read about the dial-string here: > http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Dial_String > > > EXECUTE sofia/internal/180001 at 192.168.1.130 bridge(user/180001) > 2009-09-05 16:31:29.853456 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [key_value]=[192.168.1.134] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [key_name]=[name] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [tag_name]=[domain] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [section]=[directory] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Name]=[REQUEST_PARAMS] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Core- > UUID]=[f649cdfd-3715-4d18-94d9-417aa7e26873] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [FreeSWITCH-Hostname]=[server] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [FreeSWITCH-IPv4]=[192.168.1.134] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [FreeSWITCH-IPv6]=[::1] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Date-Local]=[2009-09-05 16:31:29] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Date-GMT]=[Sat, 05 Sep 2009 08:31:29 > GMT] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Date-Timestamp]=[1252139489853456] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Calling-File]=[mod_dptools.c] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Calling- > Function]=[user_outgoing_channel] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [Event-Calling-Line-Number]=[2365] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [as_channel]=[true] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [key]=[id] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [user]=[180001] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:401 DEBUG in > xml_odbc_search, header [domain]=[192.168.1.134] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:319 DEBUG GOING TO > RENDER TEMPLATE [default] > 2009-09-05 16:31:29.854400 [INFO] mod_xml_odbc.c:214 DEBUG > Performing Query: > SELECT enabled, sip_password, account_id FROM agent WHERE sip_user > = 180001 > 2009-09-05 16:31:29.855416 [INFO] mod_xml_odbc.c:214 DEBUG > Performing Query: > SELECT sip_password FROM agent WHERE sip_user = 180001 > 2009-09-05 16:31:29.855416 [INFO] mod_xml_odbc.c:417 Debug dump of > generated XML: > >
> > > > > > > > > > >
> > 2009-09-05 16:31:29.856442 [CONSOLE] mod_xml_odbc.c:457 Generated > XML is in [/tmp/16d2f01a-4332-43ff-9535-5615a192b40e.tmp.xml] > 2009-09-05 16:31:29.856442 [ERR] mod_dptools.c:2430 No dial-string > available, please check your user directory. > 2009-09-05 16:31:29.856442 [ERR] switch_ivr_originate.c:1527 Cannot > create outgoing channel of type [user] cause: [MANDATORY_IE_MISSING] > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/b69a1cc8/attachment.html From enno.egbert at googlemail.com Mon Sep 7 02:22:21 2009 From: enno.egbert at googlemail.com (NOx-WHV) Date: Mon, 7 Sep 2009 02:22:21 -0700 (PDT) Subject: [Freeswitch-users] select batchfile after call In-Reply-To: References: <25275633.post@talk.nabble.com> Message-ID: <25327475.post@talk.nabble.com> Thanks for help! I works, but now i have a new problem. The script works with the csv file from /FS/log/cdr-csv/XXX.csv. The problem is that the action application first starts and then the FS write the entry in the csv file. Does anybody have a tipp, how to call the script after writing the csv file? Thanks NOx mercutioviz wrote: > > > > Sent from my iPhone > > On Sep 3, 2009, at 6:08 AM, NOx-WHV wrote: > >> >> Hi, >> >> does anybody have a tip how to start a batchfile after hanging up. >> >> After ext. 1000 calls 1001 and hang up, i need a request to call: >> >> /../../FS/batchfile 1000 >> >> if 1001 calls 1000 i need: >> >> /../../FS/batchfile 1001 >> >> and so on... >> > Try something like this in your Dialplan: > > > > -MC >> >> Thanks for help >> -- >> View this message in context: >> http://www.nabble.com/select-batchfile-after-call-tp25275633p25275633.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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/select-batchfile-after-call-tp25275633p25327475.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From jason at jasonjgw.net Mon Sep 7 02:41:08 2009 From: jason at jasonjgw.net (Jason White) Date: Mon, 7 Sep 2009 19:41:08 +1000 Subject: [Freeswitch-users] No dial-string available error In-Reply-To: References: <27c25bc40909050136n4ea87c08xdc541874e979d4f7@mail.gmail.com> <15b9404e0909052108j3842ef91ua13c54895a3959a5@mail.gmail.com> Message-ID: <20090907094108.GA1768@jdc.jasonjgw.net> Leon de Rooij wrote: > For examplle, right now, when I have two profiles where users can > register with the same domain (for example one with a v4 and one > with a v6 address), what should I do ? I can't alias the same domain > to two profiles, so that means I have to call sofia_contact twice, > once for each profile ? The latter partly works, but as several colleagues and I discovered, doing this with the | syntax (i.e., two sofia_contact function calls separated by |) breaks the group call functionality of FreeSWITCH. There needs to be a solution that allows a user to register over v4 or v6 and to be contacted, without breaking group calls. From enno.egbert at googlemail.com Mon Sep 7 04:47:42 2009 From: enno.egbert at googlemail.com (NOx-WHV) Date: Mon, 7 Sep 2009 04:47:42 -0700 (PDT) Subject: [Freeswitch-users] select batchfile after call In-Reply-To: <25327475.post@talk.nabble.com> References: <25275633.post@talk.nabble.com> <25327475.post@talk.nabble.com> Message-ID: <25329259.post@talk.nabble.com> I just see, that i have a second problem. If I have a call and this call is without any response on the called side, the FS doesn?t call the script. NOx-WHV wrote: > > Thanks for help! > > I works, but now i have a new problem. The script works with the csv file > from /FS/log/cdr-csv/XXX.csv. > > The problem is that the action application first starts and then the FS > write the entry in the csv file. > > Does anybody have a tipp, how to call the script after writing the csv > file? > > Thanks > > NOx > > > > mercutioviz wrote: >> >> >> >> Sent from my iPhone >> >> On Sep 3, 2009, at 6:08 AM, NOx-WHV wrote: >> >>> >>> Hi, >>> >>> does anybody have a tip how to start a batchfile after hanging up. >>> >>> After ext. 1000 calls 1001 and hang up, i need a request to call: >>> >>> /../../FS/batchfile 1000 >>> >>> if 1001 calls 1000 i need: >>> >>> /../../FS/batchfile 1001 >>> >>> and so on... >>> >> Try something like this in your Dialplan: >> >> >> >> -MC >>> >>> Thanks for help >>> -- >>> View this message in context: >>> http://www.nabble.com/select-batchfile-after-call-tp25275633p25275633.html >>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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/select-batchfile-after-call-tp25275633p25329259.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From R.Kloosterman at mtel.nl Mon Sep 7 06:26:51 2009 From: R.Kloosterman at mtel.nl (Remko Kloosterman) Date: Mon, 7 Sep 2009 15:26:51 +0200 Subject: [Freeswitch-users] ESL C questions Message-ID: <11372C8B9E603F4FACDE6AB18256DEC601C8D8AD@srvmtel.office.mtel.nl> Hi there, I wonder, is ESL documentation available for C or does someone have something in draft? I'm trying to write an outbound socket application for some generic IVR features. I didn't find exactly that on the wiki except TODO J. The perl/ruby/javascript pages help a bit and the libs/esl source code provides examples that seem useful for trial and error, but I'd rather understand a bit more first. Right now I have a socket server that forks a process, answers a call, generates beep and plays voice. How can I retrieve digits? Place an outbound call and bridge both legs or retrieve a cause if the call failed? Send/receive SIP INFO? Disconnect the call with some cause code? And all that (and some more) in C. Any help or pointers is appreciated. Thanks, Remko -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/c8279a2f/attachment.html From mattdfong at gmail.com Mon Sep 7 08:26:25 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Mon, 7 Sep 2009 22:26:25 +0700 Subject: [Freeswitch-users] Recording Only 1 Leg of a Call Message-ID: <4256bf830909070826o5cb0d7a4lcd27e07ee350b50d@mail.gmail.com> Whats the best way to record only one leg of a call? uuid_record records both channels session_record does the same (but has a stereo option) is there any way to record only an a-leg of the call? Thanks so much. --matt http://www.hellohunter.com hosted dialer & voice broadcasting -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/8f933085/attachment.html From jmesquita at freeswitch.org Mon Sep 7 10:20:58 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 7 Sep 2009 14:20:58 -0300 Subject: [Freeswitch-users] ESL C questions In-Reply-To: <11372C8B9E603F4FACDE6AB18256DEC601C8D8AD@srvmtel.office.mtel.nl> References: <11372C8B9E603F4FACDE6AB18256DEC601C8D8AD@srvmtel.office.mtel.nl> Message-ID: Remko, I wrote the documentation that is on docs.freeswitch.org Take a look there, it is far from being complete but it might help. jmesquita On Mon, Sep 7, 2009 at 10:26 AM, Remko Kloosterman wrote: > Hi there, > > > > I wonder, is ESL documentation available for C or does someone have > something in draft? I?m trying to write an outbound socket application for > some generic IVR features. I didn?t find exactly that on the wiki except > TODO J. The perl/ruby/javascript pages help a bit and the libs/esl source > code provides examples that seem useful for trial and error, but I?d rather > understand a bit more first. > > > > Right now I have a socket server that forks a process, answers a call, > generates beep and plays voice. How can I retrieve digits? Place an outbound > call and bridge both legs or retrieve a cause if the call failed? > Send/receive SIP INFO? Disconnect the call with some cause code? And all > that (and some more) in C. Any help or pointers is appreciated. > > > > Thanks, > > Remko > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/17e26573/attachment.html From stephen.l.davies at gmail.com Sun Sep 6 23:47:44 2009 From: stephen.l.davies at gmail.com (Stephen Davies) Date: Mon, 7 Sep 2009 08:47:44 +0200 Subject: [Freeswitch-users] Call Forwarding Question In-Reply-To: References: Message-ID: On 9/6/09, Nikolai Geordzhev wrote: > Hi, > I`m trying to implement Call Forwarding in my FS setup. I set a user > variable managing the type of forwarding (busy,no answer,unconditional) and > the destination the phone is forwarded to: > > > > > Exactly what I was after - thank you! On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk wrote: > > try something like this > > EventConsumer con = new EventConsumer("all", ""); > Event ev = con.pop(0); > > see lua sample - > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer > > > Phillip Jones-2 wrote: > > > > Hi there, > > > > mod_managed exposes EventReceivedFunction such that: > > > > Session.EventReceivedFunction = (e) => > > { > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", e.ToString()); > > return ""; > > }; > > > > should trap all events to which i subscribe. > > > > > > But how do I subscribe to events? What is the .NET / managed equivalent > > of: > > > > switch_event_bind(const char *id, switch_event_types_t event, const char > > *subclass_name, switch_event_callback_t callback, void *user_data); > > > > > > > > Thank you! > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090906/ee307c76/attachment.html From testeador01 at gmail.com Mon Sep 7 12:12:19 2009 From: testeador01 at gmail.com (Milena) Date: Mon, 7 Sep 2009 14:12:19 -0500 Subject: [Freeswitch-users] Recording Only 1 Leg of a Call In-Reply-To: <4256bf830909070826o5cb0d7a4lcd27e07ee350b50d@mail.gmail.com> References: <4256bf830909070826o5cb0d7a4lcd27e07ee350b50d@mail.gmail.com> Message-ID: Hello, What about this?: " " the person would have to press *2 during the call to start the recording. 2009/9/7 Matthew Fong > Whats the best way to record only one leg of a call? > uuid_record records both channels > session_record does the same (but has a stereo option) > > is there any way to record only an a-leg of the call? Thanks so much. > > --matt > http://www.hellohunter.com > hosted dialer & voice broadcasting > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/9399c97a/attachment-0001.html From josh at radianttiger.com Mon Sep 7 12:49:34 2009 From: josh at radianttiger.com (Josh Rivers) Date: Mon, 7 Sep 2009 12:49:34 -0700 Subject: [Freeswitch-users] mod_managed ILoadNotificationPlugin Message-ID: What is the purpose if the ILoadNotificationPlugin? I thought it could be used to start off background code, but code run from that point seems to be terminated when the method returns. Does it only exist to check dependencies? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/60cad069/attachment.html From mitch.capper at gmail.com Mon Sep 7 13:32:19 2009 From: mitch.capper at gmail.com (Mitch Capper) Date: Mon, 7 Sep 2009 16:32:19 -0400 Subject: [Freeswitch-users] Passing custom Dial Plans in perl In-Reply-To: References: Message-ID: I did not fully understand what you are saying. If you are saying you are running that in XML and want to access the value in perl then just $session->getVariable should work. If you are saying you are setting it in PERL and want to access it in dialplan XML afterwards the variables are accessable as if you set them in XML just remember that ALL expressions are evaluated at the start of an XML Plan, so if you change a variable used in those expressions it will not be picked up. ~Mitch On Mon, Sep 7, 2009 at 4:51 AM, Ahmed Munir wrote: > Hi, > > I've set some variables in dialplan XML and I want to call these variables > in perl, i.e. > > dialplan.xml > > > > Also after executing in perl the value of this variable pass to > dialplan.xml > > Kindly tell me how to do this. > > -- > Regards, > > Ahmed Munir > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/62201874/attachment.html From msc at freeswitch.org Mon Sep 7 17:18:43 2009 From: msc at freeswitch.org (Michael S Collins) Date: Mon, 7 Sep 2009 17:18:43 -0700 Subject: [Freeswitch-users] select batchfile after call In-Reply-To: <25329259.post@talk.nabble.com> References: <25275633.post@talk.nabble.com> <25327475.post@talk.nabble.com> <25329259.post@talk.nabble.com> Message-ID: <843B1845-25AE-4A24-AC57-A9184523E485@freeswitch.org> I get the feeling that you are trying to use the wrong tool for the job. If you need to launch a script after the call ends AND you need access to the CSV file then you either should switch to XML CDR or just write a Perl script that runs as a daemon that sits and waits for CSV files to appear and process them accordingly. Don't use api_hang_hook when you need to post process calls and work with CDR data. -MC Sent from my iPhone On Sep 7, 2009, at 4:47 AM, NOx-WHV wrote: > > I just see, that i have a second problem. > > If I have a call and this call is without any response on the called > side, > the FS doesn?t call the script. > > > > > > NOx-WHV wrote: >> >> Thanks for help! >> >> I works, but now i have a new problem. The script works with the >> csv file >> from /FS/log/cdr-csv/XXX.csv. >> >> The problem is that the action application first starts and then >> the FS >> write the entry in the csv file. >> >> Does anybody have a tipp, how to call the script after writing the >> csv >> file? >> >> Thanks >> >> NOx >> >> >> >> mercutioviz wrote: >>> >>> >>> >>> Sent from my iPhone >>> >>> On Sep 3, 2009, at 6:08 AM, NOx-WHV >>> wrote: >>> >>>> >>>> Hi, >>>> >>>> does anybody have a tip how to start a batchfile after hanging up. >>>> >>>> After ext. 1000 calls 1001 and hang up, i need a request to call: >>>> >>>> /../../FS/batchfile 1000 >>>> >>>> if 1001 calls 1000 i need: >>>> >>>> /../../FS/batchfile 1001 >>>> >>>> and so on... >>>> >>> Try something like this in your Dialplan: >>> >>> >>> >>> -MC >>>> >>>> Thanks for help >>>> -- >>>> View this message in context: >>>> http://www.nabble.com/select-batchfile-after-call-tp25275633p25275633.html >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at 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/select-batchfile-after-call-tp25275633p25329259.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From diego.viola at gmail.com Mon Sep 7 19:15:49 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 8 Sep 2009 02:15:49 +0000 Subject: [Freeswitch-users] select batchfile after call In-Reply-To: <843B1845-25AE-4A24-AC57-A9184523E485@freeswitch.org> References: <25275633.post@talk.nabble.com> <25327475.post@talk.nabble.com> <25329259.post@talk.nabble.com> <843B1845-25AE-4A24-AC57-A9184523E485@freeswitch.org> Message-ID: <86a32abc0909071915n371d59ffh8dffc2abce50d874@mail.gmail.com> I have a script that will do just that, and it's pretty simple, it's written in ruby :). http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/callcard/cdr.rb Best regards, Diego On Tue, Sep 8, 2009 at 12:18 AM, Michael S Collins wrote: > I get the feeling that you are trying to use the wrong tool for the > job. If you need to launch a script after the call ends AND you need > access to the CSV file then you either should switch to XML CDR or > just write a Perl script that runs as a daemon that sits and waits for > CSV files to appear and process them accordingly. Don't use > api_hang_hook when you need to post process calls and work with CDR > data. > > -MC > > Sent from my iPhone > > On Sep 7, 2009, at 4:47 AM, NOx-WHV wrote: > > > > > I just see, that i have a second problem. > > > > If I have a call and this call is without any response on the called > > side, > > the FS doesn?t call the script. > > > > > > > > > > > > NOx-WHV wrote: > >> > >> Thanks for help! > >> > >> I works, but now i have a new problem. The script works with the > >> csv file > >> from /FS/log/cdr-csv/XXX.csv. > >> > >> The problem is that the action application first starts and then > >> the FS > >> write the entry in the csv file. > >> > >> Does anybody have a tipp, how to call the script after writing the > >> csv > >> file? > >> > >> Thanks > >> > >> NOx > >> > >> > >> > >> mercutioviz wrote: > >>> > >>> > >>> > >>> Sent from my iPhone > >>> > >>> On Sep 3, 2009, at 6:08 AM, NOx-WHV > >>> wrote: > >>> > >>>> > >>>> Hi, > >>>> > >>>> does anybody have a tip how to start a batchfile after hanging up. > >>>> > >>>> After ext. 1000 calls 1001 and hang up, i need a request to call: > >>>> > >>>> /../../FS/batchfile 1000 > >>>> > >>>> if 1001 calls 1000 i need: > >>>> > >>>> /../../FS/batchfile 1001 > >>>> > >>>> and so on... > >>>> > >>> Try something like this in your Dialplan: > >>> > >>> > >>> > >>> -MC > >>>> > >>>> Thanks for help > >>>> -- > >>>> View this message in context: > >>>> > http://www.nabble.com/select-batchfile-after-call-tp25275633p25275633.html > >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >>>> > >>>> > >>>> _______________________________________________ > >>>> FreeSWITCH-users mailing list > >>>> FreeSWITCH-users at 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 at 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/select-batchfile-after-call-tp25275633p25329259.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/33ec0ae0/attachment.html From ahmedmunir007 at gmail.com Mon Sep 7 21:14:02 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Tue, 8 Sep 2009 10:14:02 +0600 Subject: [Freeswitch-users] Checking Dial Status in FS Message-ID: Hi, In FS, which function can be used for checking the dial status (in channel variables as well as mod_perl function/class)? In asterisk $DIALSTATUS is used to check the status i.e. busy, answer, etc. Kindly do let me know. -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/7fc70c22/attachment.html From mattdfong at gmail.com Mon Sep 7 21:50:20 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Tue, 8 Sep 2009 11:50:20 +0700 Subject: [Freeswitch-users] Recording Only 1 Leg of a Call In-Reply-To: References: <4256bf830909070826o5cb0d7a4lcd27e07ee350b50d@mail.gmail.com> Message-ID: <4256bf830909072150i73ceca68we9476e2a57b53f6a@mail.gmail.com> I want to record without the telephone user's interaction. I think uuid_record should have the option to only record the audio of the uuid channel that is being specified, and as a secondary option combine the audio of the b leg (since uuid_record only specifies one uuid anyway--this seems logical). Anyway, just my wish list :) --matt http://www.hellohunter.com voice broadcasting & hosted dialer On Tue, Sep 8, 2009 at 2:12 AM, Milena wrote: > Hello, > What about this?: > " > > > " > > the person would have to press *2 during the call to start the recording. > > 2009/9/7 Matthew Fong > >> Whats the best way to record only one leg of a call? >> uuid_record records both channels >> session_record does the same (but has a stereo option) >> >> is there any way to record only an a-leg of the call? Thanks so much. >> >> --matt >> http://www.hellohunter.com >> hosted dialer & voice broadcasting >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/8a8a8f5f/attachment-0001.html From josh at radianttiger.com Mon Sep 7 22:41:22 2009 From: josh at radianttiger.com (Josh Rivers) Date: Mon, 7 Sep 2009 22:41:22 -0700 Subject: [Freeswitch-users] Using mod_managed to create full FreeSWITCH modules Message-ID: The wiki says: mod_managed exposes nearly the entire FreeSWITCH C API (courtesy of SWIG). This allows creation of not just API functions and call apps, but any type of module that FreeSWITCH supports (codecs, endpoints, etc.). The types are in the FreeSWITCH.Native namespace. FreeSWITCH.Native. The FreeSWITCH.Native.freeswitch type contains static members to access all the functions. Does anybody have a starting point they can share for a non-API/APP managed module. I'd like to build something that runs in the SWITCH_MODULE_SHUTDOWN_FUNCTION/SWITCH_MODULE_SHUTDOWN_FUNCTION/SWITCH_MODULE_SHUTDOWN_FUNCTION lifecycle. How can this be done? Thanks! Josh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/001d1d7e/attachment.html From raffaele.p.guidi at gmail.com Mon Sep 7 23:00:43 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 8 Sep 2009 08:00:43 +0200 Subject: [Freeswitch-users] Using mod_managed to create full FreeSWITCH modules In-Reply-To: References: Message-ID: I suppose you are working in windows with the binary distribution (that doesn't contain the mod_managed binaries, sources and examples. A bit weird, I think). You can get them from here http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/languages/mod_managed. The package contains a Demo.csx file featuring three examples: api, app and load notification plugin - very simple to understand and that will be easy to extend. Once mod_managed is installed you can put that file into the "managed" dir and will be automatically deployed. The csx (wich is csharp script) can also be compiled into an exe file and will work the same way. Regards, Raffaele On Tue, Sep 8, 2009 at 07:41, Josh Rivers wrote: > The wiki says: > mod_managed exposes nearly the entire FreeSWITCH C API (courtesy of SWIG). > This allows creation of not just API functions and call apps, but any type > of module that FreeSWITCH supports (codecs, endpoints, etc.). The types are > in the FreeSWITCH.Native namespace. FreeSWITCH.Native. The > FreeSWITCH.Native.freeswitch type contains static members to access all the > functions. > Does anybody have a starting point they can share for a non-API/APP managed > module. I'd like to build something that runs in > the SWITCH_MODULE_SHUTDOWN_FUNCTION/SWITCH_MODULE_SHUTDOWN_FUNCTION/SWITCH_MODULE_SHUTDOWN_FUNCTION > lifecycle. How can this be done? > > Thanks! > Josh > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/8efa0026/attachment.html From raffaele.p.guidi at gmail.com Mon Sep 7 23:05:12 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 8 Sep 2009 08:05:12 +0200 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: Message-ID: Yes! public class LoadDemo : ILoadNotificationPlugin { public bool Load() { Log.WriteLine(LogLevel.Notice, "LoadDemo running."); return true; } } this example is from Michael Giagnocavo's Demo.csx which you can find into the mod_managed svn. And let me add that works like a charm :) Ciao, Raffaele On Sun, Sep 6, 2009 at 22:50, Josh Rivers wrote: > Is there a way to start this when FreeSWITCH starts? The lua and perl > modules have a 'startup-script' configuration preference. Is there something > similar in mod_managed? Or is there a way to have an api command executed at > a startup? > > > Exactly what I was after - thank you! > > On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk wrote: > > > > > try something like this > > > > EventConsumer con = new EventConsumer("all", ""); > > Event ev = con.pop(0); > > > > see lua sample - > > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer > > > > > > Phillip Jones-2 wrote: > > > > > > Hi there, > > > > > > mod_managed exposes EventReceivedFunction such that: > > > > > > Session.EventReceivedFunction = (e) => > > > { > > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", > e.ToString()); > > > return ""; > > > }; > > > > > > should trap all events to which i subscribe. > > > > > > > > > But how do I subscribe to events? What is the .NET / managed equivalent > > > of: > > > > > > switch_event_bind(const char *id, switch_event_types_t event, const > char > > > *subclass_name, switch_event_callback_t callback, void *user_data); > > > > > > > > > > > > Thank you! > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/97cae8ec/attachment.html From josh at radianttiger.com Mon Sep 7 23:21:42 2009 From: josh at radianttiger.com (Josh Rivers) Date: Mon, 7 Sep 2009 23:21:42 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: Message-ID: Thanks for the response! I have tried putting a long-running loop here, but then it blocks anything else managed from happening: public class TestLoop : ILoadNotificationPlugin { public bool Load() { EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } } } However, if I fork off a thread here, freeswitch crashes: public class TestLoop : ILoadNotificationPlugin { public bool Load() { ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } }); return true; } } It doesn't look like this is a good place to start a long-running process? Thanks! Josh On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi < raffaele.p.guidi at gmail.com> wrote: > Yes! > public class LoadDemo : ILoadNotificationPlugin { > public bool Load() { > Log.WriteLine(LogLevel.Notice, "LoadDemo running."); > return true; > } > } > > this example is from Michael Giagnocavo's Demo.csx which you can find into > the mod_managed svn. > > And let me add that works like a charm :) > > Ciao, > Raffaele > > On Sun, Sep 6, 2009 at 22:50, Josh Rivers wrote: > >> Is there a way to start this when FreeSWITCH starts? The lua and perl >> modules have a 'startup-script' configuration preference. Is there something >> similar in mod_managed? Or is there a way to have an api command executed at >> a startup? >> >> >> Exactly what I was after - thank you! >> >> On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk wrote: >> >> > >> > try something like this >> > >> > EventConsumer con = new EventConsumer("all", ""); >> > Event ev = con.pop(0); >> > >> > see lua sample - >> > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer >> > >> > >> > Phillip Jones-2 wrote: >> > > >> > > Hi there, >> > > >> > > mod_managed exposes EventReceivedFunction such that: >> > > >> > > Session.EventReceivedFunction = (e) => >> > > { >> > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", >> e.ToString()); >> > > return ""; >> > > }; >> > > >> > > should trap all events to which i subscribe. >> > > >> > > >> > > But how do I subscribe to events? What is the .NET / managed >> equivalent >> > > of: >> > > >> > > switch_event_bind(const char *id, switch_event_types_t event, const >> char >> > > *subclass_name, switch_event_callback_t callback, void *user_data); >> > > >> > > >> > > >> > > Thank you! >> > > >> > > >> > > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/528cd3d1/attachment-0001.html From josh at radianttiger.com Mon Sep 7 23:23:48 2009 From: josh at radianttiger.com (Josh Rivers) Date: Mon, 7 Sep 2009 23:23:48 -0700 Subject: [Freeswitch-users] Using mod_managed to create full FreeSWITCH modules In-Reply-To: References: Message-ID: Thanks for the response Raffaele, I've successfully gotten the plugins to work. What I'm looking for is a way to write a lower-level FreeSWITCH module, as is implied by the wiki document. Mostly I'm trying to get something that runs stably in it's own thread from switch startup. Josh On Mon, Sep 7, 2009 at 11:00 PM, Raffaele P. Guidi < raffaele.p.guidi at gmail.com> wrote: > I suppose you are working in windows with the binary distribution (that > doesn't contain the mod_managed binaries, sources and examples. A bit weird, > I think). You can get them from here > http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/languages/mod_managed. > The package contains a Demo.csx file featuring three examples: api, app and > load notification plugin - very simple to understand and that will be easy > to extend. Once mod_managed is installed you can put that file into the > "managed" dir and will be automatically deployed. The csx (wich is csharp > script) can also be compiled into an exe file and will work the same way. > > Regards, > Raffaele > > On Tue, Sep 8, 2009 at 07:41, Josh Rivers wrote: > >> The wiki says: >> mod_managed exposes nearly the entire FreeSWITCH C API (courtesy of SWIG). >> This allows creation of not just API functions and call apps, but any type >> of module that FreeSWITCH supports (codecs, endpoints, etc.). The types are >> in the FreeSWITCH.Native namespace. FreeSWITCH.Native. The >> FreeSWITCH.Native.freeswitch type contains static members to access all the >> functions. >> Does anybody have a starting point they can share for a non-API/APP >> managed module. I'd like to build something that runs in >> the SWITCH_MODULE_SHUTDOWN_FUNCTION/SWITCH_MODULE_SHUTDOWN_FUNCTION/SWITCH_MODULE_SHUTDOWN_FUNCTION >> lifecycle. How can this be done? >> >> Thanks! >> Josh >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/aedfeb2b/attachment.html From yehavi.bourvine at gmail.com Mon Sep 7 23:44:41 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 8 Sep 2009 09:44:41 +0300 Subject: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec. Message-ID: Hello, I have a problem when trying to put a call on hold: I get the above message and the call is disconnected. Any idea where to look for the source of the problem? One thing I've tried is limiting all phones to use only one codec, but it doesn't help... Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/f1eda56a/attachment.html From msc at freeswitch.org Tue Sep 8 00:11:57 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Sep 2009 00:11:57 -0700 Subject: [Freeswitch-users] freeswitch - q: originate calls from database In-Reply-To: <6f7c60c40909060643t24f46400ic40451e2796db861@mail.gmail.com> References: <6f7c60c40909060643t24f46400ic40451e2796db861@mail.gmail.com> Message-ID: <87f2f3b90909080011g7c726fk1da8e0c053d90372@mail.gmail.com> On Sun, Sep 6, 2009 at 6:43 AM, tom wrote: > hi, > > is this scenario doable? let the system call people , they talk do an ivr, > and can dependend on their selection end up in a real call-queue. > 1) how would i tell FS to call xyz-people from the databse? > thx > > > You have some learning to do. Most likely you will need to build an application that can connect to FS via the event socket and issue "originate" commands. If you just need to "fire and forget" then it won't be too hard. You'll need to come up with a way to build dialstrings, i.e. "originate sofia/gateway/mygw/1234567890 XXXX" as well as a dialplan to handle the call flow. Then there's the matter of what to do with busy/no answer/invalid/hangup in mid-process/message machines/etc. Start by searching wiki.freeswitch.org for these concepts: originate command event socket (and ESL - the event socket layer) bridge application mod_fifo (for queue) Also, look at the demo IVR in the source tree because it has some handy examples of how to do things. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/dd4a73d7/attachment.html From jason at jasonjgw.net Tue Sep 8 00:12:40 2009 From: jason at jasonjgw.net (Jason White) Date: Tue, 8 Sep 2009 17:12:40 +1000 Subject: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec. In-Reply-To: References: Message-ID: <20090908071240.GA12470@jdc.jasonjgw.net> Yehavi Bourvine wrote: > > I have a problem when trying to put a call on hold: I get the above > message and the call is disconnected. Any idea where to look for the source > of the problem? My next step in your situation would be to obtain a Sip trace and post relevant details from it to the list. From msc at freeswitch.org Tue Sep 8 00:34:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Sep 2009 00:34:03 -0700 Subject: [Freeswitch-users] Passing custom Dial Plans in perl In-Reply-To: References: Message-ID: <87f2f3b90909080034i7c8d1a1ei57edef5ef744967@mail.gmail.com> On Mon, Sep 7, 2009 at 1:51 AM, Ahmed Munir wrote: > Hi, > > I've set some variables in dialplan XML and I want to call these variables > in perl, i.e. > > dialplan.xml > > > > Also after executing in perl the value of this variable pass to > dialplan.xml > > Kindly tell me how to do this. > If you execute the set app prior to calling your script then the variable will be available with $session->getVariable("DIALEXECUTED"); If you modify the value inside your script then it will be available in dialplan but only if you transfer the call back through the dialplan again. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/3fdef91a/attachment.html From msc at freeswitch.org Tue Sep 8 00:37:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Sep 2009 00:37:01 -0700 Subject: [Freeswitch-users] Bind extention to a different Dialplan and cdr php? In-Reply-To: References: Message-ID: <87f2f3b90909080037q7027bb1cme36aa68dcc42c882@mail.gmail.com> On Sun, Sep 6, 2009 at 8:43 AM, Frank @ Impact wrote: > Is there a way to bind a particular extension to a different dialplan phpand a different > cdr php script than the default one? > > Could you re-phrase this question with a bit more detail? Thanks. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/fd81e609/attachment.html From R.Kloosterman at mtel.nl Tue Sep 8 01:36:34 2009 From: R.Kloosterman at mtel.nl (Remko Kloosterman) Date: Tue, 8 Sep 2009 10:36:34 +0200 Subject: [Freeswitch-users] ESL C questions In-Reply-To: References: <11372C8B9E603F4FACDE6AB18256DEC601C8D8AD@srvmtel.office.mtel.nl> Message-ID: <11372C8B9E603F4FACDE6AB18256DEC601C8D905@srvmtel.office.mtel.nl> Thanks. Indeed it's a bit 'spartan' but it does describe the esl functions and structures. Remko Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Jo?o Mesquita Verzonden: maandag 7 september 2009 19:21 Aan: freeswitch-users at lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] ESL C questions Remko, I wrote the documentation that is on docs.freeswitch.org Take a look there, it is far from being complete but it might help. jmesquita On Mon, Sep 7, 2009 at 10:26 AM, Remko Kloosterman wrote: Hi there, I wonder, is ESL documentation available for C or does someone have something in draft? I'm trying to write an outbound socket application for some generic IVR features. I didn't find exactly that on the wiki except TODO J. The perl/ruby/javascript pages help a bit and the libs/esl source code provides examples that seem useful for trial and error, but I'd rather understand a bit more first. Right now I have a socket server that forks a process, answers a call, generates beep and plays voice. How can I retrieve digits? Place an outbound call and bridge both legs or retrieve a cause if the call failed? Send/receive SIP INFO? Disconnect the call with some cause code? And all that (and some more) in C. Any help or pointers is appreciated. Thanks, Remko _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/a8ffdf70/attachment-0001.html From enno.egbert at googlemail.com Tue Sep 8 03:10:11 2009 From: enno.egbert at googlemail.com (NOx-WHV) Date: Tue, 8 Sep 2009 03:10:11 -0700 (PDT) Subject: [Freeswitch-users] select batchfile after call In-Reply-To: <86a32abc0909071915n371d59ffh8dffc2abce50d874@mail.gmail.com> References: <25275633.post@talk.nabble.com> <25327475.post@talk.nabble.com> <25329259.post@talk.nabble.com> <843B1845-25AE-4A24-AC57-A9184523E485@freeswitch.org> <86a32abc0909071915n371d59ffh8dffc2abce50d874@mail.gmail.com> Message-ID: <25343243.post@talk.nabble.com> Hi, the script looks good. But I never use scripts directly in FS. Can you help, where and how I have to do what?!?!? My way with a linux script works. But only one option doesn?t work. If I have a call to a outside phone throuth a sip-gateway and I hang up (not the party who is called hang up) the channel is closed without any reaction. If the called party hang up, the script works. How can I call the script on NORMAL_CLEARING, session ended or CS_DESTROY? Thanks Diego Viola wrote: > > I have a script that will do just that, and it's pretty simple, it's > written > in ruby :). > > http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/callcard/cdr.rb > > Best regards, > > Diego > > On Tue, Sep 8, 2009 at 12:18 AM, Michael S Collins > wrote: > >> I get the feeling that you are trying to use the wrong tool for the >> job. If you need to launch a script after the call ends AND you need >> access to the CSV file then you either should switch to XML CDR or >> just write a Perl script that runs as a daemon that sits and waits for >> CSV files to appear and process them accordingly. Don't use >> api_hang_hook when you need to post process calls and work with CDR >> data. >> >> -MC >> >> Sent from my iPhone >> >> On Sep 7, 2009, at 4:47 AM, NOx-WHV wrote: >> >> > >> > I just see, that i have a second problem. >> > >> > If I have a call and this call is without any response on the called >> > side, >> > the FS doesn?t call the script. >> > >> > >> > >> > >> > >> > NOx-WHV wrote: >> >> >> >> Thanks for help! >> >> >> >> I works, but now i have a new problem. The script works with the >> >> csv file >> >> from /FS/log/cdr-csv/XXX.csv. >> >> >> >> The problem is that the action application first starts and then >> >> the FS >> >> write the entry in the csv file. >> >> >> >> Does anybody have a tipp, how to call the script after writing the >> >> csv >> >> file? >> >> >> >> Thanks >> >> >> >> NOx >> >> >> >> >> >> >> >> mercutioviz wrote: >> >>> >> >>> >> >>> >> >>> Sent from my iPhone >> >>> >> >>> On Sep 3, 2009, at 6:08 AM, NOx-WHV >> >>> wrote: >> >>> >> >>>> >> >>>> Hi, >> >>>> >> >>>> does anybody have a tip how to start a batchfile after hanging up. >> >>>> >> >>>> After ext. 1000 calls 1001 and hang up, i need a request to call: >> >>>> >> >>>> /../../FS/batchfile 1000 >> >>>> >> >>>> if 1001 calls 1000 i need: >> >>>> >> >>>> /../../FS/batchfile 1001 >> >>>> >> >>>> and so on... >> >>>> >> >>> Try something like this in your Dialplan: >> >>> >> >>> >> >>> >> >>> -MC >> >>>> >> >>>> Thanks for help >> >>>> -- >> >>>> View this message in context: >> >>>> >> http://www.nabble.com/select-batchfile-after-call-tp25275633p25275633.html >> >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >>>> >> >>>> >> >>>> _______________________________________________ >> >>>> FreeSWITCH-users mailing list >> >>>> FreeSWITCH-users at 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 at 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/select-batchfile-after-call-tp25275633p25329259.html >> > Sent from the Freeswitch-users mailing list archive at Nabble.com. >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at 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 at 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/select-batchfile-after-call-tp25275633p25343243.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mgg at giagnocavo.net Tue Sep 8 05:04:35 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 8 Sep 2009 08:04:35 -0400 Subject: [Freeswitch-users] mod_managed ILoadNotificationPlugin In-Reply-To: References: Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B38@mse17be1.mse17.exchange.ms> It's to notify your plugin that it's being loaded, and allow your plugin to opt out of being loaded. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Monday, September 07, 2009 1:50 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_managed ILoadNotificationPlugin What is the purpose if the ILoadNotificationPlugin? I thought it could be used to start off background code, but code run from that point seems to be terminated when the method returns. Does it only exist to check dependencies? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/a3230aec/attachment-0001.html From mgg at giagnocavo.net Tue Sep 8 05:05:17 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 8 Sep 2009 08:05:17 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> Hi, Can you please elaborate on the crash you receive when you queue a thread during load? Thanks, Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Tuesday, September 08, 2009 12:22 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET Thanks for the response! I have tried putting a long-running loop here, but then it blocks anything else managed from happening: public class TestLoop : ILoadNotificationPlugin { public bool Load() { EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } } } However, if I fork off a thread here, freeswitch crashes: public class TestLoop : ILoadNotificationPlugin { public bool Load() { ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } }); return true; } } It doesn't look like this is a good place to start a long-running process? Thanks! Josh On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi > wrote: Yes! public class LoadDemo : ILoadNotificationPlugin { public bool Load() { Log.WriteLine(LogLevel.Notice, "LoadDemo running."); return true; } } this example is from Michael Giagnocavo's Demo.csx which you can find into the mod_managed svn. And let me add that works like a charm :) Ciao, Raffaele On Sun, Sep 6, 2009 at 22:50, Josh Rivers > wrote: Is there a way to start this when FreeSWITCH starts? The lua and perl modules have a 'startup-script' configuration preference. Is there something similar in mod_managed? Or is there a way to have an api command executed at a startup? Exactly what I was after - thank you! On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk > wrote: > > try something like this > > EventConsumer con = new EventConsumer("all", ""); > Event ev = con.pop(0); > > see lua sample - > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer > > > Phillip Jones-2 wrote: > > > > Hi there, > > > > mod_managed exposes EventReceivedFunction such that: > > > > Session.EventReceivedFunction = (e) => > > { > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", e.ToString()); > > return ""; > > }; > > > > should trap all events to which i subscribe. > > > > > > But how do I subscribe to events? What is the .NET / managed equivalent > > of: > > > > switch_event_bind(const char *id, switch_event_types_t event, const char > > *subclass_name, switch_event_callback_t callback, void *user_data); > > > > > > > > Thank you! > > > > > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/9301cee3/attachment-0001.html From chris at cloudtel.com Tue Sep 8 05:14:07 2009 From: chris at cloudtel.com (Chris Burns) Date: Tue, 8 Sep 2009 08:14:07 -0400 Subject: [Freeswitch-users] select batchfile after call In-Reply-To: <25343243.post@talk.nabble.com> References: <25275633.post@talk.nabble.com> <86a32abc0909071915n371d59ffh8dffc2abce50d874@mail.gmail.com> <25343243.post@talk.nabble.com> Message-ID: <200909080814.07436.chris@cloudtel.com> On September 8, 2009 06:10:11 am NOx-WHV wrote: > Hi, > > the script looks good. But I never use scripts directly in FS. > > Can you help, where and how I have to do what?!?!? Use ESL (http://wiki.freeswitch.org/wiki/Event_Socket_Library) to subscribe to hangup events, as Diego suggested. OR Use XML CDR (http://wiki.freeswitch.org/wiki/Mod_xml_cdr) to receive an HTTP POST of CDR data when any call completes, as Michael suggested. > > My way with a linux script works. But only one option doesn?t work. If I > have a call to a outside phone throuth a sip-gateway and I hang up (not the > party who is called hang up) the channel is closed without any reaction. If > the called party hang up, the script works. > How can I call the script on NORMAL_CLEARING, session ended or CS_DESTROY? You can't do what you want with the hangup hook. Listen to the suggestions being given and take a new approach to your problem. > > Thanks > > Diego Viola wrote: > > I have a script that will do just that, and it's pretty simple, it's > > written > > in ruby :). > > > > http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/c > >allcard/cdr.rb > > > > Best regards, > > > > Diego > > > > On Tue, Sep 8, 2009 at 12:18 AM, Michael S Collins > > > > wrote: > >> I get the feeling that you are trying to use the wrong tool for the > >> job. If you need to launch a script after the call ends AND you need > >> access to the CSV file then you either should switch to XML CDR or > >> just write a Perl script that runs as a daemon that sits and waits for > >> CSV files to appear and process them accordingly. Don't use > >> api_hang_hook when you need to post process calls and work with CDR > >> data. > >> > >> -MC > >> > >> Sent from my iPhone > >> > >> On Sep 7, 2009, at 4:47 AM, NOx-WHV wrote: > >> > I just see, that i have a second problem. > >> > > >> > If I have a call and this call is without any response on the called > >> > side, > >> > the FS doesn?t call the script. > >> > > >> > NOx-WHV wrote: > >> >> Thanks for help! > >> >> > >> >> I works, but now i have a new problem. The script works with the > >> >> csv file > >> >> from /FS/log/cdr-csv/XXX.csv. > >> >> > >> >> The problem is that the action application first starts and then > >> >> the FS > >> >> write the entry in the csv file. > >> >> > >> >> Does anybody have a tipp, how to call the script after writing the > >> >> csv > >> >> file? > >> >> > >> >> Thanks > >> >> > >> >> NOx > >> >> > >> >> mercutioviz wrote: > >> >>> Sent from my iPhone > >> >>> > >> >>> On Sep 3, 2009, at 6:08 AM, NOx-WHV > >> >>> > >> >>> wrote: > >> >>>> Hi, > >> >>>> > >> >>>> does anybody have a tip how to start a batchfile after hanging up. > >> >>>> > >> >>>> After ext. 1000 calls 1001 and hang up, i need a request to call: > >> >>>> > >> >>>> /../../FS/batchfile 1000 > >> >>>> > >> >>>> if 1001 calls 1000 i need: > >> >>>> > >> >>>> /../../FS/batchfile 1001 > >> >>>> > >> >>>> and so on... > >> >>> > >> >>> Try something like this in your Dialplan: > >> >>> > >> >>> > >> >>> > >> >>> -MC > >> >>> > >> >>>> Thanks for help > >> >>>> -- > >> >>>> View this message in context: > >> > >> http://www.nabble.com/select-batchfile-after-call-tp25275633p25275633.ht > >>ml > >> > >> >>>> Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> >>>> > >> >>>> > >> >>>> _______________________________________________ > >> >>>> FreeSWITCH-users mailing list > >> >>>> FreeSWITCH-users at 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 at 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/select-batchfile-after-call-tp25275633p25329259.ht > >>ml > >> > >> > Sent from the Freeswitch-users mailing list archive at Nabble.com. > >> > > >> > > >> > _______________________________________________ > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org From mgg at giagnocavo.net Tue Sep 8 05:12:31 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 8 Sep 2009 08:12:31 -0400 Subject: [Freeswitch-users] Using mod_managed to create full FreeSWITCH modules In-Reply-To: References: Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B39@mse17be1.mse17.exchange.ms> Are you looking to run when mod_managed shuts down? Or when your managed plugin reloads, or something else? (mod_managed is not unloadable, so I don't believe it gets any notification of shutting down.) As far as interop in general, it's usually possible. However, a lot of the FreeSWITCH code uses macros, and they aren't available via SWIG. So in those cases, you'll either need to manually reconstruct the macro, or write some interop code in C/C++ to do what you want, then expose that via SWIG (or, if you do it nicely, just P/Invoke it directly). Some of the native code generates some pretty ugly structures; you will probably need to become friends with the Marshal class and pass around a lot of IntPtrs to get things going. As far as I know, no one has built a non API/App with mod_managed. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Monday, September 07, 2009 11:41 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Using mod_managed to create full FreeSWITCH modules The wiki says: mod_managed exposes nearly the entire FreeSWITCH C API (courtesy of SWIG). This allows creation of not just API functions and call apps, but any type of module that FreeSWITCH supports (codecs, endpoints, etc.). The types are in the FreeSWITCH.Native namespace. FreeSWITCH.Native. The FreeSWITCH.Native.freeswitch type contains static members to access all the functions. Does anybody have a starting point they can share for a non-API/APP managed module. I'd like to build something that runs in the SWITCH_MODULE_SHUTDOWN_FUNCTION/SWITCH_MODULE_SHUTDOWN_FUNCTION/SWITCH_MODULE_SHUTDOWN_FUNCTION lifecycle. How can this be done? Thanks! Josh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/c20de8c0/attachment.html From anatoliy at kounitskiy.com Tue Sep 8 05:20:32 2009 From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy) Date: Tue, 8 Sep 2009 15:20:32 +0300 Subject: [Freeswitch-users] Usefull option for mod_cdr_csv Message-ID: <1cd828b60909080520o5b592a6bl52a5052986429bf9@mail.gmail.com> There is another useful option for the module Mod_cdr_csv, that is not described in the default configuration file: If you use the default configuration file and you're using the variable accountcode, probably you've seen for every accountcode you have separate cdr file for it. But if you need only one file with all cdr - use the option above. Possible values are "true" or "false". Cheers, -- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy at kounitskiy.com Mobile: +359898913540 From tomabroad at gmail.com Tue Sep 8 06:11:27 2009 From: tomabroad at gmail.com (tom) Date: Tue, 8 Sep 2009 09:11:27 -0400 Subject: [Freeswitch-users] Mod xml cdr Message-ID: <6f7c60c40909080611g75303bdi8833e19f17ad9b2@mail.gmail.com> hi, reading the wiki tells me i need that module to make cdr happen. please correct me if i outline wrong steps here: - enable module - make install - create file called xml_cdr.conf.xml - based on the parameters make a http post does anyone have a xml file + a php file to make the appropriate mysql-isnerts? thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/75a4c635/attachment.html From anatoliy at kounitskiy.com Tue Sep 8 06:22:04 2009 From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy) Date: Tue, 8 Sep 2009 16:22:04 +0300 Subject: [Freeswitch-users] Mod xml cdr In-Reply-To: <6f7c60c40909080611g75303bdi8833e19f17ad9b2@mail.gmail.com> References: <6f7c60c40909080611g75303bdi8833e19f17ad9b2@mail.gmail.com> Message-ID: <1cd828b60909080622q60b21b4dl42603033adb3d8e7@mail.gmail.com> There are several modules for writing cdr information to file ( http://wiki.freeswitch.org/wiki/Cdr ): - mod_xml_cdr - to write the cdrs to xml files ( http://wiki.freeswitch.org/wiki/Mod_xml_cdr ) - mod_cdr_csv - to write the cdrs to csv files ( http://wiki.freeswitch.org/wiki/Mod_cdr_csv ), also in the end of the page wiki there is sample perl script for inserting in mysql db On Tue, Sep 8, 2009 at 4:11 PM, tom wrote: > hi, > > reading the wiki tells me i need that module to make cdr happen. please > correct me if i outline wrong steps here: > - enable module > - make install > - create file called xml_cdr.conf.xml > - based on the parameters make a http post > > does anyone have a xml file + a php file to make the appropriate > mysql-isnerts? > > thx > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anatoliy Kounitskiy ------------------------- E-mail: anatoliy at kounitskiy.com Mobile: +359898913540 From tina at a2unlimited.com Tue Sep 8 06:58:50 2009 From: tina at a2unlimited.com (Tina Martinez) Date: Tue, 08 Sep 2009 09:58:50 -0400 Subject: [Freeswitch-users] Custom Variables Message-ID: <58740.1252418330@a2unlimited.com> Hello, I have created custom variables that I pass to my FreeSWITCH dialplan. In my code I am monitoring the events and using the variables (now using ESL, very nice). The custom variables appear in events such as CHANNEL_ORIGINATE, but I do not see the variable in the CUSTOM event. For example, in CHANNEL_ORIGINATE, I see the custom variables as: variable_sip_h_X-custom_variable1 : ABC variable_sip_h_X-custom_variable2 : XYZ Is there a way for me to see the variables in the CUSTOM event as well? In my dialplan, I do have action application="verbose_events" data="true". But it does not seem to help. Any thoughts? - T From hjqlopez at hotmail.com Tue Sep 8 07:00:18 2009 From: hjqlopez at hotmail.com (Humberto Quintana) Date: Tue, 8 Sep 2009 10:00:18 -0400 Subject: [Freeswitch-users] 482 Request merged, in serial forking Message-ID: Hi Brian, Yes , the Call-Id is the same for the 2nd and 3rd transaction but the branch parameter in the Via header is different. Please check the capture below. Thanks, Humberto ----------> Route 1 U 2009/09/08 09:17:38.759129 kamailio:5060 -> freeswitch:5060 INVITE sip:5145555555 at gw1:5060 SIP/2.0. Record-Route: . Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.0. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. From: "hq160" <10092020 at kamailio>;tag=8c90b1379825fa62. To: . Contact: . Supported: replaces, timer, path. Call-ID: 04d14ab631843dc1 at 192.168.2.13. CSeq: 31348 INVITE. Max-Forwards: 68. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 396. . v=0. o=10092020 8000 8001 IN IP4 192.168.2.13. s=SIP Call. c=IN IP4 MediaServer. t=0 0. m=audio 50362 RTP/AVP 0 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. m=video 50366 RTP/AVP 99. a=sendrecv. a=rtpmap:99 H264/90000. a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0KAFJWgUH5A,KM4CfIC=. a=framerate:15. U 2009/09/08 09:17:38.861646 freeswitch:5060 -> kamailio:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.0. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. Record-Route: . From: "hq160" <10092020 at kamailio>;tag=8c90b1379825fa62. To: . Call-ID: 04d14ab631843dc1 at 192.168.2.13. CSeq: 31348 INVITE. UserIP-Agent: FreeSWITCH-mod_sofia/1.0.4-exported. Content-Length: 0. . U 2009/09/08 09:17:39.000958 freeswitch:5060 -> gw1:5060 INVITE sip:5145555555 at gw1:5060 SIP/2.0. Via: SIP/2.0/UDP freeswitch;rport;branch=z9hG4bK151FSXQmjX4KH. Max-Forwards: 67. From: "hq160" ;tag=3mrtKm2rma0De. To: . Call-ID: d2bda062-171c-122d-c787-005056aa5fb7. CSeq: 120089593 INVITE. Contact: . UserIP-Agent: FreeSWITCH-mod_sofia/1.0.4-exported. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 372. X-rsbc: 04d14ab631843dc1 at 192.168.2.13. Remote-Party-ID: "hq160" ;party=calling;screen=yes;privacy=off. . v=0. o=10092020 8000 8001 IN IP4 192.168.2.13. s=SIP Call. c=IN IP4 MediaServer. t=0 0. m=audio 50362 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. a=ptime:20. m=video 50366 RTP/AVP 99. a=rtpmap:99 H264/90000. a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0KAFJWgUH5A,KM4CfIC=. a=framerate:15. U 2009/09/08 09:17:39.056082 gw1:5060 -> freeswitch:5060 SIP/2.0 100 Trying. Call-ID: d2bda062-171c-122d-c787-005056aa5fb7. Content-Length: 0. CSeq: 120089593 INVITE. From: "hq160";tag=3mrtKm2rma0De. To: ;tag=d05714dc-26334. UserIP-Agent: Quintum/1.0.0 SN/0030E100A224 SW/P106-12-00. Via: SIP/2.0/UDP freeswitch;rport;branch=z9hG4bK151FSXQmjX4KH. Quintum: 0b06343032343333. . U 2009/09/08 09:17:39.058998 gw1:5060 -> freeswitch:5060 SIP/2.0 503 Service Unavailable. Call-ID: d2bda062-171c-122d-c787-005056aa5fb7. Content-Length: 0. CSeq: 120089593 INVITE. From: "hq160";tag=3mrtKm2rma0De. To: ;tag=d05714dc-26334. UserIP-Agent: Quintum/1.0.0 SN/0030E100A224 SW/P106-12-00. Via: SIP/2.0/UDP freeswitch;rport;branch=z9hG4bK151FSXQmjX4KH. . U 2009/09/08 09:17:39.059341 freeswitch:5060 -> gw1:5060 ACK sip:5145555555 at gw1:5060 SIP/2.0. Via: SIP/2.0/UDP freeswitch;rport;branch=z9hG4bK151FSXQmjX4KH. Max-Forwards: 67. From: "hq160" ;tag=3mrtKm2rma0De. To: ;tag=d05714dc-26334. Call-ID: d2bda062-171c-122d-c787-005056aa5fb7. CSeq: 120089593 ACK. Content-Length: 0. . U 2009/09/08 09:17:39.061148 freeswitch:5060 -> kamailio:5060 SIP/2.0 503 Service Unavailable. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.0. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. From: "hq160" <10092020 at kamailio>;tag=8c90b1379825fa62. To: ;tag=2BZ1HSHNQ19tj. Call-ID: 04d14ab631843dc1 at 192.168.2.13. CSeq: 31348 INVITE. UserIP-Agent: FreeSWITCH-mod_sofia/1.0.4-exported. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Reason: Q.850;cause=41;text="NORMAL_TEMPORARY_FAILURE". Content-Length: 0. . U 2009/09/08 09:17:39.062085 kamailio:5060 -> freeswitch:5060 ACK sip:5145555555 at gw1:5060 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.0. From: "hq160" <10092020 at kamailio>;tag=8c90b1379825fa62. Call-ID: 04d14ab631843dc1 at 192.168.2.13. To: ;tag=2BZ1HSHNQ19tj. CSeq: 31348 ACK. Max-Forwards: 70. UserIP-Agent: Kamailio (1.4.4-tls (i386/linux)). Content-Length: 0. . -------> Route 2 U 2009/09/08 09:17:41.426541 kamailio:5060 -> freeswitch:5060 INVITE sip:15145555555 at gw2:5061 SIP/2.0. Record-Route: . Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.1. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. From: "hq160" <10092020 at kamailio>;tag=8c90b1379825fa62. To: . Contact: . Supported: replaces, timer, path. Call-ID: 04d14ab631843dc1 at 192.168.2.13. CSeq: 31348 INVITE. Max-Forwards: 68. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 396. . v=0. o=10092020 8000 8001 IN IP4 192.168.2.13. s=SIP Call. c=IN IP4 MediaServer. t=0 0. m=audio 50362 RTP/AVP 0 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. m=video 50366 RTP/AVP 99. a=sendrecv. a=rtpmap:99 H264/90000. a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0KAFJWgUH5A,KM4CfIC=. a=framerate:15. U 2009/09/08 09:17:41.427280 freeswitch:5060 -> kamailio:5060 SIP/2.0 482 Request merged. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.1. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. From: "hq160" <10092020 at kamailio>;tag=8c90b1379825fa62. To: ;tag=2BZ1HSHNQ19tj. Call-ID: 04d14ab631843dc1 at 192.168.2.13. CSeq: 31348 INVITE. Content-Length: 0. . U 2009/09/08 09:17:41.427901 kamailio:5060 -> freeswitch:5060 ACK sip:15145555555 at gw2:5061 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.1. From: "hq160" <10092020 at kamailio>;tag=8c90b1379825fa62. Call-ID: 04d14ab631843dc1 at 192.168.2.13. To: ;tag=2BZ1HSHNQ19tj. CSeq: 31348 ACK. Max-Forwards: 70. UserIP-Agent: Kamailio (1.4.4-tls (i386/linux)). Content-Length: 0. . --------> Route 3 U 2009/09/08 09:17:44.206445 kamailio:5060 -> freeswitch:5060 INVITE sip:5145555555 at gw3:5060 SIP/2.0. Record-Route: . Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.2. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. From: "hq160" <10092020 at kamailio>;tag=8c90b1379825fa62. To: . Contact: . Supported: replaces, timer, path. Call-ID: 04d14ab631843dc1 at 192.168.2.13. CSeq: 31348 INVITE. Max-Forwards: 68. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK. Content-Type: application/sdp. Content-Length: 396. . v=0. o=10092020 8000 8001 IN IP4 192.168.2.13. s=SIP Call. c=IN IP4 MediaServer. t=0 0. m=audio 50362 RTP/AVP 0 101. a=sendrecv. a=rtpmap:0 PCMU/8000. a=ptime:20. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-11. m=video 50366 RTP/AVP 99. a=sendrecv. a=rtpmap:99 H264/90000. a=fmtp:99 profile-level-id=42801E; packetization-mode=0; sprop-parameter-sets=J0KAFJWgUH5A,KM4CfIC=. a=framerate:15. U 2009/09/08 09:17:44.207261 freeswitch:5060 -> kamailio:5060 SIP/2.0 482 Request merged. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.2. Via: SIP/2.0/UDP 192.168.2.13:52060;rport=52060;received=UserIP;branch=z9hG4bK4d46edc4e1623ae5. From: "hq160" <10092020 at kamailio>;tag=8c90b1379825fa62. To: ;tag=2BZ1HSHNQ19tj. Call-ID: 04d14ab631843dc1 at 192.168.2.13. CSeq: 31348 INVITE. Content-Length: 0. . U 2009/09/08 09:17:44.207762 kamailio:5060 -> freeswitch:5060 ACK sip:5145555555 at gw3:5060 SIP/2.0. Via: SIP/2.0/UDP kamailio;branch=z9hG4bKede5.5c558357.2. From: "hq160" <10092020 at kamailio>;tag=8c90b1379825fa62. Call-ID: 04d14ab631843dc1 at 192.168.2.13. To: ;tag=2BZ1HSHNQ19tj. CSeq: 31348 ACK. Max-Forwards: 70. UserIP-Agent: Kamailio (1.4.4-tls (i386/linux)). Content-Length: 0. . ============================================ I'm going to gess the call-id is the same for the second transaction... can you provide a more detailed trace? /b On Sep 4, 2009, at 11:06 AM, Humberto Quintana wrote: > Hello, > > I'm a new Freeswitch user. After some reading I put Freeswitch > (Version 1.0.4) to work as Session Border Controller. I have only > one problem that I dont know how to solve it ( or which parameter to > set) and I'd appreciate if someone could give me a clue about this. > > Kamailio is sitting behind FS and it selects the route or routes in > case of failure (serial forking) . Freeswitch is configured to use > directly the Request-URI sent by Kamailio. > > So, when the 1st route fails, Kamailio receives the Reply from FS > and sends back the ACK to end the transaction. More than 1 second > later, a new INVITE from Kamailio with the next route is tried (With > the To-header's tag is empty. Same Callid, From and Cseq header but > different VIA-header's branch parameter) and FS is answering back > 482 Merged Request. It happens the same for the 3rd route. > > It seems that the transaction is still 'alive' in FS even if the ACK > was received ? > > > Thanks, > > Humberto _________________________________________________________________ Click less, chat more: Messenger on MSN.ca http://go.microsoft.com/?linkid=9677404 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/87df5b14/attachment-0001.html From brian at freeswitch.org Tue Sep 8 07:23:39 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Sep 2009 09:23:39 -0500 Subject: [Freeswitch-users] 482 Request merged, in serial forking In-Reply-To: References: Message-ID: <9E150828-D439-4010-BFF3-05B57D1E079B@freeswitch.org> Looks like FS is behind nat. You need to set local-network-acl and the ext-rtp-ip and ext-sip-ip so FreeSWITCH properly puts in the right IP's in the via headers and sdp. Please refer to internal.xml in the latest SVN for an example of how to do this. /b On Sep 8, 2009, at 9:00 AM, Humberto Quintana wrote: > Hi Brian, > > Yes , the Call-Id is the same for the 2nd and 3rd transaction but > the branch parameter in the Via header is different. Please check > the capture below. > > Thanks, > > Humberto -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/991564c8/attachment.html From christian.loeschenkohl at xpirio.com Tue Sep 8 07:28:06 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Tue, 08 Sep 2009 16:28:06 +0200 Subject: [Freeswitch-users] patch for debian init script Message-ID: <4AA669F6.6060605@xpirio.com> hi just a quick patch for the debian init script debian/freeswitch.init i do use the reload function and the script complains about the -C option it also would be perfect if the reload option is enabled by default (usefull for logrotating) -> combined in second patch br ********************************************************************************************* --- debian/freeswitch.init-old 2009-09-08 16:18:49.000000000 +0200 +++ debian/freeswitch.init 2009-09-08 16:19:13.000000000 +0200 @@ -103,7 +103,7 @@ # restarting (for example, when it is sent a SIGHUP), # then implement that here. # - start-stop-daemon -C $USER --stop --signal 1 --quiet --pidfile $PIDFILE --name $NAME + start-stop-daemon -c $USER --stop --signal 1 --quiet --pidfile $PIDFILE --name $NAME return 0 } ********************************************************************************************* --- debian/freeswitch.init-old 2009-09-08 16:26:01.000000000 +0200 +++ debian/freeswitch.init 2009-09-08 16:19:13.000000000 +0200 @@ -103,7 +103,7 @@ # restarting (for example, when it is sent a SIGHUP), # then implement that here. # - start-stop-daemon -C $USER --stop --signal 1 --quiet --pidfile $PIDFILE --name $NAME + start-stop-daemon -c $USER --stop --signal 1 --quiet --pidfile $PIDFILE --name $NAME return 0 } @@ -124,15 +124,15 @@ 2) [ "$VERBOSE" != no ] && log_end_msg 1 ;; esac ;; - reload|force-reload) + #reload|force-reload) # # If do_reload() is not implemented then leave this commented out # and leave 'force-reload' as an alias for 'restart'. # - log_daemon_msg "Reloading $DESC" "$NAME" - do_reload - log_end_msg $? - ;; + #log_daemon_msg "Reloading $DESC" "$NAME" + #do_reload + #log_end_msg $? + #;; restart|force-reload) # # If the "reload" option is implemented then remove the @@ -156,8 +156,8 @@ esac ;; *) - echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force-reload}" >&2 - #echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}" >&2 + #echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force-reload}" >&2 + echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}" >&2 exit 3 ;; esac -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com From mrene_lists at avgs.ca Tue Sep 8 08:02:16 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 8 Sep 2009 08:02:16 -0700 Subject: [Freeswitch-users] Usefull option for mod_cdr_csv In-Reply-To: <1cd828b60909080520o5b592a6bl52a5052986429bf9@mail.gmail.com> References: <1cd828b60909080520o5b592a6bl52a5052986429bf9@mail.gmail.com> Message-ID: <5450F00D-F83E-4A54-8D9F-ECFFBCACCE12@avgs.ca> Its there now. Cheers, Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 8-Sep-09, at 5:20 AM, Anatoliy Kounitskiy wrote: > There is another useful option for the module Mod_cdr_csv, that is not > described in the default configuration file: > > > > If you use the default configuration file and you're using the > variable accountcode, probably you've seen for every accountcode you > have separate cdr file for it. But if you need only one file with all > cdr - use the option above. Possible values are "true" or "false". > > Cheers, > -- > Anatoliy Kounitskiy > ------------------------- > E-mail: anatoliy at kounitskiy.com > Mobile: +359898913540 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mgende at gendesign.com Mon Sep 7 20:28:25 2009 From: mgende at gendesign.com (Michael Gende) Date: Mon, 7 Sep 2009 22:28:25 -0500 Subject: [Freeswitch-users] ESL: DTMF event is not coming In-Reply-To: <25326328.post@talk.nabble.com> References: <25326328.post@talk.nabble.com> Message-ID: I'm pretty new at this, but please let me ask you a question: Is your FreeSwitch running on a dual-homed host? On Mon, Sep 7, 2009 at 2:47 AM, Nagalenoj wrote: > > Dear friends, > I am using freeswitch-1.0.4. When I execute the sample > script(/libs/esl/perl/server2.pl), it is not receiving the DTMF events. > When > I execute the same program in freeswitch-1.0.3, it's receiving the event. > Do > I miss something to configure/upgrade.? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090907/3a8280b6/attachment.html From rob4manhere at gmail.com Tue Sep 8 08:29:45 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Tue, 8 Sep 2009 10:29:45 -0500 Subject: [Freeswitch-users] Mod fax best practices Message-ID: <2A6CC956-A18F-4631-BB73-38E9EB6D2123@gmail.com> Hi all, I built a mod_fax setup which is working well. Hats off to the authors for the module. I receive faxes regularly without issue. I can also send faxes, but seem to have a higher failure rate, especially with faxes over 10 pages, usually with error 21: No response after sending a page. I know fax-over-IP can be dicey sometimes (don't know if T.38 would help but see that mod_fax doesn't support yet anyway), but here's my question: Are there any best practices when using mod_fax? Codecs to use or avoid, jitter settings, OS tuning, etc? Things that you guys have learned through live use. Sorry if its a newbie question but I've read through the documentation and wiki but haven't seen much in this. About the system: Dedicated hosted server, Debian 5, Freeswitch 1.0.4. Nothing else running on the system. Any thoughts or lessons learned would be greatly appreciated. Cheers, Rob Forman From anthony.minessale at gmail.com Tue Sep 8 08:44:27 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Sep 2009 10:44:27 -0500 Subject: [Freeswitch-users] Custom Variables In-Reply-To: <58740.1252418330@a2unlimited.com> References: <58740.1252418330@a2unlimited.com> Message-ID: <191c3a030909080844w18a992bajf1d9d4103a7200b0@mail.gmail.com> does that happen to be the conference event you are talking about? there is a separate verbose-events param in the conference profile to do that. On Tue, Sep 8, 2009 at 8:58 AM, Tina Martinez wrote: > Hello, > > I have created custom variables that I pass to my FreeSWITCH dialplan. > In my code I am monitoring the events and using the variables (now using > ESL, > very nice). > The custom variables appear in events such as CHANNEL_ORIGINATE, but I do > not see > the variable in the CUSTOM event. > > For example, in CHANNEL_ORIGINATE, I see the custom variables as: > > variable_sip_h_X-custom_variable1 : ABC > variable_sip_h_X-custom_variable2 : XYZ > > Is there a way for me to see the variables in the CUSTOM event as well? > > In my dialplan, I do have action application="verbose_events" data="true". > But it does not seem to help. > > Any thoughts? > > - T > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/89bd9464/attachment.html From foxb at abv.bg Tue Sep 8 08:50:53 2009 From: foxb at abv.bg (Hristo Benev) Date: Tue, 8 Sep 2009 18:50:53 +0300 (EEST) Subject: [Freeswitch-users] mod_cdr_csv and mysql Message-ID: <1204832208.94051.1252425053530.JavaMail.apache@mail23.abv.bg> Hello, I saw an sql option in mod_cdr_csv. For my surprise it wrote sql code in Master.csv file instead of recording to mysql database (already setup as ODBC) Is that normal or I'm missing something? I read on the wiki that there is additional perl script to load csv to mysql. Should I do it in that way? Thanks, From diego.viola at gmail.com Tue Sep 8 09:04:20 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 8 Sep 2009 16:04:20 +0000 Subject: [Freeswitch-users] mod_cdr_csv and mysql In-Reply-To: <1204832208.94051.1252425053530.JavaMail.apache@mail23.abv.bg> References: <1204832208.94051.1252425053530.JavaMail.apache@mail23.abv.bg> Message-ID: <86a32abc0909080904p16e2c6c5oda3b8430550fedfb@mail.gmail.com> Hi Hristo, I recommend that you take a look at mod_event_socket or ESL/FSR, you should make a script that listens for the CHANNEL_HANGUP_COMPLETE and get all the CDR vars from that event, once you do that you can send the CDR info to the db or do anything else. You can find an example of how to do that here: http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/callcard/cdr.rb Best regards, Diego On Tue, Sep 8, 2009 at 3:50 PM, Hristo Benev wrote: > Hello, > > I saw an sql option in mod_cdr_csv. > > For my surprise it wrote sql code in Master.csv file instead of recording > to mysql database (already setup as ODBC) > > Is that normal or I'm missing something? > > I read on the wiki that there is additional perl script to load csv to > mysql. > > Should I do it in that way? > > Thanks, > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/1a191019/attachment-0001.html From raffaele.p.guidi at gmail.com Tue Sep 8 09:08:27 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 8 Sep 2009 18:08:27 +0200 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: Message-ID: Hi, you just have to use delegates to asynchronously call the function containing the loop and return back the control to the calling thread. Here an example (don't have my code at hand, hope it doesn't contain typos). Regards, Raffaele public class TestLoop : ILoadNotificationPlugin { Delegate void DoStuffDelegate(); public void doStuff() { EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } } public bool Load() { DoStuffDelegate dsdlg = new DoStuffDelegate(doStuff); dsdlg.BeginInvoke(); } } On Tue, Sep 8, 2009 at 08:21, Josh Rivers wrote: > Thanks for the response! > I have tried putting a long-running loop here, but then it blocks anything > else managed from happening: > > public class TestLoop : ILoadNotificationPlugin > { > public bool Load() > { > EventConsumer con = new EventConsumer("all", ""); > while (true) > { > Event ev = con.pop(0); > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > freeswitch.msleep(100); > } > } > } > > However, if I fork off a thread here, freeswitch crashes: > public class TestLoop : ILoadNotificationPlugin > { > public bool Load() > { > ThreadPool.QueueUserWorkItem((o) => > { > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > EventConsumer con = new EventConsumer("all", ""); > while (true) > { > Event ev = con.pop(0); > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > freeswitch.msleep(100); > } > }); > return true; > } > } > > It doesn't look like this is a good place to start a long-running process? > > Thanks! > Josh > > On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi < > raffaele.p.guidi at gmail.com> wrote: > >> Yes! >> public class LoadDemo : ILoadNotificationPlugin { >> public bool Load() { >> Log.WriteLine(LogLevel.Notice, "LoadDemo running."); >> return true; >> } >> } >> >> this example is from Michael Giagnocavo's Demo.csx which you can find into >> the mod_managed svn. >> >> And let me add that works like a charm :) >> >> Ciao, >> Raffaele >> >> On Sun, Sep 6, 2009 at 22:50, Josh Rivers wrote: >> >>> Is there a way to start this when FreeSWITCH starts? The lua and perl >>> modules have a 'startup-script' configuration preference. Is there something >>> similar in mod_managed? Or is there a way to have an api command executed at >>> a startup? >>> >>> >>> Exactly what I was after - thank you! >>> >>> On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk wrote: >>> >>> > >>> > try something like this >>> > >>> > EventConsumer con = new EventConsumer("all", ""); >>> > Event ev = con.pop(0); >>> > >>> > see lua sample - >>> > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer >>> > >>> > >>> > Phillip Jones-2 wrote: >>> > > >>> > > Hi there, >>> > > >>> > > mod_managed exposes EventReceivedFunction such that: >>> > > >>> > > Session.EventReceivedFunction = (e) => >>> > > { >>> > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", >>> e.ToString()); >>> > > return ""; >>> > > }; >>> > > >>> > > should trap all events to which i subscribe. >>> > > >>> > > >>> > > But how do I subscribe to events? What is the .NET / managed >>> equivalent >>> > > of: >>> > > >>> > > switch_event_bind(const char *id, switch_event_types_t event, const >>> char >>> > > *subclass_name, switch_event_callback_t callback, void *user_data); >>> > > >>> > > >>> > > >>> > > Thank you! >>> > > >>> > > >>> > > >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/0c4216c4/attachment.html From hjqlopez at hotmail.com Tue Sep 8 09:16:04 2009 From: hjqlopez at hotmail.com (Humberto Quintana) Date: Tue, 8 Sep 2009 12:16:04 -0400 Subject: [Freeswitch-users] 482 Request merged, in serial forking Message-ID: Hi Brian, Thank you very much for your answer but both, Freeswitch and Kamailio have public IPs, it's my NAT'd IP phone who has private IP but this is fixed by Kamailio. The problem is not the 1st call is failing ( the test is set that way), the problem is FS answers back 482 when Kamailio tries a 2nd route ( or 3rd ) for the same call... Freeswitch is configured to use the Requested-URI sent by Kamailio: I noticed that there is no Log message in Freeswitch when receiving the INVITE for the 2nd route. The process in FS seems to be destroyed (11:46:21.396593) before the 2nd INVITE is received (11:46:21.401419 ). U 2009/09/08 11:46:21.395702 freeswitch:5060 -> kamailio:5060 SIP/2.0 503 Service Unavailable. Call-ID: ba748cd27cd163b5 at 192.168.2.13 U 2009/09/08 11:46:21.395897 kamailio:5060 -> freeswitch:5060 ACK sip:5145555555 at gw1:5060 SIP/2.0. Call-ID: ba748cd27cd163b5 at 192.168.2.13 U 2009/09/08 11:46:21.401419 kamailio:5060 -> freeswitch:5060 INVITE sip:15145555555 at gw2:5061 SIP/2.0. Call-ID: ba748cd27cd163b5 at 192.168.2.13 U 2009/09/08 11:46:21.401845 freeswitch:5060 -> kamailio:5060 SIP/2.0 482 Request merged. Call-ID: ba748cd27cd163b5 at 192.168.2.13 2009-09-08 11:46:21.395503 [DEBUG] mod_sofia.c:417 Responding to INVITE with: 503 2009-09-08 11:46:21.395503 [DEBUG] switch_core_state_machine.c:46 sofia/external/10092020 at freeswitch Standard HANGUP, cause: NORMAL_TEMPORARY_FAILURE 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:434 (sofia/external/10092020 at freeswitch) State HANGUP going to sleep 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:476 (sofia/external/10092020 at freeswitch) State Change CS_HANGUP -> CS_REPORTING 2009-09-08 11:46:21.396593 [DEBUG] switch_core_session.c:932 Send signal sofia/external/10092020 at freeswitch [BREAK] 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:398 (sofia/external/10092020 at freeswitch) Running State Change CS_REPORTING 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:612 (sofia/external/10092020 at freeswitch) State REPORTING 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:53 sofia/external/10092020 at freeswitch Standard REPORTING, cause: NORMAL_TEMPORARY_FAILURE 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:612 (sofia/external/10092020 at freeswitch) State REPORTING going to sleep 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:411 (sofia/external/10092020 at freeswitch) State Change CS_REPORTING -> CS_DESTROY 2009-09-08 11:46:21.396593 [DEBUG] switch_core_session.c:1068 Session 3 (sofia/external/10092020 at freeswitch) Locked, Waiting on external entities 2009-09-08 11:46:21.396593 [NOTICE] switch_core_session.c:1086 Session 3 (sofia/external/10092020 at freeswitch) Ended 2009-09-08 11:46:21.396593 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/10092020 at freeswitch [CS_DESTROY] 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:564 (sofia/external/10092020 at freeswitch) State DESTROY 2009-09-08 11:46:21.396593 [DEBUG] mod_sofia.c:255 sofia/external/10092020 at freeswitch SOFIA DESTROY 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:60 sofia/external/10092020 at freeswitch Standard DESTROY 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:564 (sofia/external/10092020 at freeswitch) State DESTROY going to sleep Note: I'm using only the external sofia profile. Thanks, Humberto ========================================== Looks like FS is behind nat. You need to set local-network-acl and the ext-rtp-ip and ext-sip-ip so FreeSWITCH properly puts in the right IP's in the via headers and sdp. Please refer to internal.xml in the latest SVN for an example of how to do this. /b _________________________________________________________________ New! Open Messenger faster on the MSN homepage http://go.microsoft.com/?linkid=9677405 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/645dca1b/attachment.html From jerry.richards at teotech.com Tue Sep 8 09:25:23 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 8 Sep 2009 09:25:23 -0700 Subject: [Freeswitch-users] Minimum/Recommended Freeswitch SystemConfiguration In-Reply-To: <8672FA11-0B1C-4967-852F-6B2D82A1B405@enterux.com> References: <93C3A49CAB7743DFB845CFA0A8714A4E@greyhawk.tonecommander.com> <8672FA11-0B1C-4967-852F-6B2D82A1B405@enterux.com> Message-ID: <0D2DE92AE7BD49AB9E856EFD39116A4F@greyhawk.tonecommander.com> Mitul, Thank you for your reply. Freeswitch is new to me, so I am not yet able to take measurements of FS under a load of traffic. I was just asking for future planning purposes. After I do some more development with it perhaps I can record some of these measurements. Thanks and Regards, Jerry -----Original Message----- From: Mitul Limbani [mailto:mitul at enterux.com] Sent: Friday, September 04, 2009 2:21 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Minimum/Recommended Freeswitch SystemConfiguration Jerry, As far as I understand freeswitch, it using kernel to thread and this operation eats good amount of RAM, but since the internal strructure of fs is to store all these sip details in runtime sqlite db, which is compressed text data earlier written in XML but while fs loads this configs it gets it in sqlite and that's what it used instead of asterisks astdb. Although what you see as recommended config for 500 users is true but it also depends on which processor you are trying this on. Intel or AMD is still ok but if you trying it on arm I don't have any data as such, interestingly if you have some test hardware scenario you can actually test and let us all know about it, it's quite useful bit of info that can be positioned on the FS Wiki, in case you want to take this experiment offlist do write to me, im interested to document :) Look forward to hear from you, Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 05-Sep-2009, at 12:03 AM, "Jerry Richards" wrote: > > Under the Minimum/Recommended System Requirements, what is meant by > "We recommend you plan for 50% duty cycle"? What is this duty cycle? > > Also, I see that the system requirements indicate Freeswitch > recommends 1GB RAM and 50MB disk space. I guess I'm wondering how the > number of extensions and external interfaces drive size of RAM and > disk space? For example, would these recommendations support 100 > extensions and one external interface? 500 extensions and 10 external > interfaces? Etc.? > > Best Regards, > Jerry > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Tue Sep 8 09:26:34 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 8 Sep 2009 12:26:34 -0400 Subject: [Freeswitch-users] patch for debian init script In-Reply-To: <4AA669F6.6060605@xpirio.com> References: <4AA669F6.6060605@xpirio.com> Message-ID: Please post this patch to http://jira.freeswitch.org in the build system project and I will get this merged in Mike On Sep 8, 2009, at 10:28 AM, Christian L?schenkohl wrote: > hi > > just a quick patch for the debian init script debian/freeswitch.init > i do use the reload function and the script complains about the -C > option > it also would be perfect if the reload option is enabled by default > (usefull > for logrotating) -> combined in second patch > > br > > ********************************************************************************************* > > --- debian/freeswitch.init-old 2009-09-08 16:18:49.000000000 +0200 > +++ debian/freeswitch.init 2009-09-08 16:19:13.000000000 +0200 > @@ -103,7 +103,7 @@ > # restarting (for example, when it is sent a SIGHUP), > # then implement that here. > # > - start-stop-daemon -C $USER --stop --signal 1 --quiet -- > pidfile $PIDFILE --name $NAME > + start-stop-daemon -c $USER --stop --signal 1 --quiet -- > pidfile $PIDFILE --name $NAME > return 0 > } > > ********************************************************************************************* > > --- debian/freeswitch.init-old 2009-09-08 16:26:01.000000000 +0200 > +++ debian/freeswitch.init 2009-09-08 16:19:13.000000000 +0200 > @@ -103,7 +103,7 @@ > # restarting (for example, when it is sent a SIGHUP), > # then implement that here. > # > - start-stop-daemon -C $USER --stop --signal 1 --quiet -- > pidfile $PIDFILE --name $NAME > + start-stop-daemon -c $USER --stop --signal 1 --quiet -- > pidfile $PIDFILE --name $NAME > return 0 > } > > @@ -124,15 +124,15 @@ > 2) [ "$VERBOSE" != no ] && log_end_msg 1 ;; > esac > ;; > - reload|force-reload) > + #reload|force-reload) > # > # If do_reload() is not implemented then leave this > commented out > # and leave 'force-reload' as an alias for 'restart'. > # > - log_daemon_msg "Reloading $DESC" "$NAME" > - do_reload > - log_end_msg $? > - ;; > + #log_daemon_msg "Reloading $DESC" "$NAME" > + #do_reload > + #log_end_msg $? > + #;; > restart|force-reload) > # > # If the "reload" option is implemented then remove the > @@ -156,8 +156,8 @@ > esac > ;; > *) > - echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force- > reload}" >&2 > - #echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}" > >&2 > + #echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force- > reload}" >&2 > + echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}" > >&2 > exit 3 > ;; > esac > > > -- > Ing. Christian L?schenkohl > 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.loeschenkohl at xpirio.com From tina at a2unlimited.com Tue Sep 8 09:27:00 2009 From: tina at a2unlimited.com (Tina Martinez) Date: Tue, 08 Sep 2009 12:27:00 -0400 Subject: [Freeswitch-users] Custom Variables Message-ID: <47720.1252427220@a2unlimited.com> Yes, it is a conference event. In looking at the mod_conference wiki, I see verbose-events mentioned, but the example value is ???. What is an appropriate setting for this param? (i.e., is it numeric, boolean, latin?) -T From mgg at giagnocavo.net Tue Sep 8 09:43:00 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 8 Sep 2009 12:43:00 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4C0C@mse17be1.mse17.exchange.ms> That's what his sample does, but he says it crashes. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raffaele P. Guidi Sent: Tuesday, September 08, 2009 10:08 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET Hi, you just have to use delegates to asynchronously call the function containing the loop and return back the control to the calling thread. Here an example (don't have my code at hand, hope it doesn't contain typos). Regards, Raffaele public class TestLoop : ILoadNotificationPlugin { Delegate void DoStuffDelegate(); public void doStuff() { EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } } public bool Load() { DoStuffDelegate dsdlg = new DoStuffDelegate(doStuff); dsdlg.BeginInvoke(); } } On Tue, Sep 8, 2009 at 08:21, Josh Rivers > wrote: Thanks for the response! I have tried putting a long-running loop here, but then it blocks anything else managed from happening: public class TestLoop : ILoadNotificationPlugin { public bool Load() { EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } } } However, if I fork off a thread here, freeswitch crashes: public class TestLoop : ILoadNotificationPlugin { public bool Load() { ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } }); return true; } } It doesn't look like this is a good place to start a long-running process? Thanks! Josh On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi > wrote: Yes! public class LoadDemo : ILoadNotificationPlugin { public bool Load() { Log.WriteLine(LogLevel.Notice, "LoadDemo running."); return true; } } this example is from Michael Giagnocavo's Demo.csx which you can find into the mod_managed svn. And let me add that works like a charm :) Ciao, Raffaele On Sun, Sep 6, 2009 at 22:50, Josh Rivers > wrote: Is there a way to start this when FreeSWITCH starts? The lua and perl modules have a 'startup-script' configuration preference. Is there something similar in mod_managed? Or is there a way to have an api command executed at a startup? Exactly what I was after - thank you! On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk > wrote: > > try something like this > > EventConsumer con = new EventConsumer("all", ""); > Event ev = con.pop(0); > > see lua sample - > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer > > > Phillip Jones-2 wrote: > > > > Hi there, > > > > mod_managed exposes EventReceivedFunction such that: > > > > Session.EventReceivedFunction = (e) => > > { > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", e.ToString()); > > return ""; > > }; > > > > should trap all events to which i subscribe. > > > > > > But how do I subscribe to events? What is the .NET / managed equivalent > > of: > > > > switch_event_bind(const char *id, switch_event_types_t event, const char > > *subclass_name, switch_event_callback_t callback, void *user_data); > > > > > > > > Thank you! > > > > > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/20840b11/attachment-0001.html From anthony.minessale at gmail.com Tue Sep 8 09:43:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Sep 2009 11:43:40 -0500 Subject: [Freeswitch-users] Custom Variables In-Reply-To: <47720.1252427220@a2unlimited.com> References: <47720.1252427220@a2unlimited.com> Message-ID: <191c3a030909080943i732a331cv4cd2a49595351fe3@mail.gmail.com> "true" should suffice. On Tue, Sep 8, 2009 at 11:27 AM, Tina Martinez wrote: > Yes, it is a conference event. > > In looking at the mod_conference wiki, I see verbose-events mentioned, but > the > example value is ???. > > What is an appropriate setting for this param? (i.e., is it numeric, > boolean, latin?) > > -T > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/861b10b9/attachment.html From anthony.minessale at gmail.com Tue Sep 8 09:44:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Sep 2009 11:44:40 -0500 Subject: [Freeswitch-users] Recording Only 1 Leg of a Call In-Reply-To: <4256bf830909072150i73ceca68we9476e2a57b53f6a@mail.gmail.com> References: <4256bf830909070826o5cb0d7a4lcd27e07ee350b50d@mail.gmail.com> <4256bf830909072150i73ceca68we9476e2a57b53f6a@mail.gmail.com> Message-ID: <191c3a030909080944o3fa6f386ua42c56e3535ffd69@mail.gmail.com> that would have to be filed as a feature request as we do not currently have a way to do that. On Mon, Sep 7, 2009 at 11:50 PM, Matthew Fong wrote: > I want to record without the telephone user's interaction. > I think uuid_record should have the option to only record the audio of the > uuid channel that is being specified, and as a secondary option combine the > audio of the b leg (since uuid_record only specifies one uuid anyway--this > seems logical). > > Anyway, just my wish list :) > > --matt > http://www.hellohunter.com > voice broadcasting & hosted dialer > > On Tue, Sep 8, 2009 at 2:12 AM, Milena wrote: > >> Hello, >> What about this?: >> " >> >> >> " >> >> the person would have to press *2 during the call to start the recording. >> >> 2009/9/7 Matthew Fong >> >>> Whats the best way to record only one leg of a call? >>> uuid_record records both channels >>> session_record does the same (but has a stereo option) >>> >>> is there any way to record only an a-leg of the call? Thanks so much. >>> >>> --matt >>> http://www.hellohunter.com >>> hosted dialer & voice broadcasting >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/eabaf19b/attachment.html From bjbrashier at gmail.com Tue Sep 8 10:26:44 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 8 Sep 2009 11:26:44 -0600 Subject: [Freeswitch-users] Conference DTMFs heard by participants Message-ID: <7bcfdd290909081026n1282f55xdc4d6e4d000c2e8e@mail.gmail.com> I have a FreeSWITCH conference with a list of DTMFs, some of which are handled through the event socket (like mute-all), some of which are handled by FreeSWITCH itself (like mute-self). There are a number of commands available and all of them are 2 digits in length. The issue is that when a command is pressed on one phone in the conference, all users hear the tones of the first key pressed. My expectation is that no other users should hear any of the keys, at least not unless they do not correspond to any command. This happens regardless of whether it's a command processed by the event app or if it's processed by FS. I have tried using single-digit commands and the same thing happens -- that single digit is heard by all conference members. Questions: 1) Is this expected behavior (ie. is there some reason you would want this)? 2) Is this something that I can change with some parameter somewhere? BB From msc at freeswitch.org Tue Sep 8 10:35:45 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 8 Sep 2009 10:35:45 -0700 Subject: [Freeswitch-users] mod_cdr_csv and mysql In-Reply-To: <1204832208.94051.1252425053530.JavaMail.apache@mail23.abv.bg> References: <1204832208.94051.1252425053530.JavaMail.apache@mail23.abv.bg> Message-ID: <87f2f3b90909081035v2060e913l30aae5378936713c@mail.gmail.com> On Tue, Sep 8, 2009 at 8:50 AM, Hristo Benev wrote: > Hello, > > I saw an sql option in mod_cdr_csv. > > For my surprise it wrote sql code in Master.csv file instead of recording > to mysql database (already setup as ODBC) > > Is that normal or I'm missing something? > The purpose of the SQL template is to create SQL-ready commands that will load your CDRs into a table. It does not perform the operation itself, it merely creates the records in a file on the HDD. > > I read on the wiki that there is additional perl script to load csv to > mysql. > > Should I do it in that way? > > That is one way to do it. By dropping the records into a disk file you have a layer of protection for those annoying occasions when your db goes down. Other ways to handle CDRs would include the event socket (which Diego mentioned in his post) as well as mod_xml_cdr. They each have their advantages. Using SQL statements in CSV is pretty easy compared to the other options, however the other options will give you much more information about each call. If the CSV records contain all the data you need then I would use them since it's the easiest to implement. -MC > Thanks, > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/568e0e31/attachment.html From anthony.minessale at gmail.com Tue Sep 8 10:48:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Sep 2009 12:48:35 -0500 Subject: [Freeswitch-users] Conference DTMFs heard by participants In-Reply-To: <7bcfdd290909081026n1282f55xdc4d6e4d000c2e8e@mail.gmail.com> References: <7bcfdd290909081026n1282f55xdc4d6e4d000c2e8e@mail.gmail.com> Message-ID: <191c3a030909081048k149955d7le90237f33ac29a13@mail.gmail.com> you must have some tdm equipment somewhere that is decoding the dtmf tones and passing them w/o removing them from the audio stream. On Tue, Sep 8, 2009 at 12:26 PM, Bradley Brashier wrote: > I have a FreeSWITCH conference with a list of DTMFs, some of which are > handled through the event socket (like mute-all), some of which are > handled by FreeSWITCH itself (like mute-self). There are a number of > commands available and all of them are 2 digits in length. > > The issue is that when a command is pressed on one phone in the > conference, all users hear the tones of the first key pressed. My > expectation is that no other users should hear any of the keys, at > least not unless they do not correspond to any command. This happens > regardless of whether it's a command processed by the event app or if > it's processed by FS. I have tried using single-digit commands and the > same thing happens -- that single digit is heard by all conference > members. > > Questions: > 1) Is this expected behavior (ie. is there some reason you would want > this)? > 2) Is this something that I can change with some parameter somewhere? > > BB > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/2c92313a/attachment-0001.html From brian at freeswitch.org Tue Sep 8 10:53:03 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 8 Sep 2009 12:53:03 -0500 Subject: [Freeswitch-users] Conference DTMFs heard by participants In-Reply-To: <7bcfdd290909081026n1282f55xdc4d6e4d000c2e8e@mail.gmail.com> References: <7bcfdd290909081026n1282f55xdc4d6e4d000c2e8e@mail.gmail.com> Message-ID: Its your gateway provider not squelching the DTMF I suspect. /b On Sep 8, 2009, at 12:26 PM, Bradley Brashier wrote: > The issue is that when a command is pressed on one phone in the > conference, all users hear the tones of the first key pressed. My > expectation is that no other users should hear any of the keys, at > least not unless they do not correspond to any command. This happens > regardless of whether it's a command processed by the event app or if > it's processed by FS. I have tried using single-digit commands and the > same thing happens -- that single digit is heard by all conference > members. From bjbrashier at gmail.com Tue Sep 8 10:53:15 2009 From: bjbrashier at gmail.com (Bradley Brashier) Date: Tue, 8 Sep 2009 11:53:15 -0600 Subject: [Freeswitch-users] Conference DTMFs heard by participants In-Reply-To: <191c3a030909081048k149955d7le90237f33ac29a13@mail.gmail.com> References: <7bcfdd290909081026n1282f55xdc4d6e4d000c2e8e@mail.gmail.com> <191c3a030909081048k149955d7le90237f33ac29a13@mail.gmail.com> Message-ID: <7bcfdd290909081053o7386b843tf8296c3e46430be3@mail.gmail.com> Good thought. I'll look into that. BB On Tue, Sep 8, 2009 at 11:48 AM, Anthony Minessale wrote: > you must have some tdm equipment somewhere that is decoding the dtmf tones > and passing them w/o removing them from the audio stream. > > > On Tue, Sep 8, 2009 at 12:26 PM, Bradley Brashier > wrote: >> >> I have a FreeSWITCH conference with a list of DTMFs, some of which are >> handled through the event socket (like mute-all), some of which are >> handled by FreeSWITCH itself (like mute-self). There are a number of >> commands available and all of them are 2 digits in length. >> >> The issue is that when a command is pressed on one phone in the >> conference, all users hear the tones of the first key pressed. My >> expectation is that no other users should hear any of the keys, at >> least not unless they do not correspond to any command. This happens >> regardless of whether it's a command processed by the event app or if >> it's processed by FS. I have tried using single-digit commands and the >> same thing happens -- that single digit is heard by all conference >> members. >> >> Questions: >> 1) Is this expected behavior (ie. is there some reason you would want >> this)? >> 2) Is this something that I can change with some parameter somewhere? >> >> BB >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From pjintheusa at gmail.com Tue Sep 8 11:12:05 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 8 Sep 2009 14:12:05 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: Message-ID: <367751820909081112t759bed1eve65ba8feab0ee8c4@mail.gmail.com> What is: freeswitch.msleep(100); Why aren't you using Thread.Sleep? On Tue, Sep 8, 2009 at 2:21 AM, Josh Rivers wrote: > Thanks for the response! > I have tried putting a long-running loop here, but then it blocks anything > else managed from happening: > > public class TestLoop : ILoadNotificationPlugin > { > public bool Load() > { > EventConsumer con = new EventConsumer("all", ""); > while (true) > { > Event ev = con.pop(0); > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > freeswitch.msleep(100); > } > } > } > > However, if I fork off a thread here, freeswitch crashes: > public class TestLoop : ILoadNotificationPlugin > { > public bool Load() > { > ThreadPool.QueueUserWorkItem((o) => > { > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > EventConsumer con = new EventConsumer("all", ""); > while (true) > { > Event ev = con.pop(0); > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > freeswitch.msleep(100); > } > }); > return true; > } > } > > It doesn't look like this is a good place to start a long-running process? > > Thanks! > Josh > > On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi < > raffaele.p.guidi at gmail.com> wrote: > >> Yes! >> public class LoadDemo : ILoadNotificationPlugin { >> public bool Load() { >> Log.WriteLine(LogLevel.Notice, "LoadDemo running."); >> return true; >> } >> } >> >> this example is from Michael Giagnocavo's Demo.csx which you can find into >> the mod_managed svn. >> >> And let me add that works like a charm :) >> >> Ciao, >> Raffaele >> >> On Sun, Sep 6, 2009 at 22:50, Josh Rivers wrote: >> >>> Is there a way to start this when FreeSWITCH starts? The lua and perl >>> modules have a 'startup-script' configuration preference. Is there something >>> similar in mod_managed? Or is there a way to have an api command executed at >>> a startup? >>> >>> >>> Exactly what I was after - thank you! >>> >>> On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk wrote: >>> >>> > >>> > try something like this >>> > >>> > EventConsumer con = new EventConsumer("all", ""); >>> > Event ev = con.pop(0); >>> > >>> > see lua sample - >>> > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer >>> > >>> > >>> > Phillip Jones-2 wrote: >>> > > >>> > > Hi there, >>> > > >>> > > mod_managed exposes EventReceivedFunction such that: >>> > > >>> > > Session.EventReceivedFunction = (e) => >>> > > { >>> > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", >>> e.ToString()); >>> > > return ""; >>> > > }; >>> > > >>> > > should trap all events to which i subscribe. >>> > > >>> > > >>> > > But how do I subscribe to events? What is the .NET / managed >>> equivalent >>> > > of: >>> > > >>> > > switch_event_bind(const char *id, switch_event_types_t event, const >>> char >>> > > *subclass_name, switch_event_callback_t callback, void *user_data); >>> > > >>> > > >>> > > >>> > > Thank you! >>> > > >>> > > >>> > > >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/148a01e1/attachment.html From raffaele.p.guidi at gmail.com Tue Sep 8 11:22:07 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 8 Sep 2009 20:22:07 +0200 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4C0C@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4C0C@mse17be1.mse17.exchange.ms> Message-ID: Well, I can't see any delegate in josh sample, just a ThreadPool.QueueUserWorkItem. Here is an example that, at least on my system (I reached my home pc in the meanwhile), works fine. public class LoadPluginDemo : ILoadNotificationPlugin { delegate void Listener(); private void EventListener() { EventConsumer con = new EventConsumer("all", null); while (true){ Event ev = con.pop(1); Log.WriteLine(LogLevel.Notice, "Got event " + ev.GetHeader("Event-Name")); } } public bool Load() { Log.WriteLine(LogLevel.Notice, "LoadDemo running."); new Listener(EventListener).BeginInvoke(null,null); return true; } } On Tue, Sep 8, 2009 at 18:43, Michael Giagnocavo wrote: > That?s what his sample does, but he says it crashes. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Raffaele P. > Guidi > *Sent:* Tuesday, September 08, 2009 10:08 AM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > Hi, you just have to use delegates to asynchronously call the function > containing the loop and return back the control to the calling thread. Here > an example (don't have my code at hand, hope it doesn't contain typos). > > > > Regards, > > Raffaele > > > > public class TestLoop : ILoadNotificationPlugin > > { > > > > Delegate void DoStuffDelegate(); > > > > public void doStuff() > > { > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > } > > public bool Load() > > { > > DoStuffDelegate dsdlg = new DoStuffDelegate(doStuff); > > dsdlg.BeginInvoke(); > > } > > } > > On Tue, Sep 8, 2009 at 08:21, Josh Rivers wrote: > > Thanks for the response! > > > > I have tried putting a long-running loop here, but then it blocks anything > else managed from happening: > > > > public class TestLoop : ILoadNotificationPlugin > > { > > public bool Load() > > { > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > } > > } > > > > However, if I fork off a thread here, freeswitch crashes: > > public class TestLoop : ILoadNotificationPlugin > > { > > public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => > > { > > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > }); > > return true; > > } > > } > > > > It doesn't look like this is a good place to start a long-running process? > > > > Thanks! > > Josh > > > > On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi < > raffaele.p.guidi at gmail.com> wrote: > > Yes! > > > > public class LoadDemo : ILoadNotificationPlugin { > > public bool Load() { > > Log.WriteLine(LogLevel.Notice, "LoadDemo running."); > > return true; > > } > > } > > > > this example is from Michael Giagnocavo's Demo.csx which you can find into > the mod_managed svn. > > > > And let me add that works like a charm :) > > > > Ciao, > > Raffaele > > > > On Sun, Sep 6, 2009 at 22:50, Josh Rivers wrote: > > Is there a way to start this when FreeSWITCH starts? The lua and perl > modules have a 'startup-script' configuration preference. Is there something > similar in mod_managed? Or is there a way to have an api command executed at > a startup? > > > > > > Exactly what I was after - thank you! > > > > On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk wrote: > > > > > > > > try something like this > > > > > > EventConsumer con = new EventConsumer("all", ""); > > > Event ev = con.pop(0); > > > > > > see lua sample - > > > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer > > > > > > > > > Phillip Jones-2 wrote: > > > > > > > > Hi there, > > > > > > > > mod_managed exposes EventReceivedFunction such that: > > > > > > > > Session.EventReceivedFunction = (e) => > > > > { > > > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", > e.ToString()); > > > > return ""; > > > > }; > > > > > > > > should trap all events to which i subscribe. > > > > > > > > > > > > But how do I subscribe to events? What is the .NET / managed equivalent > > > > of: > > > > > > > > switch_event_bind(const char *id, switch_event_types_t event, const > char > > > > *subclass_name, switch_event_callback_t callback, void *user_data); > > > > > > > > > > > > > > > > Thank you! > > > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/2035745f/attachment-0001.html From raffaele.p.guidi at gmail.com Tue Sep 8 11:50:20 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 8 Sep 2009 20:50:20 +0200 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <367751820909081112t759bed1eve65ba8feab0ee8c4@mail.gmail.com> References: <367751820909081112t759bed1eve65ba8feab0ee8c4@mail.gmail.com> Message-ID: Yeah, probably just using con.pop(1) - that waits for an event to be caught, instead than con.pop(0), thus avoiding "sleeps" would do the trick On Tue, Sep 8, 2009 at 20:12, Phillip Jones wrote: > What is: > > freeswitch.msleep(100); > > Why aren't you using Thread.Sleep? > > > On Tue, Sep 8, 2009 at 2:21 AM, Josh Rivers wrote: > >> Thanks for the response! >> I have tried putting a long-running loop here, but then it blocks anything >> else managed from happening: >> >> public class TestLoop : ILoadNotificationPlugin >> { >> public bool Load() >> { >> EventConsumer con = new EventConsumer("all", ""); >> while (true) >> { >> Event ev = con.pop(0); >> Log.WriteLine(LogLevel.Notice, "Event: " + >> ev.serialized_string); >> freeswitch.msleep(100); >> } >> } >> } >> >> However, if I fork off a thread here, freeswitch crashes: >> public class TestLoop : ILoadNotificationPlugin >> { >> public bool Load() >> { >> ThreadPool.QueueUserWorkItem((o) => >> { >> Log.WriteLine(LogLevel.Notice, "Thread Starting. "); >> EventConsumer con = new EventConsumer("all", ""); >> while (true) >> { >> Event ev = con.pop(0); >> Log.WriteLine(LogLevel.Notice, "Event: " + >> ev.serialized_string); >> freeswitch.msleep(100); >> } >> }); >> return true; >> } >> } >> >> It doesn't look like this is a good place to start a long-running process? >> >> Thanks! >> Josh >> >> On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi < >> raffaele.p.guidi at gmail.com> wrote: >> >>> Yes! >>> public class LoadDemo : ILoadNotificationPlugin { >>> public bool Load() { >>> Log.WriteLine(LogLevel.Notice, "LoadDemo running."); >>> return true; >>> } >>> } >>> >>> this example is from Michael Giagnocavo's Demo.csx which you can find >>> into the mod_managed svn. >>> >>> And let me add that works like a charm :) >>> >>> Ciao, >>> Raffaele >>> >>> On Sun, Sep 6, 2009 at 22:50, Josh Rivers wrote: >>> >>>> Is there a way to start this when FreeSWITCH starts? The lua and perl >>>> modules have a 'startup-script' configuration preference. Is there something >>>> similar in mod_managed? Or is there a way to have an api command executed at >>>> a startup? >>>> >>>> >>>> Exactly what I was after - thank you! >>>> >>>> On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk >>>> wrote: >>>> >>>> > >>>> > try something like this >>>> > >>>> > EventConsumer con = new EventConsumer("all", ""); >>>> > Event ev = con.pop(0); >>>> > >>>> > see lua sample - >>>> > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer >>>> > >>>> > >>>> > Phillip Jones-2 wrote: >>>> > > >>>> > > Hi there, >>>> > > >>>> > > mod_managed exposes EventReceivedFunction such that: >>>> > > >>>> > > Session.EventReceivedFunction = (e) => >>>> > > { >>>> > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", >>>> e.ToString()); >>>> > > return ""; >>>> > > }; >>>> > > >>>> > > should trap all events to which i subscribe. >>>> > > >>>> > > >>>> > > But how do I subscribe to events? What is the .NET / managed >>>> equivalent >>>> > > of: >>>> > > >>>> > > switch_event_bind(const char *id, switch_event_types_t event, const >>>> char >>>> > > *subclass_name, switch_event_callback_t callback, void *user_data); >>>> > > >>>> > > >>>> > > >>>> > > Thank you! >>>> > > >>>> > > >>>> > > >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/de19150f/attachment.html From christian.loeschenkohl at xpirio.com Tue Sep 8 12:29:27 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Tue, 08 Sep 2009 21:29:27 +0200 Subject: [Freeswitch-users] patch for debian init script In-Reply-To: References: <4AA669F6.6060605@xpirio.com> Message-ID: <4AA6B097.2030000@xpirio.com> is done br On 2009-09-08 18:26, Michael Jerris wrote: > Please post this patch to http://jira.freeswitch.org in the build > system project and I will get this merged in > > Mike > > On Sep 8, 2009, at 10:28 AM, Christian L?schenkohl wrote: > >> hi >> >> just a quick patch for the debian init script debian/freeswitch.init >> i do use the reload function and the script complains about the -C >> option >> it also would be perfect if the reload option is enabled by default >> (usefull >> for logrotating) -> combined in second patch >> >> br >> >> ********************************************************************************************* >> >> --- debian/freeswitch.init-old 2009-09-08 16:18:49.000000000 +0200 >> +++ debian/freeswitch.init 2009-09-08 16:19:13.000000000 +0200 >> @@ -103,7 +103,7 @@ >> # restarting (for example, when it is sent a SIGHUP), >> # then implement that here. >> # >> - start-stop-daemon -C $USER --stop --signal 1 --quiet -- >> pidfile $PIDFILE --name $NAME >> + start-stop-daemon -c $USER --stop --signal 1 --quiet -- >> pidfile $PIDFILE --name $NAME >> return 0 >> } >> >> ********************************************************************************************* >> >> --- debian/freeswitch.init-old 2009-09-08 16:26:01.000000000 +0200 >> +++ debian/freeswitch.init 2009-09-08 16:19:13.000000000 +0200 >> @@ -103,7 +103,7 @@ >> # restarting (for example, when it is sent a SIGHUP), >> # then implement that here. >> # >> - start-stop-daemon -C $USER --stop --signal 1 --quiet -- >> pidfile $PIDFILE --name $NAME >> + start-stop-daemon -c $USER --stop --signal 1 --quiet -- >> pidfile $PIDFILE --name $NAME >> return 0 >> } >> >> @@ -124,15 +124,15 @@ >> 2) [ "$VERBOSE" != no ]&& log_end_msg 1 ;; >> esac >> ;; >> - reload|force-reload) >> + #reload|force-reload) >> # >> # If do_reload() is not implemented then leave this >> commented out >> # and leave 'force-reload' as an alias for 'restart'. >> # >> - log_daemon_msg "Reloading $DESC" "$NAME" >> - do_reload >> - log_end_msg $? >> - ;; >> + #log_daemon_msg "Reloading $DESC" "$NAME" >> + #do_reload >> + #log_end_msg $? >> + #;; >> restart|force-reload) >> # >> # If the "reload" option is implemented then remove the >> @@ -156,8 +156,8 @@ >> esac >> ;; >> *) >> - echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force- >> reload}">&2 >> - #echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}" >>> &2 >> + #echo "Usage: $SCRIPTNAME {start|stop|restart|reload|force- >> reload}">&2 >> + echo "Usage: $SCRIPTNAME {start|stop|restart|force-reload}" >>> &2 >> exit 3 >> ;; >> esac >> >> >> -- >> Ing. Christian L?schenkohl >> 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.loeschenkohl at xpirio.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com From anthony.minessale at gmail.com Tue Sep 8 12:32:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 8 Sep 2009 14:32:05 -0500 Subject: [Freeswitch-users] ESL: DTMF event is not coming In-Reply-To: <25326328.post@talk.nabble.com> References: <25326328.post@talk.nabble.com> Message-ID: <191c3a030909081232g5550542fq4bde3a21c33b6a52@mail.gmail.com> did you specify the "async" keyword to the socket app in your dialplan? On Mon, Sep 7, 2009 at 2:47 AM, Nagalenoj wrote: > > Dear friends, > I am using freeswitch-1.0.4. When I execute the sample > script(/libs/esl/perl/server2.pl), it is not receiving the DTMF events. > When > I execute the same program in freeswitch-1.0.3, it's receiving the event. > Do > I miss something to configure/upgrade.? > -- > View this message in context: > http://www.nabble.com/ESL%3A-DTMF-event-is-not-coming-tp25326328p25326328.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/12611bdb/attachment.html From R.Kloosterman at mtel.nl Tue Sep 8 12:48:59 2009 From: R.Kloosterman at mtel.nl (Remko Kloosterman) Date: Tue, 8 Sep 2009 21:48:59 +0200 Subject: [Freeswitch-users] ESL: DTMF event is not coming In-Reply-To: References: <25326328.post@talk.nabble.com> Message-ID: <11372C8B9E603F4FACDE6AB18256DEC695A9E3@srvmtel.office.mtel.nl> I'm currently playing around with ESL myself. By default events are not sent to the application. You should enable this by with for example the "myevents" command. For a better understanding, try catching the IO with "nc -l -p 8084" for outbound socket applications. See http://wiki.freeswitch.org/wiki/Mod_event_socket for more info. Remko ________________________________ Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Michael Gende Verzonden: dinsdag 8 september 2009 5:28 Aan: freeswitch-users at lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] ESL: DTMF event is not coming I'm pretty new at this, but please let me ask you a question: Is your FreeSwitch running on a dual-homed host? On Mon, Sep 7, 2009 at 2:47 AM, Nagalenoj wrote: Dear friends, I am using freeswitch-1.0.4. When I execute the sample script(/libs/esl/perl/server2.pl), it is not receiving the DTMF events. When I execute the same program in freeswitch-1.0.3, it's receiving the event. Do I miss something to configure/upgrade.? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/62201e2b/attachment-0001.html From n.geordzhev at gmail.com Tue Sep 8 13:20:45 2009 From: n.geordzhev at gmail.com (Nikolai Geordzhev) Date: Tue, 8 Sep 2009 23:20:45 +0300 Subject: [Freeswitch-users] Call Forwarding Question In-Reply-To: <7d0bfd8c0909061654r30e6f906o5c9b25f1b27720cc@mail.gmail.com> References: <7d0bfd8c0909061654r30e6f906o5c9b25f1b27720cc@mail.gmail.com> Message-ID: I`ve already tried the legs variable in cdr_csv.conf.xml, I have also tried to use the loopback endpoint and to bridge the call to the internal interface(so it can go out and in again generating the 2cdr-s I need) and still haven`t achieved any success. Can anyone please share some experience in doing CallForwarding in FreeSwitch. I beleive I`m not the only guy tryiig to achieve this, what`s the Best Practices for this task? Regards, NG On Mon, Sep 7, 2009 at 2:54 AM, Nandy Dagondon wrote: > nik, > > please try the "legs" variable > http://www.nabble.com/CDR-accounting-question-td19212516.html > > /nandy > > > On Sun, Sep 6, 2009 at 6:40 PM, Nikolai Geordzhev wrote: > >> Hi, >> I`m trying to implement Call Forwarding in my FS setup. I set a user >> variable managing the type of forwarding (busy,no answer,unconditional) and >> the destination the phone is forwarded to: >> >> >> >> >> > wrote: That's what his sample does, but he says it crashes. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raffaele P. Guidi Sent: Tuesday, September 08, 2009 10:08 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET Hi, you just have to use delegates to asynchronously call the function containing the loop and return back the control to the calling thread. Here an example (don't have my code at hand, hope it doesn't contain typos). Regards, Raffaele public class TestLoop : ILoadNotificationPlugin { Delegate void DoStuffDelegate(); public void doStuff() { EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } } public bool Load() { DoStuffDelegate dsdlg = new DoStuffDelegate(doStuff); dsdlg.BeginInvoke(); } } On Tue, Sep 8, 2009 at 08:21, Josh Rivers > wrote: Thanks for the response! I have tried putting a long-running loop here, but then it blocks anything else managed from happening: public class TestLoop : ILoadNotificationPlugin { public bool Load() { EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } } } However, if I fork off a thread here, freeswitch crashes: public class TestLoop : ILoadNotificationPlugin { public bool Load() { ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } }); return true; } } It doesn't look like this is a good place to start a long-running process? Thanks! Josh On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi > wrote: Yes! public class LoadDemo : ILoadNotificationPlugin { public bool Load() { Log.WriteLine(LogLevel.Notice, "LoadDemo running."); return true; } } this example is from Michael Giagnocavo's Demo.csx which you can find into the mod_managed svn. And let me add that works like a charm :) Ciao, Raffaele On Sun, Sep 6, 2009 at 22:50, Josh Rivers > wrote: Is there a way to start this when FreeSWITCH starts? The lua and perl modules have a 'startup-script' configuration preference. Is there something similar in mod_managed? Or is there a way to have an api command executed at a startup? Exactly what I was after - thank you! On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk > wrote: > > try something like this > > EventConsumer con = new EventConsumer("all", ""); > Event ev = con.pop(0); > > see lua sample - > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer > > > Phillip Jones-2 wrote: > > > > Hi there, > > > > mod_managed exposes EventReceivedFunction such that: > > > > Session.EventReceivedFunction = (e) => > > { > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", e.ToString()); > > return ""; > > }; > > > > should trap all events to which i subscribe. > > > > > > But how do I subscribe to events? What is the .NET / managed equivalent > > of: > > > > switch_event_bind(const char *id, switch_event_types_t event, const char > > *subclass_name, switch_event_callback_t callback, void *user_data); > > > > > > > > Thank you! > > > > > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/deb1e6ef/attachment-0001.html From raffaele.p.guidi at gmail.com Tue Sep 8 14:20:49 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Tue, 8 Sep 2009 23:20:49 +0200 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4D2C@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4C0C@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4D2C@mse17be1.mse17.exchange.ms> Message-ID: Oh, I see... all those years wasted doing java stuff! :D On Tue, Sep 8, 2009 at 22:46, Michael Giagnocavo wrote: > ? ThreadPool.QueueUserWorkItem((o) =>? > > That starts a lambda, which is compiled to a delegate, same as anonymous > methods. > > Guess I?ll wait for him to respond on the crash he gets. > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Raffaele P. > Guidi > *Sent:* Tuesday, September 08, 2009 12:22 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > Well, I can't see any delegate in josh sample, just a > ThreadPool.QueueUserWorkItem. Here is an example that, at least on my > system (I reached my home pc in the meanwhile), works fine. > > > > public class LoadPluginDemo : ILoadNotificationPlugin { > > delegate void Listener(); > > private void EventListener() { > > EventConsumer con = new EventConsumer("all", null); > > while (true){ > > Event ev = con.pop(1); > > Log.WriteLine(LogLevel.Notice, "Got event " + > ev.GetHeader("Event-Name")); > > } > > } > > public bool Load() { > > Log.WriteLine(LogLevel.Notice, "LoadDemo running."); > > new Listener(EventListener).BeginInvoke(null,null); > > return true; > > } > > } > > > > > > > > On Tue, Sep 8, 2009 at 18:43, Michael Giagnocavo > wrote: > > That?s what his sample does, but he says it crashes. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Raffaele P. > Guidi > *Sent:* Tuesday, September 08, 2009 10:08 AM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > Hi, you just have to use delegates to asynchronously call the function > containing the loop and return back the control to the calling thread. Here > an example (don't have my code at hand, hope it doesn't contain typos). > > > > Regards, > > Raffaele > > > > public class TestLoop : ILoadNotificationPlugin > > { > > > > Delegate void DoStuffDelegate(); > > > > public void doStuff() > > { > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > } > > public bool Load() > > { > > DoStuffDelegate dsdlg = new DoStuffDelegate(doStuff); > > dsdlg.BeginInvoke(); > > } > > } > > On Tue, Sep 8, 2009 at 08:21, Josh Rivers wrote: > > Thanks for the response! > > > > I have tried putting a long-running loop here, but then it blocks anything > else managed from happening: > > > > public class TestLoop : ILoadNotificationPlugin > > { > > public bool Load() > > { > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > } > > } > > > > However, if I fork off a thread here, freeswitch crashes: > > public class TestLoop : ILoadNotificationPlugin > > { > > public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => > > { > > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > }); > > return true; > > } > > } > > > > It doesn't look like this is a good place to start a long-running process? > > > > Thanks! > > Josh > > > > On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi < > raffaele.p.guidi at gmail.com> wrote: > > Yes! > > > > public class LoadDemo : ILoadNotificationPlugin { > > public bool Load() { > > Log.WriteLine(LogLevel.Notice, "LoadDemo running."); > > return true; > > } > > } > > > > this example is from Michael Giagnocavo's Demo.csx which you can find into > the mod_managed svn. > > > > And let me add that works like a charm :) > > > > Ciao, > > Raffaele > > > > On Sun, Sep 6, 2009 at 22:50, Josh Rivers wrote: > > Is there a way to start this when FreeSWITCH starts? The lua and perl > modules have a 'startup-script' configuration preference. Is there something > similar in mod_managed? Or is there a way to have an api command executed at > a startup? > > > > > > Exactly what I was after - thank you! > > > > On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk wrote: > > > > > > > > try something like this > > > > > > EventConsumer con = new EventConsumer("all", ""); > > > Event ev = con.pop(0); > > > > > > see lua sample - > > > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer > > > > > > > > > Phillip Jones-2 wrote: > > > > > > > > Hi there, > > > > > > > > mod_managed exposes EventReceivedFunction such that: > > > > > > > > Session.EventReceivedFunction = (e) => > > > > { > > > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", > e.ToString()); > > > > return ""; > > > > }; > > > > > > > > should trap all events to which i subscribe. > > > > > > > > > > > > But how do I subscribe to events? What is the .NET / managed equivalent > > > > of: > > > > > > > > switch_event_bind(const char *id, switch_event_types_t event, const > char > > > > *subclass_name, switch_event_callback_t callback, void *user_data); > > > > > > > > > > > > > > > > Thank you! > > > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/18bc8736/attachment-0001.html From tina at a2unlimited.com Tue Sep 8 14:27:52 2009 From: tina at a2unlimited.com (Tina Martinez) Date: Tue, 08 Sep 2009 17:27:52 -0400 Subject: [Freeswitch-users] Custom Variables Message-ID: <40292.1252445272@a2unlimited.com> Using the verbose-events definitely improved my ability to see the custom variables, but now I noticed that the "Member-ID" variable does not appear in the DTMF event. Would this be related, or did I screw something else up? - T From pjintheusa at gmail.com Tue Sep 8 14:50:27 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Tue, 8 Sep 2009 17:50:27 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4C0C@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4D2C@mse17be1.mse17.exchange.ms> Message-ID: <367751820909081450n4e6faae7h14b4f22d28d1a6d9@mail.gmail.com> I build this out. This seems to work fine: ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); EventConsumer con = new EventConsumer("all", ""); while (true) { * Event ev = con.pop(1);* Log.WriteLine(LogLevel.Alert, "Event: " + ev.GetHeader("Event-Name")); //freeswitch.msleep(100); } }); With Event ev = con.pop(0) however FS crashes with a System.NullReferenceException (attached) On Tue, Sep 8, 2009 at 5:20 PM, Raffaele P. Guidi < raffaele.p.guidi at gmail.com> wrote: > Oh, I see... all those years wasted doing java stuff! :D > > > On Tue, Sep 8, 2009 at 22:46, Michael Giagnocavo wrote: > >> ? ThreadPool.QueueUserWorkItem((o) =>? >> >> That starts a lambda, which is compiled to a delegate, same as anonymous >> methods. >> >> Guess I?ll wait for him to respond on the crash he gets. >> >> >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Raffaele P. >> Guidi >> *Sent:* Tuesday, September 08, 2009 12:22 PM >> >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / >> .NET >> >> >> >> Well, I can't see any delegate in josh sample, just a >> ThreadPool.QueueUserWorkItem. Here is an example that, at least on my >> system (I reached my home pc in the meanwhile), works fine. >> >> >> >> public class LoadPluginDemo : ILoadNotificationPlugin { >> >> delegate void Listener(); >> >> private void EventListener() { >> >> EventConsumer con = new EventConsumer("all", null); >> >> while (true){ >> >> Event ev = con.pop(1); >> >> Log.WriteLine(LogLevel.Notice, "Got event " + >> ev.GetHeader("Event-Name")); >> >> } >> >> } >> >> public bool Load() { >> >> Log.WriteLine(LogLevel.Notice, "LoadDemo running."); >> >> new Listener(EventListener).BeginInvoke(null,null); >> >> return true; >> >> } >> >> } >> >> >> >> >> >> >> >> On Tue, Sep 8, 2009 at 18:43, Michael Giagnocavo >> wrote: >> >> That?s what his sample does, but he says it crashes. >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Raffaele P. >> Guidi >> *Sent:* Tuesday, September 08, 2009 10:08 AM >> >> >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / >> .NET >> >> >> >> Hi, you just have to use delegates to asynchronously call the function >> containing the loop and return back the control to the calling thread. Here >> an example (don't have my code at hand, hope it doesn't contain typos). >> >> >> >> Regards, >> >> Raffaele >> >> >> >> public class TestLoop : ILoadNotificationPlugin >> >> { >> >> >> >> Delegate void DoStuffDelegate(); >> >> >> >> public void doStuff() >> >> { >> >> EventConsumer con = new EventConsumer("all", ""); >> >> while (true) >> >> { >> >> Event ev = con.pop(0); >> >> Log.WriteLine(LogLevel.Notice, "Event: " + >> ev.serialized_string); >> >> freeswitch.msleep(100); >> >> } >> >> } >> >> public bool Load() >> >> { >> >> DoStuffDelegate dsdlg = new DoStuffDelegate(doStuff); >> >> dsdlg.BeginInvoke(); >> >> } >> >> } >> >> On Tue, Sep 8, 2009 at 08:21, Josh Rivers wrote: >> >> Thanks for the response! >> >> >> >> I have tried putting a long-running loop here, but then it blocks anything >> else managed from happening: >> >> >> >> public class TestLoop : ILoadNotificationPlugin >> >> { >> >> public bool Load() >> >> { >> >> EventConsumer con = new EventConsumer("all", ""); >> >> while (true) >> >> { >> >> Event ev = con.pop(0); >> >> Log.WriteLine(LogLevel.Notice, "Event: " + >> ev.serialized_string); >> >> freeswitch.msleep(100); >> >> } >> >> } >> >> } >> >> >> >> However, if I fork off a thread here, freeswitch crashes: >> >> public class TestLoop : ILoadNotificationPlugin >> >> { >> >> public bool Load() >> >> { >> >> ThreadPool.QueueUserWorkItem((o) => >> >> { >> >> Log.WriteLine(LogLevel.Notice, "Thread Starting. "); >> >> EventConsumer con = new EventConsumer("all", ""); >> >> while (true) >> >> { >> >> Event ev = con.pop(0); >> >> Log.WriteLine(LogLevel.Notice, "Event: " + >> ev.serialized_string); >> >> freeswitch.msleep(100); >> >> } >> >> }); >> >> return true; >> >> } >> >> } >> >> >> >> It doesn't look like this is a good place to start a long-running process? >> >> >> >> Thanks! >> >> Josh >> >> >> >> On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi < >> raffaele.p.guidi at gmail.com> wrote: >> >> Yes! >> >> >> >> public class LoadDemo : ILoadNotificationPlugin { >> >> public bool Load() { >> >> Log.WriteLine(LogLevel.Notice, "LoadDemo running."); >> >> return true; >> >> } >> >> } >> >> >> >> this example is from Michael Giagnocavo's Demo.csx which you can find into >> the mod_managed svn. >> >> >> >> And let me add that works like a charm :) >> >> >> >> Ciao, >> >> Raffaele >> >> >> >> On Sun, Sep 6, 2009 at 22:50, Josh Rivers wrote: >> >> Is there a way to start this when FreeSWITCH starts? The lua and perl >> modules have a 'startup-script' configuration preference. Is there something >> similar in mod_managed? Or is there a way to have an api command executed at >> a startup? >> >> >> >> >> >> Exactly what I was after - thank you! >> >> >> >> On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk wrote: >> >> >> >> > >> >> > try something like this >> >> > >> >> > EventConsumer con = new EventConsumer("all", ""); >> >> > Event ev = con.pop(0); >> >> > >> >> > see lua sample - >> >> > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer >> >> > >> >> > >> >> > Phillip Jones-2 wrote: >> >> > > >> >> > > Hi there, >> >> > > >> >> > > mod_managed exposes EventReceivedFunction such that: >> >> > > >> >> > > Session.EventReceivedFunction = (e) => >> >> > > { >> >> > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", >> e.ToString()); >> >> > > return ""; >> >> > > }; >> >> > > >> >> > > should trap all events to which i subscribe. >> >> > > >> >> > > >> >> > > But how do I subscribe to events? What is the .NET / managed >> equivalent >> >> > > of: >> >> > > >> >> > > switch_event_bind(const char *id, switch_event_types_t event, const >> char >> >> > > *subclass_name, switch_event_callback_t callback, void *user_data); >> >> > > >> >> > > >> >> > > >> >> > > Thank you! >> >> > > >> >> > > >> >> > > >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/1fd64de9/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: FS_Crash.jpg Type: image/jpeg Size: 147452 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/1fd64de9/attachment-0001.jpg From dujinfang at gmail.com Tue Sep 8 15:57:16 2009 From: dujinfang at gmail.com (Seven Du) Date: Wed, 9 Sep 2009 06:57:16 +0800 Subject: [Freeswitch-users] Recording Only 1 Leg of a Call In-Reply-To: <191c3a030909080944o3fa6f386ua42c56e3535ffd69@mail.gmail.com> References: <4256bf830909070826o5cb0d7a4lcd27e07ee350b50d@mail.gmail.com> <4256bf830909072150i73ceca68we9476e2a57b53f6a@mail.gmail.com> <191c3a030909080944o3fa6f386ua42c56e3535ffd69@mail.gmail.com> Message-ID: <91146DC4-9BA6-48B9-A4EC-7C95AA7FE98B@gmail.com> As a work around, record to stereo, and use sox to split channes ? On Sep 9, 2009, at 12:44 AM, Anthony Minessale wrote: > that would have to be filed as a feature request as we do not > currently have a way to do that. > > > On Mon, Sep 7, 2009 at 11:50 PM, Matthew Fong > wrote: > I want to record without the telephone user's interaction. > > I think uuid_record should have the option to only record the audio > of the uuid channel that is being specified, and as a secondary > option combine the audio of the b leg (since uuid_record only > specifies one uuid anyway--this seems logical). > > Anyway, just my wish list :) > > --matt > http://www.hellohunter.com > voice broadcasting & hosted dialer > > On Tue, Sep 8, 2009 at 2:12 AM, Milena wrote: > Hello, > What about this?: > > " > > " > > the person would have to press *2 during the call to start the > recording. > > 2009/9/7 Matthew Fong > Whats the best way to record only one leg of a call? > > uuid_record records both channels > session_record does the same (but has a stereo option) > > is there any way to record only an a-leg of the call? Thanks so much. > > --matt > http://www.hellohunter.com > hosted dialer & voice broadcasting > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mgg at giagnocavo.net Tue Sep 8 19:06:54 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 8 Sep 2009 22:06:54 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <367751820909081450n4e6faae7h14b4f22d28d1a6d9@mail.gmail.com> References: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4C0C@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4D2C@mse17be1.mse17.exchange.ms> <367751820909081450n4e6faae7h14b4f22d28d1a6d9@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4DC9@mse17be1.mse17.exchange.ms> I'm not sure how EventConsumer is supposed to work - maybe one of the real devs can explain how pop works and if it should fail on pop 0 or not. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Tuesday, September 08, 2009 3:50 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET I build this out. This seems to work fine: ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(1); Log.WriteLine(LogLevel.Alert, "Event: " + ev.GetHeader("Event-Name")); //freeswitch.msleep(100); } }); With Event ev = con.pop(0) however FS crashes with a System.NullReferenceException (attached) On Tue, Sep 8, 2009 at 5:20 PM, Raffaele P. Guidi > wrote: Oh, I see... all those years wasted doing java stuff! :D On Tue, Sep 8, 2009 at 22:46, Michael Giagnocavo > wrote: " ThreadPool.QueueUserWorkItem((o) =>" That starts a lambda, which is compiled to a delegate, same as anonymous methods. Guess I'll wait for him to respond on the crash he gets. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raffaele P. Guidi Sent: Tuesday, September 08, 2009 12:22 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET Well, I can't see any delegate in josh sample, just a ThreadPool.QueueUserWorkItem. Here is an example that, at least on my system (I reached my home pc in the meanwhile), works fine. public class LoadPluginDemo : ILoadNotificationPlugin { delegate void Listener(); private void EventListener() { EventConsumer con = new EventConsumer("all", null); while (true){ Event ev = con.pop(1); Log.WriteLine(LogLevel.Notice, "Got event " + ev.GetHeader("Event-Name")); } } public bool Load() { Log.WriteLine(LogLevel.Notice, "LoadDemo running."); new Listener(EventListener).BeginInvoke(null,null); return true; } } On Tue, Sep 8, 2009 at 18:43, Michael Giagnocavo > wrote: That's what his sample does, but he says it crashes. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raffaele P. Guidi Sent: Tuesday, September 08, 2009 10:08 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET Hi, you just have to use delegates to asynchronously call the function containing the loop and return back the control to the calling thread. Here an example (don't have my code at hand, hope it doesn't contain typos). Regards, Raffaele public class TestLoop : ILoadNotificationPlugin { Delegate void DoStuffDelegate(); public void doStuff() { EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } } public bool Load() { DoStuffDelegate dsdlg = new DoStuffDelegate(doStuff); dsdlg.BeginInvoke(); } } On Tue, Sep 8, 2009 at 08:21, Josh Rivers > wrote: Thanks for the response! I have tried putting a long-running loop here, but then it blocks anything else managed from happening: public class TestLoop : ILoadNotificationPlugin { public bool Load() { EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } } } However, if I fork off a thread here, freeswitch crashes: public class TestLoop : ILoadNotificationPlugin { public bool Load() { ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); freeswitch.msleep(100); } }); return true; } } It doesn't look like this is a good place to start a long-running process? Thanks! Josh On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi > wrote: Yes! public class LoadDemo : ILoadNotificationPlugin { public bool Load() { Log.WriteLine(LogLevel.Notice, "LoadDemo running."); return true; } } this example is from Michael Giagnocavo's Demo.csx which you can find into the mod_managed svn. And let me add that works like a charm :) Ciao, Raffaele On Sun, Sep 6, 2009 at 22:50, Josh Rivers > wrote: Is there a way to start this when FreeSWITCH starts? The lua and perl modules have a 'startup-script' configuration preference. Is there something similar in mod_managed? Or is there a way to have an api command executed at a startup? Exactly what I was after - thank you! On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk > wrote: > > try something like this > > EventConsumer con = new EventConsumer("all", ""); > Event ev = con.pop(0); > > see lua sample - > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer > > > Phillip Jones-2 wrote: > > > > Hi there, > > > > mod_managed exposes EventReceivedFunction such that: > > > > Session.EventReceivedFunction = (e) => > > { > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", e.ToString()); > > return ""; > > }; > > > > should trap all events to which i subscribe. > > > > > > But how do I subscribe to events? What is the .NET / managed equivalent > > of: > > > > switch_event_bind(const char *id, switch_event_types_t event, const char > > *subclass_name, switch_event_callback_t callback, void *user_data); > > > > > > > > Thank you! > > > > > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/0fcf462d/attachment-0001.html From rogelio.perez at gmail.com Tue Sep 8 19:15:05 2009 From: rogelio.perez at gmail.com (Rogelio Perez) Date: Tue, 8 Sep 2009 23:15:05 -0300 Subject: [Freeswitch-users] Mod_nibblebill for CDR billing In-Reply-To: <86a32abc0909041828n1fc4b55es47898f408c5fc0d7@mail.gmail.com> References: <7F5518A9-FF8A-4E94-9E4A-5FE2CC6D8F50@gmail.com> <86a32abc0909041828n1fc4b55es47898f408c5fc0d7@mail.gmail.com> Message-ID: <662810D9-E7C2-48A1-AED3-15D095D750B3@gmail.com> I love FS! it shows all the info I need. Thanks guys. On Sep 4, 2009, at 10:28 PM, Diego Viola wrote: > If you do "event plain all" from the FS CLI you should see the > variable exported on the CHANNEL_HANGUP_COMPLETE event, with the > other CDR variables as well. These information should be available > on mod_xml_cdr and mod_cdr_csv as well. > > Diego > > On Fri, Sep 4, 2009 at 11:28 PM, Rogelio Perez > wrote: > From the mod_nibblebill documentation: > > At the end of a call, the module sets a variable named > nibble_total_billed. You can use mod_cdr to record this variable to > your CDR log. > > Is it possible to do the same with mod_xml_cdr? > I'm looking for a simple way of billing my CDRs and this one looks > like a good solution. > Has anyone tried doing anything similar? > > Thanks, > Rogelio > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/5ca17cbe/attachment.html From nagalenoj at gmail.com Tue Sep 8 20:42:12 2009 From: nagalenoj at gmail.com (Nagalenoj) Date: Tue, 8 Sep 2009 20:42:12 -0700 (PDT) Subject: [Freeswitch-users] ESL: DTMF event is not coming In-Reply-To: <191c3a030909081232g5550542fq4bde3a21c33b6a52@mail.gmail.com> References: <25326328.post@talk.nabble.com> <191c3a030909081232g5550542fq4bde3a21c33b6a52@mail.gmail.com> Message-ID: <25358074.post@talk.nabble.com> Yes.!! I missed "async" keyword in dialplan in freeswitch-1.0.4. Thanks.. Anthony Minessale-2 wrote: > > did you specify the "async" keyword to the socket app in your dialplan? > > On Mon, Sep 7, 2009 at 2:47 AM, Nagalenoj wrote: > >> >> Dear friends, >> I am using freeswitch-1.0.4. When I execute the sample >> script(/libs/esl/perl/server2.pl), it is not receiving the DTMF events. >> When >> I execute the same program in freeswitch-1.0.3, it's receiving the event. >> Do >> I miss something to configure/upgrade.? >> -- >> View this message in context: >> http://www.nabble.com/ESL%3A-DTMF-event-is-not-coming-tp25326328p25326328.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/ESL%3A-DTMF-event-is-not-coming-tp25326328p25358074.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From rogelio.perez at gmail.com Tue Sep 8 20:59:43 2009 From: rogelio.perez at gmail.com (Rogelio Perez) Date: Wed, 9 Sep 2009 00:59:43 -0300 Subject: [Freeswitch-users] mod_opal segmentation fault error Message-ID: <5BC8E502-1F0D-4490-A01A-A2251E3F4C36@gmail.com> Hi guys, My FS setup was working smoothly with mod_opal enabled until I had to rebuild everything from scratch. Now I have compiled everything following the same procedure (I even have a script for that) and mod_opal stopped working. The SVN commands are: 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 ...and the compilation commands follow the documentation isntructions and there are no output errors. I start FS with mod_opal disabled and then when I run "load mod_opal"I get the error: Segmentation fault (core dumped). The log output shows nothing, and I see there are core.xxxxx files on the FS directory but I dont know how to read them. Any ideas? Thanks, Rogelio From gkuri at ieee.org Tue Sep 8 21:26:41 2009 From: gkuri at ieee.org (Gabriel Kuri) Date: Tue, 08 Sep 2009 21:26:41 -0700 Subject: [Freeswitch-users] nibblebill zero balance Message-ID: <4AA72E81.20106@ieee.org> We've been testing mod_nibblebill, it's a great module, cheers to the author! A couple questions regarding nibblebill: 1) We noticed that when an account is below the minimum balance and a call is attempted with that account, FS begins to connect the B leg for the call but then cancels the INVITE. Why is FS sending the INVITE for the B leg only to CANCEL it immediately, if the account is below the minimum balance? I would think it would just send the 404 immediately back to the user without sending any INVITEs to establish the B leg? 2) Any progress on implementing minimum billing increments (ie 60/60) ? Cheers, Gabe From josh at radianttiger.com Tue Sep 8 22:25:50 2009 From: josh at radianttiger.com (Josh Rivers) Date: Tue, 8 Sep 2009 22:25:50 -0700 Subject: [Freeswitch-users] Using mod_managed to create full FreeSWITCH modules In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B39@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B39@mse17be1.mse17.exchange.ms> Message-ID: Thanks. I can stop trying to figure out how to do that then. -Josh On Tue, Sep 8, 2009 at 5:12 AM, Michael Giagnocavo wrote: > Are you looking to run when mod_managed shuts down? Or when your managed > plugin reloads, or something else? (mod_managed is not unloadable, so I > don?t believe it gets any notification of shutting down.) > > > > As far as interop in general, it?s usually possible. However, a lot of the > FreeSWITCH code uses macros, and they aren?t available via SWIG. So in those > cases, you?ll either need to manually reconstruct the macro, or write some > interop code in C/C++ to do what you want, then expose that via SWIG (or, if > you do it nicely, just P/Invoke it directly). > > > > Some of the native code generates some pretty ugly structures; you will > probably need to become friends with the Marshal class and pass around a lot > of IntPtrs to get things going. > > > > As far as I know, no one has built a non API/App with mod_managed. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers > *Sent:* Monday, September 07, 2009 11:41 PM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Using mod_managed to create full FreeSWITCH > modules > > > > The wiki says: > > mod_managed exposes nearly the entire FreeSWITCH C API (courtesy of SWIG). > This allows creation of not just API functions and call apps, but any type > of module that FreeSWITCH supports (codecs, endpoints, etc.). The types are > in the FreeSWITCH.Native namespace. FreeSWITCH.Native. The > FreeSWITCH.Native.freeswitch type contains static members to access all the > functions. > > > > Does anybody have a starting point they can share for a non-API/APP managed > module. I'd like to build something that runs in > the SWITCH_MODULE_SHUTDOWN_FUNCTION/SWITCH_MODULE_SHUTDOWN_FUNCTION/SWITCH_MODULE_SHUTDOWN_FUNCTION > lifecycle. How can this be done? > > > > Thanks! > > Josh > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/2954c47a/attachment.html From josh at radianttiger.com Tue Sep 8 22:30:00 2009 From: josh at radianttiger.com (Josh Rivers) Date: Tue, 8 Sep 2009 22:30:00 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> Message-ID: Here is the error I get with the loop I mentioned. -Josh [image: Capture.PNG] On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo wrote: > Hi, > > > > Can you please elaborate on the crash you receive when you > queue a thread during load? > > > > Thanks, > > Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers > *Sent:* Tuesday, September 08, 2009 12:22 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > Thanks for the response! > > > > I have tried putting a long-running loop here, but then it blocks anything > else managed from happening: > > > > public class TestLoop : ILoadNotificationPlugin > > { > > public bool Load() > > { > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > } > > } > > > > However, if I fork off a thread here, freeswitch crashes: > > public class TestLoop : ILoadNotificationPlugin > > { > > public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => > > { > > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > }); > > return true; > > } > > } > > > > It doesn't look like this is a good place to start a long-running process? > > > > Thanks! > > Josh > > > > On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi < > raffaele.p.guidi at gmail.com> wrote: > > Yes! > > > > public class LoadDemo : ILoadNotificationPlugin { > > public bool Load() { > > Log.WriteLine(LogLevel.Notice, "LoadDemo running."); > > return true; > > } > > } > > > > this example is from Michael Giagnocavo's Demo.csx which you can find into > the mod_managed svn. > > > > And let me add that works like a charm :) > > > > Ciao, > > Raffaele > > > > On Sun, Sep 6, 2009 at 22:50, Josh Rivers wrote: > > Is there a way to start this when FreeSWITCH starts? The lua and perl > modules have a 'startup-script' configuration preference. Is there something > similar in mod_managed? Or is there a way to have an api command executed at > a startup? > > > > > > Exactly what I was after - thank you! > > > > On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk wrote: > > > > > > > > try something like this > > > > > > EventConsumer con = new EventConsumer("all", ""); > > > Event ev = con.pop(0); > > > > > > see lua sample - > > > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer > > > > > > > > > Phillip Jones-2 wrote: > > > > > > > > Hi there, > > > > > > > > mod_managed exposes EventReceivedFunction such that: > > > > > > > > Session.EventReceivedFunction = (e) => > > > > { > > > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", > e.ToString()); > > > > return ""; > > > > }; > > > > > > > > should trap all events to which i subscribe. > > > > > > > > > > > > But how do I subscribe to events? What is the .NET / managed equivalent > > > > of: > > > > > > > > switch_event_bind(const char *id, switch_event_types_t event, const > char > > > > *subclass_name, switch_event_callback_t callback, void *user_data); > > > > > > > > > > > > > > > > Thank you! > > > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 8873 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/4ef5064e/attachment-0001.png From josh at radianttiger.com Tue Sep 8 22:32:16 2009 From: josh at radianttiger.com (Josh Rivers) Date: Tue, 8 Sep 2009 22:32:16 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <367751820909081112t759bed1eve65ba8feab0ee8c4@mail.gmail.com> References: <367751820909081112t759bed1eve65ba8feab0ee8c4@mail.gmail.com> Message-ID: You are probably right, but commenting out the msleep doesn't prevent the crash. -Josh On Tue, Sep 8, 2009 at 11:12 AM, Phillip Jones wrote: > What is: > > freeswitch.msleep(100); > > Why aren't you using Thread.Sleep? > > > On Tue, Sep 8, 2009 at 2:21 AM, Josh Rivers wrote: > >> Thanks for the response! >> I have tried putting a long-running loop here, but then it blocks anything >> else managed from happening: >> >> public class TestLoop : ILoadNotificationPlugin >> { >> public bool Load() >> { >> EventConsumer con = new EventConsumer("all", ""); >> while (true) >> { >> Event ev = con.pop(0); >> Log.WriteLine(LogLevel.Notice, "Event: " + >> ev.serialized_string); >> freeswitch.msleep(100); >> } >> } >> } >> >> However, if I fork off a thread here, freeswitch crashes: >> public class TestLoop : ILoadNotificationPlugin >> { >> public bool Load() >> { >> ThreadPool.QueueUserWorkItem((o) => >> { >> Log.WriteLine(LogLevel.Notice, "Thread Starting. "); >> EventConsumer con = new EventConsumer("all", ""); >> while (true) >> { >> Event ev = con.pop(0); >> Log.WriteLine(LogLevel.Notice, "Event: " + >> ev.serialized_string); >> freeswitch.msleep(100); >> } >> }); >> return true; >> } >> } >> >> It doesn't look like this is a good place to start a long-running process? >> >> Thanks! >> Josh >> >> On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi < >> raffaele.p.guidi at gmail.com> wrote: >> >>> Yes! >>> public class LoadDemo : ILoadNotificationPlugin { >>> public bool Load() { >>> Log.WriteLine(LogLevel.Notice, "LoadDemo running."); >>> return true; >>> } >>> } >>> >>> this example is from Michael Giagnocavo's Demo.csx which you can find >>> into the mod_managed svn. >>> >>> And let me add that works like a charm :) >>> >>> Ciao, >>> Raffaele >>> >>> On Sun, Sep 6, 2009 at 22:50, Josh Rivers wrote: >>> >>>> Is there a way to start this when FreeSWITCH starts? The lua and perl >>>> modules have a 'startup-script' configuration preference. Is there something >>>> similar in mod_managed? Or is there a way to have an api command executed at >>>> a startup? >>>> >>>> >>>> Exactly what I was after - thank you! >>>> >>>> On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk >>>> wrote: >>>> >>>> > >>>> > try something like this >>>> > >>>> > EventConsumer con = new EventConsumer("all", ""); >>>> > Event ev = con.pop(0); >>>> > >>>> > see lua sample - >>>> > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer >>>> > >>>> > >>>> > Phillip Jones-2 wrote: >>>> > > >>>> > > Hi there, >>>> > > >>>> > > mod_managed exposes EventReceivedFunction such that: >>>> > > >>>> > > Session.EventReceivedFunction = (e) => >>>> > > { >>>> > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", >>>> e.ToString()); >>>> > > return ""; >>>> > > }; >>>> > > >>>> > > should trap all events to which i subscribe. >>>> > > >>>> > > >>>> > > But how do I subscribe to events? What is the .NET / managed >>>> equivalent >>>> > > of: >>>> > > >>>> > > switch_event_bind(const char *id, switch_event_types_t event, const >>>> char >>>> > > *subclass_name, switch_event_callback_t callback, void *user_data); >>>> > > >>>> > > >>>> > > >>>> > > Thank you! >>>> > > >>>> > > >>>> > > >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/1c35a125/attachment.html From mrene_lists at avgs.ca Tue Sep 8 22:33:56 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 8 Sep 2009 22:33:56 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> Message-ID: <63A023F4-1AA8-4A56-8029-91F5A5C74084@avgs.ca> Click Break, then go in Window, Debug, Stack Trace (or something similar, I don't have any VS nearby), then copy paste that. Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 8-Sep-09, at 10:30 PM, Josh Rivers wrote: > Here is the error I get with the loop I mentioned. -Josh > > > On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo > wrote: > Hi, > > > Can you please elaborate on the crash you receive > when you queue a thread during load? > > > Thanks, > > Michael > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Josh Rivers > Sent: Tuesday, September 08, 2009 12:22 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Subscribing to events in managed > C# / .NET > > > Thanks for the response! > > > I have tried putting a long-running loop here, but then it blocks > anything else managed from happening: > > > public class TestLoop : ILoadNotificationPlugin > > { > > public bool Load() > > { > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > } > > } > > > However, if I fork off a thread here, freeswitch crashes: > > public class TestLoop : ILoadNotificationPlugin > > { > > public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => > > { > > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > }); > > return true; > > } > > } > > > It doesn't look like this is a good place to start a long-running > process? > > > Thanks! > > Josh > > > On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi > wrote: > > Yes! > > > public class LoadDemo : ILoadNotificationPlugin { > > public bool Load() { > > Log.WriteLine(LogLevel.Notice, "LoadDemo running."); > > return true; > > } > > } > > > this example is from Michael Giagnocavo's Demo.csx which you can > find into the mod_managed svn. > > > And let me add that works like a charm :) > > > Ciao, > > Raffaele > > > On Sun, Sep 6, 2009 at 22:50, Josh Rivers > wrote: > > Is there a way to start this when FreeSWITCH starts? The lua and > perl modules have a 'startup-script' configuration preference. Is > there something similar in mod_managed? Or is there a way to have an > api command executed at a startup? > > > > > Exactly what I was after - thank you! > > > On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk > wrote: > > > > > > > try something like this > > > > > > EventConsumer con = new EventConsumer("all", ""); > > > Event ev = con.pop(0); > > > > > > see lua sample - > > > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer > > > > > > > > > Phillip Jones-2 wrote: > > > > > > > > Hi there, > > > > > > > > mod_managed exposes EventReceivedFunction such that: > > > > > > > > Session.EventReceivedFunction = (e) => > > > > { > > > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", > e.ToString()); > > > > return ""; > > > > }; > > > > > > > > should trap all events to which i subscribe. > > > > > > > > > > > > But how do I subscribe to events? What is the .NET / managed > equivalent > > > > of: > > > > > > > > switch_event_bind(const char *id, switch_event_types_t event, > const char > > > > *subclass_name, switch_event_callback_t callback, void > *user_data); > > > > > > > > > > > > > > > > Thank you! > > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/07dcc10a/attachment-0001.html From josh at radianttiger.com Tue Sep 8 22:46:48 2009 From: josh at radianttiger.com (Josh Rivers) Date: Tue, 8 Sep 2009 22:46:48 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4DC9@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4C0C@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4D2C@mse17be1.mse17.exchange.ms> <367751820909081450n4e6faae7h14b4f22d28d1a6d9@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4DC9@mse17be1.mse17.exchange.ms> Message-ID: I confirmed that pop(1) works for me. since it's spinning in it's own thread, that's probably the right solution. These are the relevant pieces of core code, though: APU_DECLARE(apr_status_t) apr_queue_trypop(apr_queue_t *queue, void **data) { apr_status_t rv; if (queue->terminated) { return APR_EOF; /* no more elements ever again */ } rv = apr_thread_mutex_lock(queue->one_big_mutex); if (rv != APR_SUCCESS) { return rv; } if (apr_queue_empty(queue)) { rv = apr_thread_mutex_unlock(queue->one_big_mutex); return APR_EAGAIN; } *data = queue->data[queue->out]; queue->nelts--; queue->out = (queue->out + 1) % queue->bounds; if (queue->full_waiters) { Q_DBG("signal !full", queue); rv = apr_thread_cond_signal(queue->not_full); if (rv != APR_SUCCESS) { apr_thread_mutex_unlock(queue->one_big_mutex); return rv; } } rv = apr_thread_mutex_unlock(queue->one_big_mutex); return rv; } AND APU_DECLARE(apr_status_t) apr_queue_pop(apr_queue_t *queue, void **data) { apr_status_t rv; if (queue->terminated) { return APR_EOF; /* no more elements ever again */ } rv = apr_thread_mutex_lock(queue->one_big_mutex); if (rv != APR_SUCCESS) { return rv; } /* Keep waiting until we wake up and find that the queue is not empty. */ if (apr_queue_empty(queue)) { if (!queue->terminated) { queue->empty_waiters++; rv = apr_thread_cond_wait(queue->not_empty, queue->one_big_mutex); queue->empty_waiters--; if (rv != APR_SUCCESS) { apr_thread_mutex_unlock(queue->one_big_mutex); return rv; } } /* If we wake up and it's still empty, then we were interrupted */ if (apr_queue_empty(queue)) { Q_DBG("queue empty (intr)", queue); rv = apr_thread_mutex_unlock(queue->one_big_mutex); if (rv != APR_SUCCESS) { return rv; } if (queue->terminated) { return APR_EOF; /* no more elements ever again */ } else { return APR_EINTR; } } } *data = queue->data[queue->out]; queue->nelts--; queue->out = (queue->out + 1) % queue->bounds; if (queue->full_waiters) { Q_DBG("signal !full", queue); rv = apr_thread_cond_signal(queue->not_full); if (rv != APR_SUCCESS) { apr_thread_mutex_unlock(queue->one_big_mutex); return rv; } } rv = apr_thread_mutex_unlock(queue->one_big_mutex); return rv; } Is the trypop method returning without unlocking the mutex? rv = apr_thread_mutex_lock(queue->one_big_mutex); if (rv != APR_SUCCESS) { return rv; } On Tue, Sep 8, 2009 at 7:06 PM, Michael Giagnocavo wrote: > I?m not sure how EventConsumer is supposed to work ? maybe one of the > real devs can explain how pop works and if it should fail on pop 0 or not. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Phillip > Jones > *Sent:* Tuesday, September 08, 2009 3:50 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > I build this out. > > This seems to work fine: > > ThreadPool.QueueUserWorkItem((o) => > { > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > EventConsumer con = new EventConsumer("all", ""); > while (true) > { > * Event ev = con.pop(1);* > Log.WriteLine(LogLevel.Alert, "Event: " + > ev.GetHeader("Event-Name")); > //freeswitch.msleep(100); > } > }); > > With > Event ev = con.pop(0) however FS crashes with a > System.NullReferenceException (attached) > > > > > > > > > > On Tue, Sep 8, 2009 at 5:20 PM, Raffaele P. Guidi < > raffaele.p.guidi at gmail.com> wrote: > > Oh, I see... all those years wasted doing java stuff! :D > > > > On Tue, Sep 8, 2009 at 22:46, Michael Giagnocavo > wrote: > > ? ThreadPool.QueueUserWorkItem((o) =>? > > That starts a lambda, which is compiled to a delegate, same as anonymous > methods. > > Guess I?ll wait for him to respond on the crash he gets. > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Raffaele P. > Guidi > *Sent:* Tuesday, September 08, 2009 12:22 PM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > Well, I can't see any delegate in josh sample, just a > ThreadPool.QueueUserWorkItem. Here is an example that, at least on my > system (I reached my home pc in the meanwhile), works fine. > > > > public class LoadPluginDemo : ILoadNotificationPlugin { > > delegate void Listener(); > > private void EventListener() { > > EventConsumer con = new EventConsumer("all", null); > > while (true){ > > Event ev = con.pop(1); > > Log.WriteLine(LogLevel.Notice, "Got event " + > ev.GetHeader("Event-Name")); > > } > > } > > public bool Load() { > > Log.WriteLine(LogLevel.Notice, "LoadDemo running."); > > new Listener(EventListener).BeginInvoke(null,null); > > return true; > > } > > } > > > > > > > > On Tue, Sep 8, 2009 at 18:43, Michael Giagnocavo > wrote: > > That?s what his sample does, but he says it crashes. > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Raffaele P. > Guidi > *Sent:* Tuesday, September 08, 2009 10:08 AM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > Hi, you just have to use delegates to asynchronously call the function > containing the loop and return back the control to the calling thread. Here > an example (don't have my code at hand, hope it doesn't contain typos). > > > > Regards, > > Raffaele > > > > public class TestLoop : ILoadNotificationPlugin > > { > > > > Delegate void DoStuffDelegate(); > > > > public void doStuff() > > { > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > } > > public bool Load() > > { > > DoStuffDelegate dsdlg = new DoStuffDelegate(doStuff); > > dsdlg.BeginInvoke(); > > } > > } > > On Tue, Sep 8, 2009 at 08:21, Josh Rivers wrote: > > Thanks for the response! > > > > I have tried putting a long-running loop here, but then it blocks anything > else managed from happening: > > > > public class TestLoop : ILoadNotificationPlugin > > { > > public bool Load() > > { > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > } > > } > > > > However, if I fork off a thread here, freeswitch crashes: > > public class TestLoop : ILoadNotificationPlugin > > { > > public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => > > { > > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > }); > > return true; > > } > > } > > > > It doesn't look like this is a good place to start a long-running process? > > > > Thanks! > > Josh > > > > On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi < > raffaele.p.guidi at gmail.com> wrote: > > Yes! > > > > public class LoadDemo : ILoadNotificationPlugin { > > public bool Load() { > > Log.WriteLine(LogLevel.Notice, "LoadDemo running."); > > return true; > > } > > } > > > > this example is from Michael Giagnocavo's Demo.csx which you can find into > the mod_managed svn. > > > > And let me add that works like a charm :) > > > > Ciao, > > Raffaele > > > > On Sun, Sep 6, 2009 at 22:50, Josh Rivers wrote: > > Is there a way to start this when FreeSWITCH starts? The lua and perl > modules have a 'startup-script' configuration preference. Is there something > similar in mod_managed? Or is there a way to have an api command executed at > a startup? > > > > > > Exactly what I was after - thank you! > > > > On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk wrote: > > > > > > > > try something like this > > > > > > EventConsumer con = new EventConsumer("all", ""); > > > Event ev = con.pop(0); > > > > > > see lua sample - > > > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer > > > > > > > > > Phillip Jones-2 wrote: > > > > > > > > Hi there, > > > > > > > > mod_managed exposes EventReceivedFunction such that: > > > > > > > > Session.EventReceivedFunction = (e) => > > > > { > > > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", > e.ToString()); > > > > return ""; > > > > }; > > > > > > > > should trap all events to which i subscribe. > > > > > > > > > > > > But how do I subscribe to events? What is the .NET / managed equivalent > > > > of: > > > > > > > > switch_event_bind(const char *id, switch_event_types_t event, const > char > > > > *subclass_name, switch_event_callback_t callback, void *user_data); > > > > > > > > > > > > > > > > Thank you! > > > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/f560d363/attachment-0001.html From josh at radianttiger.com Tue Sep 8 22:50:53 2009 From: josh at radianttiger.com (Josh Rivers) Date: Tue, 8 Sep 2009 22:50:53 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <63A023F4-1AA8-4A56-8029-91F5A5C74084@avgs.ca> References: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <63A023F4-1AA8-4A56-8029-91F5A5C74084@avgs.ca> Message-ID: I'm running of the binary release, so I don't have debug symbols for the freeswitch core. I can do a build...but does somebody else already have one handy? -Josh On Tue, Sep 8, 2009 at 10:33 PM, Mathieu Rene wrote: > Click Break, then go in Window, Debug, Stack Trace (or something similar, I > don't have any VS nearby), then copy paste that. > Mathieu Rene > Avant-Garde Solutions Inc > Office: + 1 (514) 664-1044 x100 > Cell: +1 (514) 664-1044 x200 > mrene at avgs.ca > > > > > On 8-Sep-09, at 10:30 PM, Josh Rivers wrote: > > Here is the error I get with the loop I mentioned. -Josh > > > On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo wrote: > >> Hi, >> >> >> Can you please elaborate on the crash you receive when you >> queue a thread during load? >> >> >> Thanks, >> >> Michael >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers >> *Sent:* Tuesday, September 08, 2009 12:22 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / >> .NET >> >> >> Thanks for the response! >> >> >> I have tried putting a long-running loop here, but then it blocks anything >> else managed from happening: >> >> >> public class TestLoop : ILoadNotificationPlugin >> >> { >> >> public bool Load() >> >> { >> >> EventConsumer con = new EventConsumer("all", ""); >> >> while (true) >> >> { >> >> Event ev = con.pop(0); >> >> Log.WriteLine(LogLevel.Notice, "Event: " + >> ev.serialized_string); >> >> freeswitch.msleep(100); >> >> } >> >> } >> >> } >> >> >> However, if I fork off a thread here, freeswitch crashes: >> >> public class TestLoop : ILoadNotificationPlugin >> >> { >> >> public bool Load() >> >> { >> >> ThreadPool.QueueUserWorkItem((o) => >> >> { >> >> Log.WriteLine(LogLevel.Notice, "Thread Starting. "); >> >> EventConsumer con = new EventConsumer("all", ""); >> >> while (true) >> >> { >> >> Event ev = con.pop(0); >> >> Log.WriteLine(LogLevel.Notice, "Event: " + >> ev.serialized_string); >> >> freeswitch.msleep(100); >> >> } >> >> }); >> >> return true; >> >> } >> >> } >> >> >> It doesn't look like this is a good place to start a long-running process? >> >> >> Thanks! >> >> Josh >> >> >> On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi < >> raffaele.p.guidi at gmail.com> wrote: >> >> Yes! >> >> >> public class LoadDemo : ILoadNotificationPlugin { >> >> public bool Load() { >> >> Log.WriteLine(LogLevel.Notice, "LoadDemo running."); >> >> return true; >> >> } >> >> } >> >> >> this example is from Michael Giagnocavo's Demo.csx which you can find into >> the mod_managed svn. >> >> >> And let me add that works like a charm :) >> >> >> Ciao, >> >> Raffaele >> >> >> On Sun, Sep 6, 2009 at 22:50, Josh Rivers wrote: >> >> Is there a way to start this when FreeSWITCH starts? The lua and perl >> modules have a 'startup-script' configuration preference. Is there something >> similar in mod_managed? Or is there a way to have an api command executed at >> a startup? >> >> >> >> >> Exactly what I was after - thank you! >> >> >> On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk wrote: >> >> >> > >> >> > try something like this >> >> > >> >> > EventConsumer con = new EventConsumer("all", ""); >> >> > Event ev = con.pop(0); >> >> > >> >> > see lua sample - >> >> > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer >> >> > >> >> > >> >> > Phillip Jones-2 wrote: >> >> > > >> >> > > Hi there, >> >> > > >> >> > > mod_managed exposes EventReceivedFunction such that: >> >> > > >> >> > > Session.EventReceivedFunction = (e) => >> >> > > { >> >> > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", >> e.ToString()); >> >> > > return ""; >> >> > > }; >> >> > > >> >> > > should trap all events to which i subscribe. >> >> > > >> >> > > >> >> > > But how do I subscribe to events? What is the .NET / managed >> equivalent >> >> > > of: >> >> > > >> >> > > switch_event_bind(const char *id, switch_event_types_t event, const >> char >> >> > > *subclass_name, switch_event_callback_t callback, void *user_data); >> >> > > >> >> > > >> >> > > >> >> > > Thank you! >> >> > > >> >> > > >> >> > > >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/17c7677d/attachment.html From mrene_lists at avgs.ca Tue Sep 8 22:51:48 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Tue, 8 Sep 2009 22:51:48 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4C0C@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4D2C@mse17be1.mse17.exchange.ms> <367751820909081450n4e6faae7h14b4f22d28d1a6d9@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4DC9@mse17be1.mse17.exchange.ms> Message-ID: <6CCB2860-F96F-4F04-A11F-3F65E96CC717@avgs.ca> "Is the trypop method returning without unlocking the mutex?" No, since the condition returns if a lock could NOT be acquired. (hence the try). Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 8-Sep-09, at 10:46 PM, Josh Rivers wrote: > I confirmed that pop(1) works for me. since it's spinning in it's > own thread, that's probably the right solution. These are the > relevant pieces of core code, though: > > > APU_DECLARE(apr_status_t) apr_queue_trypop(apr_queue_t *queue, void > **data) > { > apr_status_t rv; > > if (queue->terminated) { > return APR_EOF; /* no more elements ever again */ > } > > rv = apr_thread_mutex_lock(queue->one_big_mutex); > if (rv != APR_SUCCESS) { > return rv; > } > > if (apr_queue_empty(queue)) { > rv = apr_thread_mutex_unlock(queue->one_big_mutex); > return APR_EAGAIN; > } > > *data = queue->data[queue->out]; > queue->nelts--; > > queue->out = (queue->out + 1) % queue->bounds; > if (queue->full_waiters) { > Q_DBG("signal !full", queue); > rv = apr_thread_cond_signal(queue->not_full); > if (rv != APR_SUCCESS) { > apr_thread_mutex_unlock(queue->one_big_mutex); > return rv; > } > } > > rv = apr_thread_mutex_unlock(queue->one_big_mutex); > return rv; > } > > AND > > APU_DECLARE(apr_status_t) apr_queue_pop(apr_queue_t *queue, void > **data) > { > apr_status_t rv; > > if (queue->terminated) { > return APR_EOF; /* no more elements ever again */ > } > > rv = apr_thread_mutex_lock(queue->one_big_mutex); > if (rv != APR_SUCCESS) { > return rv; > } > > /* Keep waiting until we wake up and find that the queue is not > empty. */ > if (apr_queue_empty(queue)) { > if (!queue->terminated) { > queue->empty_waiters++; > rv = apr_thread_cond_wait(queue->not_empty, queue- > >one_big_mutex); > queue->empty_waiters--; > if (rv != APR_SUCCESS) { > apr_thread_mutex_unlock(queue->one_big_mutex); > return rv; > } > } > /* If we wake up and it's still empty, then we were > interrupted */ > if (apr_queue_empty(queue)) { > Q_DBG("queue empty (intr)", queue); > rv = apr_thread_mutex_unlock(queue->one_big_mutex); > if (rv != APR_SUCCESS) { > return rv; > } > if (queue->terminated) { > return APR_EOF; /* no more elements ever again */ > } > else { > return APR_EINTR; > } > } > } > > *data = queue->data[queue->out]; > queue->nelts--; > > queue->out = (queue->out + 1) % queue->bounds; > if (queue->full_waiters) { > Q_DBG("signal !full", queue); > rv = apr_thread_cond_signal(queue->not_full); > if (rv != APR_SUCCESS) { > apr_thread_mutex_unlock(queue->one_big_mutex); > return rv; > } > } > > rv = apr_thread_mutex_unlock(queue->one_big_mutex); > return rv; > } > > Is the trypop method returning without unlocking the mutex? > > rv = apr_thread_mutex_lock(queue->one_big_mutex); > if (rv != APR_SUCCESS) { > return rv; > } > > On Tue, Sep 8, 2009 at 7:06 PM, Michael Giagnocavo > wrote: > I?m not sure how EventConsumer is supposed to work ? maybe one of > the real devs can explain how pop works and if it should fail on pop > 0 or not. > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Phillip Jones > Sent: Tuesday, September 08, 2009 3:50 PM > > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Subscribing to events in managed > C# / .NET > > > I build this out. > > This seems to work fine: > > ThreadPool.QueueUserWorkItem((o) => > { > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > EventConsumer con = new EventConsumer("all", ""); > while (true) > { > Event ev = con.pop(1); > Log.WriteLine(LogLevel.Alert, "Event: " + > ev.GetHeader("Event-Name")); > //freeswitch.msleep(100); > } > }); > > With > Event ev = con.pop(0) however FS crashes with a > System.NullReferenceException (attached) > > > > > > > > > > On Tue, Sep 8, 2009 at 5:20 PM, Raffaele P. Guidi > wrote: > > Oh, I see... all those years wasted doing java stuff! :D > > > On Tue, Sep 8, 2009 at 22:46, Michael Giagnocavo > wrote: > > ? ThreadPool.QueueUserWorkItem((o) =>? > > That starts a lambda, which is compiled to a delegate, same as > anonymous methods. > > Guess I?ll wait for him to respond on the crash he gets. > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Raffaele P. Guidi > Sent: Tuesday, September 08, 2009 12:22 PM > > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Subscribing to events in managed > C# / .NET > > > Well, I can't see any delegate in josh sample, just a > ThreadPool.QueueUserWorkItem. Here is an example that, at least on > my system (I reached my home pc in the meanwhile), works fine. > > > public class LoadPluginDemo : ILoadNotificationPlugin { > > delegate void Listener(); > > private void EventListener() { > > EventConsumer con = new EventConsumer("all", null); > > while (true){ > > Event ev = con.pop(1); > > Log.WriteLine(LogLevel.Notice, "Got event " + > ev.GetHeader("Event-Name")); > > } > > } > > public bool Load() { > > Log.WriteLine(LogLevel.Notice, "LoadDemo running."); > > new Listener(EventListener).BeginInvoke(null,null); > > return true; > > } > > } > > > > > On Tue, Sep 8, 2009 at 18:43, Michael Giagnocavo > wrote: > > That?s what his sample does, but he says it crashes. > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] On Behalf Of Raffaele P. Guidi > Sent: Tuesday, September 08, 2009 10:08 AM > > > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Subscribing to events in managed > C# / .NET > > > Hi, you just have to use delegates to asynchronously call the > function containing the loop and return back the control to the > calling thread. Here an example (don't have my code at hand, hope it > doesn't contain typos). > > > Regards, > > Raffaele > > > public class TestLoop : ILoadNotificationPlugin > > { > > > Delegate void DoStuffDelegate(); > > > public void doStuff() > > { > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > } > > public bool Load() > > { > > DoStuffDelegate dsdlg = new DoStuffDelegate(doStuff); > > dsdlg.BeginInvoke(); > > } > > } > > On Tue, Sep 8, 2009 at 08:21, Josh Rivers > wrote: > > Thanks for the response! > > > I have tried putting a long-running loop here, but then it blocks > anything else managed from happening: > > > public class TestLoop : ILoadNotificationPlugin > > { > > public bool Load() > > { > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > } > > } > > > However, if I fork off a thread here, freeswitch crashes: > > public class TestLoop : ILoadNotificationPlugin > > { > > public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => > > { > > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > freeswitch.msleep(100); > > } > > }); > > return true; > > } > > } > > > It doesn't look like this is a good place to start a long-running > process? > > > Thanks! > > Josh > > > On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi > wrote: > > Yes! > > > public class LoadDemo : ILoadNotificationPlugin { > > public bool Load() { > > Log.WriteLine(LogLevel.Notice, "LoadDemo running."); > > return true; > > } > > } > > > this example is from Michael Giagnocavo's Demo.csx which you can > find into the mod_managed svn. > > > And let me add that works like a charm :) > > > Ciao, > > Raffaele > > > On Sun, Sep 6, 2009 at 22:50, Josh Rivers > wrote: > > Is there a way to start this when FreeSWITCH starts? The lua and > perl modules have a 'startup-script' configuration preference. Is > there something similar in mod_managed? Or is there a way to have an > api command executed at a startup? > > > > > Exactly what I was after - thank you! > > > On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk > wrote: > > > > > > > try something like this > > > > > > EventConsumer con = new EventConsumer("all", ""); > > > Event ev = con.pop(0); > > > > > > see lua sample - > > > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer > > > > > > > > > Phillip Jones-2 wrote: > > > > > > > > Hi there, > > > > > > > > mod_managed exposes EventReceivedFunction such that: > > > > > > > > Session.EventReceivedFunction = (e) => > > > > { > > > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", > e.ToString()); > > > > return ""; > > > > }; > > > > > > > > should trap all events to which i subscribe. > > > > > > > > > > > > But how do I subscribe to events? What is the .NET / managed > equivalent > > > > of: > > > > > > > > switch_event_bind(const char *id, switch_event_types_t event, > const char > > > > *subclass_name, switch_event_callback_t callback, void > *user_data); > > > > > > > > > > > > > > > > Thank you! > > > > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090908/e042fdfa/attachment-0001.html From panayotov.vd at gmail.com Tue Sep 8 23:20:41 2009 From: panayotov.vd at gmail.com (Vassil Panayotov) Date: Wed, 9 Sep 2009 09:20:41 +0300 Subject: [Freeswitch-users] Sangoma A500 - dial out from specific port group? Message-ID: <8a9b664c0909082320l3be47aedk8d67c2c83d5b411d@mail.gmail.com> Hi, Is it possible to originate calls from specific A500 ports with FreeSWITCH? I am using a A504 (8 BRI interfaces), and I want some outbound calls to be made from specific BRI interfaces. I tried to modify OpenZAP config as follows: conf/openzap.conf [span wanpipe boostbri1] trunk_type => bri b-channel => 1:1-2 b-channel => 2:1-2 b-channel => 3:1-2 b-channel => 4:1-2 b-channel => 5:1-2 b-channel => 6:1-2 [span wanpipe boostbri2] trunk_type => bri b-channel => 7:1-2 b-channel => 8:1-2 conf/autoload_configs/openzap.conf.xml: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/0cbb489a/attachment.html From panayotov.vd at gmail.com Tue Sep 8 23:31:18 2009 From: panayotov.vd at gmail.com (Vassil Panayotov) Date: Wed, 9 Sep 2009 09:31:18 +0300 Subject: [Freeswitch-users] Sangoma A500 - dial out from specific port group? In-Reply-To: <8a9b664c0909082320l3be47aedk8d67c2c83d5b411d@mail.gmail.com> References: <8a9b664c0909082320l3be47aedk8d67c2c83d5b411d@mail.gmail.com> Message-ID: <8a9b664c0909082331x3bbb5510g970bec0167213ac4@mail.gmail.com> Sorry I hit 'send' by mistake... Hi, Is it possible to originate calls from specific A500 ports with FreeSWITCH? I am using a A504 (8 BRI interfaces), and I want some outbound calls to be made from specific BRI interfaces. I tried to modify OpenZAP config as follows: conf/openzap.conf [span wanpipe boostbri1] trunk_type => bri b-channel => 1:1-2 b-channel => 2:1-2 b-channel => 3:1-2 b-channel => 4:1-2 b-channel => 5:1-2 b-channel => 6:1-2 [span wanpipe boostbri2] trunk_type => bri b-channel => 7:1-2 b-channel => 8:1-2 conf/autoload_configs/openzap.conf.xml: When I try to originate call I am getting errors: freeswitch at emo-voip> originate openzap/2/a/123456 music API CALL [originate(openzap/2/a/123456 music)] output: -ERR NORMAL_CIRCUIT_CONGESTION 2009-09-04 09:23:34.87253 [CRIT] ozmod_ss7_boost.c:244 SPAN is not online. 2009-09-04 09:23:34.87253 [ERR] mod_openzap.c:1043 No channels available 2009-09-04 09:23:34.87253 [ERR] switch_ivr_originate.c:1510 Cannot create outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION] Then I tried to modify the /etc/wanpipe/smg_bri.conf: ;Sangoma AFT-A500 port 11 [slot:8 bus:1 span:7] group=2 country=europe operator=etsi connection_type=point_to_point signalling=bri_nt spans=7 ;Sangoma AFT-A500 port 12 [slot:8 bus:1 span:8] group=2 country=europe operator=etsi connection_type=point_to_point signalling=bri_nt spans=8 i.e. changed 'group' to 2, but this doesn't help either. Marc Celsie from Sangoma's techdesk told me that I should ' dial X at gY with X being the number and Y being the group number'. How originate command should look like in this case? originate openzap/1/a/123456 at g2 someExt ? I tried this syntax but with no effect. Marc also told me that there is a bug in FS which prevents groups from working. Should I fill bug report or feature request? Best regards, Vassil Panayotov -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/b536f909/attachment.html From math.parent at gmail.com Wed Sep 9 02:16:34 2009 From: math.parent at gmail.com (Mathieu Parent) Date: Wed, 9 Sep 2009 11:16:34 +0200 Subject: [Freeswitch-users] mod_fax not working In-Reply-To: <960738410909010406x4d938d0cx26c79e0fc83b8c1f@mail.gmail.com> References: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com> <4A980D58.5020808@coppice.org> <960738410909010406x4d938d0cx26c79e0fc83b8c1f@mail.gmail.com> Message-ID: <960738410909090216h1c369a94jbf0591816c6874ba@mail.gmail.com> Hello, On Tue, Sep 1, 2009 at 1:06 PM, Mathieu Parent wrote: > Hi, > > > On Fri, Aug 28, 2009 at 7:01 PM, Steve Underwood wrote: > (snip) >>> >> The log shows the same thing happening every time. A bad CRC from the >> far end, followed by a good DCS frame followed by what seems to be >> rubbish. I think I'd need an audio log from one of these calls to figure >> out any more. >> > > I have attached a pcap file only with SIP and RTP. I still have the same problem. Anyone can analyse the traces ? Note that it works without problems when sending a fax. Mathieu Parent From fraunhofer.lists.freeswitch-001 at traced.net Wed Sep 9 02:58:21 2009 From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer) Date: Wed, 9 Sep 2009 11:58:21 +0200 Subject: [Freeswitch-users] memory leak In-Reply-To: <191c3a030909040757y93c105bs8939559f90142b9e@mail.gmail.com> References: <191c3a030909020815k6006f96fh3df8512018f9b28d@mail.gmail.com> <191c3a030909020842j61369a45va923080ea0a88b40@mail.gmail.com> <191c3a030909040757y93c105bs8939559f90142b9e@mail.gmail.com> Message-ID: Hello *, the latest bugfixes for luarun (pool allocation) fixed it. It's working now with luarun (friday-monday) and bgapi (monday-today). The last test with the new "&" operator for sched_api is currently running. Thx! Beni. -------------- next part -------------- A non-text attachment was scrubbed... Name: mem-sz_luarun_bgapi.png Type: image/png Size: 18188 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/9389b1d8/attachment-0001.png From tzury.by at reguluslabs.com Wed Sep 9 04:19:38 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Wed, 9 Sep 2009 14:19:38 +0300 Subject: [Freeswitch-users] No audio on caller side when both side support speex/8000 only Message-ID: <10128ef10909090419g196d2280p5d851728fb8c6309@mail.gmail.com> Hi, Owe to the network bandwidth limitations (running on cellular phones ip link) we are using speex/8000 as our voice codec. However, when both parties are using that codec the sound is not to be heard on the caller side. looking at the log dumps one can see that a) at the caller side, it supports speex/8000 in pt=102 and receives from the server speex/8000 in pt=102 b) at the callee side FreeSwitch supports support speex/8000 in pt=98 although it receives from the client speex/8000 in pt=102 When the voice starts caller sends RTP with pt=102 and expect to receive RTP with pt=102, while the callee sends RTP with pt=98 and expect to receive RTP with pt=102. The RTP packets that received in the caller side are with pt=98 instead of 102 and thusly the client drops them. Attached are the 2 files recorded from a call between 2 pjsip clients that support only speex/8000 codec. un_FSCallerSide-speexClient.TXT ? is the caller side SIP messages. un_FSAnswerSide-speexClient.TXT ? is the answer side of SIP messages. Is there anything can be done at the configuration level to avoid this? Thanks in advance for your help /tzury -------------- next part -------------- --start msg (RX)-- INVITE sip:1002 at 95.35.241.89:5060 SIP/2.0 Via: SIP/2.0/UDP 67.23.5.142;rport;branch=z9hG4bKgvD702De7e0Se Max-Forwards: 69 From: "Extension 1001" ;tag=2rH67Q3aa1rpe To: Call-ID: e56c2918-17ad-122d-de9e-40402384297d CSeq: 120120747 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 415 Remote-Party-ID: "Extension 1001" ;screen=yes;privacy=off v=0 o=FreeSWITCH 4131815555116427886 953315150658749217 IN IP4 67.23.5.142 s=FreeSWITCH c=IN IP4 67.23.5.142 t=0 0 m=audio 26662 RTP/AVP 98 0 8 3 9 99 103 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 SPEEX/16000 a=rtpmap:103 SPEEX/32000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 --end msg-- --start msg (TX)-- SIP/2.0 100 Trying Via: SIP/2.0/UDP 67.23.5.142;rport=5060;received=67.23.5.142;branch=z9hG4bKgvD702De7e0Se Call-ID: e56c2918-17ad-122d-de9e-40402384297d From: "Extension 1001" ;tag=2rH67Q3aa1rpe To: CSeq: 120120747 INVITE Content-Length: 0 --end msg-- --start msg (TX)-- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 67.23.5.142;rport=5060;received=67.23.5.142;branch=z9hG4bKgvD702De7e0Se Call-ID: e56c2918-17ad-122d-de9e-40402384297d From: "Extension 1001" ;tag=2rH67Q3aa1rpe To: ;tag=e4f9fb648edef1c48dbc8b8b474409e6 CSeq: 120120747 INVITE Contact: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 --end msg-- --start msg (RX)-- INVITE sip:1002 at 95.35.241.89:5060 SIP/2.0 Via: SIP/2.0/UDP 67.23.5.142;rport;branch=z9hG4bKgvD702De7e0Se Max-Forwards: 69 From: "Extension 1001" ;tag=2rH67Q3aa1rpe To: Call-ID: e56c2918-17ad-122d-de9e-40402384297d CSeq: 120120747 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 415 Remote-Party-ID: "Extension 1001" ;screen=yes;privacy=off v=0 o=FreeSWITCH 4131815555116427886 953315150658749217 IN IP4 67.23.5.142 s=FreeSWITCH c=IN IP4 67.23.5.142 t=0 0 m=audio 26662 RTP/AVP 98 0 8 3 9 99 103 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 SPEEX/16000 a=rtpmap:103 SPEEX/32000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 --end msg-- --start msg (TX)-- SIP/2.0 180 Ringing Via: SIP/2.0/UDP 67.23.5.142;rport=5060;received=67.23.5.142;branch=z9hG4bKgvD702De7e0Se Call-ID: e56c2918-17ad-122d-de9e-40402384297d From: "Extension 1001" ;tag=2rH67Q3aa1rpe To: ;tag=e4f9fb648edef1c48dbc8b8b474409e6 CSeq: 120120747 INVITE Contact: Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Content-Length: 0 --end msg-- --start msg (TX)-- SIP/2.0 200 OK Via: SIP/2.0/UDP 67.23.5.142;rport=5060;received=67.23.5.142;branch=z9hG4bKgvD702De7e0Se Call-ID: e56c2918-17ad-122d-de9e-40402384297d From: "Extension 1001" ;tag=2rH67Q3aa1rpe To: ;tag=e4f9fb648edef1c48dbc8b8b474409e6 CSeq: 120120747 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 256 v=0 o=- 3461503025 3461503026 IN IP4 95.35.241.89 s=pjmedia c=IN IP4 95.35.241.89 t=0 0 a=X-nat:5 m=audio 4000 RTP/AVP 102 101 a=rtcp:4001 IN IP4 95.35.241.89 a=rtpmap:102 speex/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- --start msg (RX)-- NOTIFY sip:1002 at 95.35.241.89:5060 SIP/2.0 Via: SIP/2.0/UDP 67.23.5.142;rport;branch=z9hG4bKeaUNXcc7cXKmQ Max-Forwards: 70 From: ;tag=S7D7Q4BB4ggep To: Call-ID: d74d73dc-17ad-122d-de9e-40402384297d CSeq: 120120735 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;timeout Content-Type: application/simple-message-summary Content-Length: 97 Messages-Waiting: yes Message-Account: sip:1002 at cheerfulsanity.net Voice-Message: 3/0 (0/0) --end msg-- --start msg (TX)-- SIP/2.0 200 OK Via: SIP/2.0/UDP 67.23.5.142;rport=5060;received=67.23.5.142;branch=z9hG4bKgvD702De7e0Se Call-ID: e56c2918-17ad-122d-de9e-40402384297d From: "Extension 1001" ;tag=2rH67Q3aa1rpe To: ;tag=e4f9fb648edef1c48dbc8b8b474409e6 CSeq: 120120747 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 256 v=0 o=- 3461503025 3461503026 IN IP4 95.35.241.89 s=pjmedia c=IN IP4 95.35.241.89 t=0 0 a=X-nat:5 m=audio 4000 RTP/AVP 102 101 a=rtcp:4001 IN IP4 95.35.241.89 a=rtpmap:102 speex/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- -------------- next part -------------- --start msg (TX)-- INVITE sip:1002 at cheerfulsanity.net SIP/2.0 Via: SIP/2.0/UDP 192.118.11.112:64680;rport;branch=z9hG4bKPj3caad40720064a8f124c21cf99b8b1c1 Max-Forwards: 70 From: sip:1001 at cheerfulsanity.net;tag=6d6ef114e663e48226f2b1e598313a2e To: sip:1002 at cheerfulsanity.net Contact: Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23264 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 264 v=0 o=- 3461521040 3461521040 IN IP4 192.118.11.112 s=pjmedia c=IN IP4 192.118.11.112 t=0 0 a=X-nat:8 m=audio 64976 RTP/AVP 102 101 a=rtcp:64980 IN IP4 192.118.11.112 a=rtpmap:102 speex/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- --start msg (RX)-- SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.171.9.67:5060;rport=64680;branch=z9hG4bKPj3caad40720064a8f124c21cf99b8b1c1 From: ;tag=6d6ef114e663e48226f2b1e598313a2e To: Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23264 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Content-Length: 0 --end msg-- --start msg (RX)-- SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 10.171.9.67:5060;rport=64680;branch=z9hG4bKPj3caad40720064a8f124c21cf99b8b1c1 From: ;tag=6d6ef114e663e48226f2b1e598313a2e To: ;tag=ymc30B0vNXZBg Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23264 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Proxy-Authenticate: Digest realm="cheerfulsanity.net", nonce="0d21b0de-9d0b-11de-afe2-9bb840543c49", algorithm=MD5, qop="auth" Content-Length: 0 --end msg-- --start msg (TX)-- ACK sip:1002 at cheerfulsanity.net SIP/2.0 Via: SIP/2.0/UDP 192.118.11.112:64680;rport;branch=z9hG4bKPj3caad40720064a8f124c21cf99b8b1c1 Max-Forwards: 70 From: sip:1001 at cheerfulsanity.net;tag=6d6ef114e663e48226f2b1e598313a2e To: sip:1002 at cheerfulsanity.net;tag=ymc30B0vNXZBg Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23264 ACK Content-Length: 0 --end msg-- --start msg (TX)-- INVITE sip:1002 at cheerfulsanity.net SIP/2.0 Via: SIP/2.0/UDP 192.118.11.112:64680;rport;branch=z9hG4bKPje8deea3c8476457603fe0cc301731002 Max-Forwards: 70 From: sip:1001 at cheerfulsanity.net;tag=6d6ef114e663e48226f2b1e598313a2e To: sip:1002 at cheerfulsanity.net Contact: Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23265 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Proxy-Authorization: Digest username="1001", realm="cheerfulsanity.net", nonce="0d21b0de-9d0b-11de-afe2-9bb840543c49", uri="sip:1002 at cheerfulsanity.net", response="5eab15df8541648b8890866f976a23e1", algorithm=MD5, cnonce="03d7b5789aacc7b832d4d618ca295ba2", qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 264 v=0 o=- 3461521040 3461521040 IN IP4 192.118.11.112 s=pjmedia c=IN IP4 192.118.11.112 t=0 0 a=X-nat:8 m=audio 64976 RTP/AVP 102 101 a=rtcp:64980 IN IP4 192.118.11.112 a=rtpmap:102 speex/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- --start msg (RX)-- SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.171.9.67:5060;rport=64680;branch=z9hG4bKPje8deea3c8476457603fe0cc301731002 From: ;tag=6d6ef114e663e48226f2b1e598313a2e To: Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23265 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Content-Length: 0 --end msg-- --start msg (RX)-- NOTIFY sip:1001 at 10.171.9.67:5060 SIP/2.0 Via: SIP/2.0/UDP 67.23.5.142;rport;branch=z9hG4bKFKmeZ7vaa696j Max-Forwards: 70 From: ;tag=XBKaZgFSrm9rm To: Call-ID: e07248c2-17ad-122d-de9e-40402384297d CSeq: 120120743 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;timeout Content-Type: application/simple-message-summary Content-Length: 70 Messages-Waiting: no Message-Account: sip:1001 at cheerfulsanity.net --end msg-- --start msg (RX)-- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.171.9.67:5060;rport=64680;branch=z9hG4bKPje8deea3c8476457603fe0cc301731002 From: ;tag=6d6ef114e663e48226f2b1e598313a2e To: ;tag=06ym4113FFcHQ Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23265 INVITE Contact: RSeq: 669378962 User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Require: 100rel Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 268 v=0 o=FreeSWITCH 4446028933093139022 3405868075899860026 IN IP4 67.23.5.142 s=FreeSWITCH c=IN IP4 67.23.5.142 t=0 0 m=audio 26664 RTP/AVP 102 101 a=rtpmap:102 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 --end msg-- --start msg (TX)-- PRACK sip:mod_sofia at 67.23.5.142:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.118.11.112:64680;rport;branch=z9hG4bKPj3163ba6def02fe4c795703c1d1bef593 Max-Forwards: 70 From: sip:1001 at cheerfulsanity.net;tag=6d6ef114e663e48226f2b1e598313a2e To: sip:1002 at cheerfulsanity.net;tag=06ym4113FFcHQ Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23266 PRACK RAck: 669378962 23265 INVITE Content-Length: 0 --end msg-- --start msg (RX)-- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 10.171.9.67:5060;rport=64680;branch=z9hG4bKPje8deea3c8476457603fe0cc301731002 From: ;tag=6d6ef114e663e48226f2b1e598313a2e To: ;tag=06ym4113FFcHQ Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23265 INVITE Contact: RSeq: 669378962 User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Require: 100rel Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 268 v=0 o=FreeSWITCH 4446028933093139022 3405868075899860026 IN IP4 67.23.5.142 s=FreeSWITCH c=IN IP4 67.23.5.142 t=0 0 m=audio 26664 RTP/AVP 102 101 a=rtpmap:102 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 --end msg-- --start msg (RX)-- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.171.9.67:5060;rport=64680;branch=z9hG4bKPj3163ba6def02fe4c795703c1d1bef593 From: ;tag=6d6ef114e663e48226f2b1e598313a2e To: ;tag=06ym4113FFcHQ Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23266 PRACK Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Content-Length: 0 --end msg-- --start msg (RX)-- NOTIFY sip:1001 at 10.171.9.67:5060 SIP/2.0 Via: SIP/2.0/UDP 67.23.5.142;rport;branch=z9hG4bKFKmeZ7vaa696j Max-Forwards: 70 From: ;tag=XBKaZgFSrm9rm To: Call-ID: e07248c2-17ad-122d-de9e-40402384297d CSeq: 120120743 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;timeout Content-Type: application/simple-message-summary Content-Length: 70 Messages-Waiting: no Message-Account: sip:1001 at cheerfulsanity.net --end msg-- --start msg (RX)-- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.171.9.67:5060;rport=64680;branch=z9hG4bKPje8deea3c8476457603fe0cc301731002 From: ;tag=6d6ef114e663e48226f2b1e598313a2e To: ;tag=06ym4113FFcHQ Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23265 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 1800;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 268 v=0 o=FreeSWITCH 4446028933093139022 3405868075899860026 IN IP4 67.23.5.142 s=FreeSWITCH c=IN IP4 67.23.5.142 t=0 0 m=audio 26664 RTP/AVP 102 101 a=rtpmap:102 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 --end msg-- --start msg (TX)-- ACK sip:mod_sofia at 67.23.5.142:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.118.11.112:64680;rport;branch=z9hG4bKPj949393fcf5e36fe32bb24b420101cd20 Max-Forwards: 70 From: sip:1001 at cheerfulsanity.net;tag=6d6ef114e663e48226f2b1e598313a2e To: sip:1002 at cheerfulsanity.net;tag=06ym4113FFcHQ Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23265 ACK Content-Length: 0 --end msg-- ---------------------------------------------- Start receiving RTP packets with PT=98 instead of 102 From tzury.by at reguluslabs.com Wed Sep 9 04:24:46 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Wed, 9 Sep 2009 14:24:46 +0300 Subject: [Freeswitch-users] No audio on caller side when both side support speex/8000 only In-Reply-To: <10128ef10909090419g196d2280p5d851728fb8c6309@mail.gmail.com> References: <10128ef10909090419g196d2280p5d851728fb8c6309@mail.gmail.com> Message-ID: <10128ef10909090424g58b2a26bx837e9d71a4c42347@mail.gmail.com> Hi, Owe to the network bandwidth limitations (running on cellular phones ip link) we are using speex/8000 as our voice codec. However, when both parties are using that codec the sound is not to be heard on the caller side. looking at the log dumps one can see that a) at the caller side, it supports speex/8000 in pt=102 and receives from the server speex/8000 in pt=102 b) at the callee side FreeSwitch supports support speex/8000 in pt=98 although it receives from the client speex/8000 in pt=102 When the voice starts caller sends RTP with pt=102 and expect to receive RTP with pt=102, while the callee sends RTP with pt=98 and expect to receive RTP with pt=102. The RTP packets that received in the caller side are with pt=98 instead of 102 and thusly the client drops them. ############### LOG DUMPS ############### ############### CALLER SIDE ## start msg (TX) INVITE sip:1002 at SERVER_DOMAIN SIP/2.0 Via: SIP/2.0/UDP X.X.X.X:64680;rport;branch=z9hG4bKPj3caad40720064a8f124c21cf99b8b1c1 Max-Forwards: 70 From: sip:1001 at SERVER_DOMAIN;tag=6d6ef114e663e48226f2b1e598313a2e To: sip:1002 at SERVER_DOMAIN Contact: Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23264 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 Content-Type: application/sdp Content-Length: 264 v=0 o=- 3461521040 3461521040 IN IP4 X.X.X.X s=pjmedia c=IN IP4 X.X.X.X t=0 0 a=X-nat:8 m=audio 64976 RTP/AVP 102 101 a=rtcp:64980 IN IP4 X.X.X.X a=rtpmap:102 speex/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- ## start msg (RX) SIP/2.0 200 OK Via: SIP/2.0/UDP 10.171.9.67:5060;rport=64680;branch=z9hG4bKPje8deea3c8476457603fe0cc301731002 From: ;tag=6d6ef114e663e48226f2b1e598313a2e To: ;tag=06ym4113FFcHQ Call-ID: 1473e9e828658e3fb0370fabf2ce8986 CSeq: 23265 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Require: timer Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Session-Expires: 1800;refresher=uac Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 268 v=0 o=FreeSWITCH 4446028933093139022 3405868075899860026 IN IP4 X.X.X.X s=FreeSWITCH c=IN IP4 X.X.X.X t=0 0 m=audio 26664 RTP/AVP 102 101 a=rtpmap:102 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 --end msg-- ############### CALLEE SIDE ## start msg (RX) INVITE sip:1002 at X.X.X.X:5060 SIP/2.0 Via: SIP/2.0/UDP X.X.X.X;rport;branch=z9hG4bKgvD702De7e0Se Max-Forwards: 69 From: "Extension 1001" ;tag=2rH67Q3aa1rpe To: Call-ID: e56c2918-17ad-122d-de9e-40402384297d CSeq: 120120747 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.3-exported Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 415 Remote-Party-ID: "Extension 1001" ;screen=yes;privacy=off v=0 o=FreeSWITCH 4131815555116427886 953315150658749217 IN IP4 X.X.X.X s=FreeSWITCH c=IN IP4 X.X.X.X t=0 0 m=audio 26662 RTP/AVP 98 0 8 3 9 99 103 101 13 a=rtpmap:98 SPEEX/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:9 G722/8000 a=rtpmap:99 SPEEX/16000 a=rtpmap:103 SPEEX/32000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 --end msg-- ## start msg (TX) SIP/2.0 200 OK Via: SIP/2.0/UDP X.X.X.X;rport=5060;received=X.X.X.X;branch=z9hG4bKgvD702De7e0Se Call-ID: e56c2918-17ad-122d-de9e-40402384297d From: "Extension 1001" ;tag=2rH67Q3aa1rpe To: ;tag=e4f9fb648edef1c48dbc8b8b474409e6 CSeq: 120120747 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Contact: Supported: replaces, 100rel, timer, norefersub Content-Type: application/sdp Content-Length: 256 v=0 o=- 3461503025 3461503026 IN IP4 X.X.X.X s=pjmedia c=IN IP4 X.X.X.X t=0 0 a=X-nat:5 m=audio 4000 RTP/AVP 102 101 a=rtcp:4001 IN IP4 X.X.X.X a=rtpmap:102 speex/8000 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --end msg-- Attached are the 2 files recorded from a call between 2 pjsip clients that support only speex/8000 codec. un_FSCallerSide-speexClient.TXT ? is the caller side SIP messages. un_FSAnswerSide-speexClient.TXT ? is the answer side of SIP messages. Is there anything can be done at the configuration level to avoid this? Thanks in advance for your help /tzury From jh+freeswitch at jh72.de Wed Sep 9 04:16:57 2009 From: jh+freeswitch at jh72.de (=?ISO-8859-1?Q?J=F6rg_Hartmann?=) Date: Wed, 9 Sep 2009 13:16:57 +0200 Subject: [Freeswitch-users] example configs for FS outside of NAT? In-Reply-To: <938ad7be0909090343x29263d6cq3eeee6bbafbb799c@mail.gmail.com> References: <938ad7be0909090316p2f85f104w8aa3c95b3c8a6ae8@mail.gmail.com> <938ad7be0909090343x29263d6cq3eeee6bbafbb799c@mail.gmail.com> Message-ID: <938ad7be0909090416y1d2986ddyceb017c188235d05@mail.gmail.com> Hi there, the internal.xml and external.xml examples are for situations where FS is running inside a company's private network, behind a NAT router. So internal.xml connects the clients to FS without crossing a NAT, within the same private network, while external.xml connects SIP providers through the NAT router. But what if FS is running with a public IP (and DNS entry) outside the private network, so that the clients have to pass the NAT router to connect with FS, while FS can connect to SIP providers directly? Are there any example configs for such a configuration? Thanks in advance, Cheers, JH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/0d044117/attachment.html From brian at freeswitch.org Wed Sep 9 06:19:34 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Sep 2009 08:19:34 -0500 Subject: [Freeswitch-users] No audio on caller side when both side support speex/8000 only In-Reply-To: <10128ef10909090419g196d2280p5d851728fb8c6309@mail.gmail.com> References: <10128ef10909090419g196d2280p5d851728fb8c6309@mail.gmail.com> Message-ID: This looks and sounds like a case where pjsip isn't listening to our SDP. If we 200 OK with speex on 102 and the far end starts sending it on 98 then I suspect the client is broken if I'm not mistaken. /b On Sep 9, 2009, at 6:19 AM, Tzury Bar Yochay wrote: > Hi, > > Owe to the network bandwidth limitations (running on cellular phones > ip link) we are using speex/8000 as our voice codec. > > However, when both parties are using that codec the sound is not to be > heard on the caller side. > > looking at the log dumps one can see that > > a) at the caller side, it supports speex/8000 in pt=102 and receives > from the server speex/8000 in pt=102 > b) at the callee side FreeSwitch supports support speex/8000 in pt=98 > although it receives from the client speex/8000 in pt=102 > > When the voice starts caller sends RTP with pt=102 and expect to > receive RTP with pt=102, while the callee sends RTP with pt=98 and > expect to receive RTP with pt=102. > > The RTP packets that received in the caller side are with pt=98 > instead of 102 and thusly the client drops them. > > Attached are the 2 files recorded from a call between 2 pjsip clients > that support only speex/8000 codec. > > un_FSCallerSide-speexClient.TXT ? is the caller side SIP messages. > > un_FSAnswerSide-speexClient.TXT ? is the answer side of SIP messages. > > > Is there anything can be done at the configuration level to avoid > this? > Thanks in advance for your help > > /tzury > speexClient.TXT>_______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From ivanov.maxim at gmail.com Wed Sep 9 06:29:40 2009 From: ivanov.maxim at gmail.com (Max Ivanov) Date: Wed, 9 Sep 2009 17:29:40 +0400 Subject: [Freeswitch-users] auto_hunt=true vs execute_extenstion Message-ID: Hi all! Is there any difference between auto_hunt=True and execute_extenstion? From brian at freeswitch.org Wed Sep 9 06:44:54 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Sep 2009 08:44:54 -0500 Subject: [Freeswitch-users] auto_hunt=true vs execute_extenstion In-Reply-To: References: Message-ID: <446DAA70-FF48-4A0E-AB8D-A9361E2FDBE0@freeswitch.org> So if you have an extension name that is "testing" and the destination number is "testing" then if testing is at the bottom of the dialplan auto_hunt will make it warp right to it. /b On Sep 9, 2009, at 8:29 AM, Max Ivanov wrote: > Hi all! > Is there any difference between auto_hunt=True and execute_extenstion? From christian.loeschenkohl at xpirio.com Wed Sep 9 07:07:47 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Wed, 09 Sep 2009 16:07:47 +0200 Subject: [Freeswitch-users] stability problems In-Reply-To: <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> References: <4A9F7067.2090306@xpirio.com> <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> <4A9FD6EE.7070107@xpirio.com> <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> Message-ID: <4AA7B6B3.2030401@xpirio.com> hello anthony i'm sorry the cleanup didn't solve my problem i have opend a jira bug n this - key FSCORE-432 hope this is right br On 2009-09-03 17:02, Anthony Minessale wrote: > Which revision are you using? > > If you are not running the latest trunk, please upgrade to that in case > your problem requires us to change the code > we need it to be up to date. > > > 1) Remove any binary files which may get mixed in from an older build > rm /usr/local/freeswitch/bin/* > rm /usr/local/freeswitch/lib/* > rm /usr/local/freeswitch/mod > 2) Build Latest Trunk > 3) Reproduce the problem. > > If you get the problem keep FreeSWITCH running and capture a gcore back > trace. > > ./scripts/freeswitch-gcore > gcore.txt > > Send us the file as an attachment or attached to a new jira issue. > http://jira.freeswitch.org > > > > > > > 2009/9/3 Christian L?schenkohl > > > on debian lenny amd64 with the build-essential package > > an then with > > ./configure --prefix=/opt/freeswitch > make > make install > > nothing else > > br > > On 2009-09-03 16:12, Brian West wrote: > > Sounds like you have some build skew... can you tell us how you built > > FreeSWITCH? > > > > /b > > > > On Sep 3, 2009, at 2:29 AM, Christian L?schenkohl wrote: > > > >> hello > >> > >> we have regular (every 4-6 days) stability problems with freeswitch > >> when the problme occurs > >> > >> - no registers are done bythe server (olny 1 ack of the initial > >> register) > >> - no more calls are working > >> - the calls are all ending with a timeout (cdr caues > >> ORIGINATOR_CANCEL) > >> - only a restart of the whole server cures the problem > >> > >> the server doesn't crash or segfault > >> my first try was to enable the crash-protection flag, but with no > >> difference > >> the server is restartet every night and the last stand still was > >> after about 15h uptime > >> > >> the system is an sun fire 2400 with debian 64 bit system > >> > >> how could i offer you more information to solve this big problem > >> > >> br > >> > >> -- > >> Ing. Christian L?schenkohl > >> 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.loeschenkohl at xpirio.com > > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > 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.loeschenkohl at xpirio.com > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com From rupa at rupa.com Wed Sep 9 07:23:19 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 9 Sep 2009 09:23:19 -0500 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <191c3a030909020842j61369a45va923080ea0a88b40@mail.gmail.com> <191c3a030909040757y93c105bs8939559f90142b9e@mail.gmail.com> Message-ID: On Wed, Sep 9, 2009 at 4:58 AM, Benedikt Fraunhofer wrote: > Hello *, > > the latest bugfixes for luarun (pool allocation) fixed it. > It's working now with luarun (friday-monday) and bgapi (monday-today). > The last test with the new "&" operator for sched_api is currently running. > > Thx! > > ?Beni. Great news! Another one bites the dust. :) -- -Rupa From brian at freeswitch.org Wed Sep 9 07:28:39 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 9 Sep 2009 09:28:39 -0500 Subject: [Freeswitch-users] example configs for FS outside of NAT? In-Reply-To: <938ad7be0909090416y1d2986ddyceb017c188235d05@mail.gmail.com> References: <938ad7be0909090316p2f85f104w8aa3c95b3c8a6ae8@mail.gmail.com> <938ad7be0909090343x29263d6cq3eeee6bbafbb799c@mail.gmail.com> <938ad7be0909090416y1d2986ddyceb017c188235d05@mail.gmail.com> Message-ID: Those configs will still work. /b On Sep 9, 2009, at 6:16 AM, J?rg Hartmann wrote: > Hi there, > > the internal.xml and external.xml examples are for situations where > FS is running inside a company's private network, behind a NAT > router. So internal.xml connects the clients to FS without crossing > a NAT, within the same private network, while external.xml connects > SIP providers through the NAT router. > > But what if FS is running with a public IP (and DNS entry) outside > the private network, so that the clients have to pass the NAT router > to connect with FS, while FS can connect to SIP providers directly? > Are there any example configs for such a configuration? > > Thanks in advance, > Cheers, > JH -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/73064088/attachment.html From ivanov.maxim at gmail.com Wed Sep 9 07:49:20 2009 From: ivanov.maxim at gmail.com (Max Ivanov) Date: Wed, 9 Sep 2009 18:49:20 +0400 Subject: [Freeswitch-users] auto_hunt=true vs execute_extenstion In-Reply-To: <446DAA70-FF48-4A0E-AB8D-A9361E2FDBE0@freeswitch.org> References: <446DAA70-FF48-4A0E-AB8D-A9361E2FDBE0@freeswitch.org> Message-ID: > So if you have an extension name that is "testing" ?and the > destination number is "testing" then if testing is at the bottom of > the dialplan auto_hunt will make it warp right to it. Ah, I see. Would it be correct to say that auto_hunt is similar to "goto" and execute_extenstion behave like "include" ? From anthony.minessale at gmail.com Wed Sep 9 08:18:53 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 9 Sep 2009 10:18:53 -0500 Subject: [Freeswitch-users] stability problems In-Reply-To: <4AA7B6B3.2030401@xpirio.com> References: <4A9F7067.2090306@xpirio.com> <692A0B57-D6F4-4BF8-B2CD-68B40C1E8E79@freeswitch.org> <4A9FD6EE.7070107@xpirio.com> <191c3a030909030802u7fce03b9wf4cd64c02fe6a829@mail.gmail.com> <4AA7B6B3.2030401@xpirio.com> Message-ID: <191c3a030909090818g58ca7afcj5f18a071e437af71@mail.gmail.com> the instructions said build latest trunk. did you actually do that? because lines of code in this gcore file do not correspond to current trunk which is why I asked you to update to it first. Did you just rebuild 1.0.4 again? If you did rebuild trunk what version was it? we can't fix problems on tarball release you have to use the development version. 2009/9/9 Christian L?schenkohl > hello anthony > > i'm sorry the cleanup didn't solve my problem > i have opend a jira bug n this - key FSCORE-432 > hope this is right > > br > > On 2009-09-03 17:02, Anthony Minessale wrote: > > Which revision are you using? > > > > If you are not running the latest trunk, please upgrade to that in case > > your problem requires us to change the code > > we need it to be up to date. > > > > > > 1) Remove any binary files which may get mixed in from an older build > > rm /usr/local/freeswitch/bin/* > > rm /usr/local/freeswitch/lib/* > > rm /usr/local/freeswitch/mod > > 2) Build Latest Trunk > > 3) Reproduce the problem. > > > > If you get the problem keep FreeSWITCH running and capture a gcore back > > trace. > > > > ./scripts/freeswitch-gcore > gcore.txt > > > > Send us the file as an attachment or attached to a new jira issue. > > http://jira.freeswitch.org > > > > > > > > > > > > > > 2009/9/3 Christian L?schenkohl > > > > > > on debian lenny amd64 with the build-essential package > > > > an then with > > > > ./configure --prefix=/opt/freeswitch > > make > > make install > > > > nothing else > > > > br > > > > On 2009-09-03 16:12, Brian West wrote: > > > Sounds like you have some build skew... can you tell us how you > built > > > FreeSWITCH? > > > > > > /b > > > > > > On Sep 3, 2009, at 2:29 AM, Christian L?schenkohl wrote: > > > > > >> hello > > >> > > >> we have regular (every 4-6 days) stability problems with > freeswitch > > >> when the problme occurs > > >> > > >> - no registers are done bythe server (olny 1 ack of the initial > > >> register) > > >> - no more calls are working > > >> - the calls are all ending with a timeout (cdr caues > > >> ORIGINATOR_CANCEL) > > >> - only a restart of the whole server cures the problem > > >> > > >> the server doesn't crash or segfault > > >> my first try was to enable the crash-protection flag, but with > no > > >> difference > > >> the server is restartet every night and the last stand still was > > >> after about 15h uptime > > >> > > >> the system is an sun fire 2400 with debian 64 bit system > > >> > > >> how could i offer you more information to solve this big problem > > >> > > >> br > > >> > > >> -- > > >> Ing. Christian L?schenkohl > > >> 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.loeschenkohl at xpirio.com > > > > >> > > >> _______________________________________________ > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at 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 at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > -- > > Ing. Christian L?schenkohl > > 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.loeschenkohl at xpirio.com > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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_minessale at hotmail.com > > > > > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > > > > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > > > > > iax:guest at conference.freeswitch.org/888 > > > > googletalk:conf+888 at conference.freeswitch.org > > > > > > pstn:213-799-1400 > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Ing. Christian L?schenkohl > 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.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/2e852d0b/attachment-0001.html From jerry.richards at teotech.com Wed Sep 9 09:30:20 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 9 Sep 2009 09:30:20 -0700 Subject: [Freeswitch-users] [ERR] mod_sofia.c:2645 Invalid Gateway Message-ID: I have phones registered internally and can call among them. However, when I dial "711" from an internal phone, freeswitch replies with "484 Address Incomplete" with reason "INVALID_NUMBER_FORMAT". At the server console, I see the following error: [ERR] mod_sofia.c:2645 Invalid Gateway Does anyone know why I get this error? Is there something more I must do to add the gateway below? I already added the following to the usr/local/freesitch/conf/dialplan/default.xml: I already created a usr/local/freeswitch/conf/sip_profiles/external/mediant1000.xml file: Best Regards, Jerry From mrene_lists at avgs.ca Wed Sep 9 09:34:43 2009 From: mrene_lists at avgs.ca (Mathieu Rene) Date: Wed, 9 Sep 2009 09:34:43 -0700 Subject: [Freeswitch-users] [ERR] mod_sofia.c:2645 Invalid Gateway In-Reply-To: References: Message-ID: Because you named your gateway 192.168.72.253, not mediant1000. You could name it mediant1000 and set , or use sofia/gateway/192.168.72.253/... Mathieu Rene Avant-Garde Solutions Inc Office: + 1 (514) 664-1044 x100 Cell: +1 (514) 664-1044 x200 mrene at avgs.ca On 9-Sep-09, at 9:30 AM, Jerry Richards wrote: > I have phones registered internally and can call among them. > However, when > I dial "711" from an internal phone, freeswitch replies with "484 > Address > Incomplete" with reason "INVALID_NUMBER_FORMAT". At the server > console, I > see the following error: > > [ERR] mod_sofia.c:2645 Invalid Gateway > > Does anyone know why I get this error? Is there something more I > must do to > add the gateway below? > > I already added the following to the > usr/local/freesitch/conf/dialplan/default.xml: > > > > > > > > I already created a > usr/local/freeswitch/conf/sip_profiles/external/mediant1000.xml file: > > > > > > > > > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dome at tel.co.th Wed Sep 9 10:17:04 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Thu, 10 Sep 2009 00:17:04 +0700 Subject: [Freeswitch-users] filter in fs_cli Message-ID: <8ccbff060909091017u5c9ec05t8fcf01532618ef2@mail.gmail.com> Dear All, I'm looking for document,example for /filter command. where to get it ? BG Dome C. From tzury.by at reguluslabs.com Wed Sep 9 11:20:48 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Wed, 9 Sep 2009 21:20:48 +0300 Subject: [Freeswitch-users] No audio on caller side when both side support speex/8000 only In-Reply-To: References: <10128ef10909090419g196d2280p5d851728fb8c6309@mail.gmail.com> Message-ID: <10128ef10909091120w560300c0s974ac14b646a8304@mail.gmail.com> > This looks and sounds like a case where pjsip isn't listening to our > SDP. ?If we 200 OK with speex on 102 and the far end starts sending it > on 98 then I suspect the client is broken if I'm not mistaken. > > /b Could be, anyhow, note that this happens only both side using speex/8000. If one party uses a different codec the problem does not exists. Moreover, these same two clients (with speex/8000) works fine when connected to iptel.org. The most concerning fact is that a=rtpmap:98 SPEEX/8000 sent by FS to the callee even though the caller sent a=rtpmap:102 SPEEX/8000. Does this make any sense? From jlenk at frontiernet.net Wed Sep 9 12:57:08 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Wed, 9 Sep 2009 14:57:08 -0500 (CDT) Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> Message-ID: <1252526228140-3613195.post@n2.nabble.com> I think the problem here is that the loader only keeps this method in scope until completion then it drops the remoted connection. Therefore you should not use threads in this method. Michael please correct me if I am wrong here. As an example of the failure simply just put a Sleep(10000) call in the thread and you will see the failure. As Michael said this method was only designed to allow the option to opt out of being loaded. In order to support this perhaps a configuration flag simular to the lua "startup-script" should be added. Here is the error I get with the loop I mentioned. -Josh [image: Capture.PNG] On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo wrote: > Hi, > > > > Can you please elaborate on the crash you receive when you > queue a thread during load? > > > > Thanks, > > Michael > > -- View this message in context: http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html Sent from the freeswitch-users mailing list archive at Nabble.com. From dmitry.bely at gmail.com Wed Sep 9 13:04:39 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Thu, 10 Sep 2009 00:04:39 +0400 Subject: [Freeswitch-users] Skypiax false DTMF event Message-ID: <90823c940909091304j2e2c7ad2m50f80e1e605fb9f9@mail.gmail.com> I have a problem. After 10-20 minutes of Skype talk via cordless phone connected to ATA the latter erroneously generated DTMF 'D' event. Then skypiax looses connection while the call remain active in Skype client. The only way to terminate it is to ask another party to hang up: (...) 2009-09-09 22:20:07.474051 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 5,21] READING: |||CALL 307 DURATION 500||| 2009-09-09 22:20:08.473755 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 5,21] READING: |||CALL 307 DURATION 501||| 2009-09-09 22:20:09.474247 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 5,21] READING: |||CALL 307 DURATION 502||| 2009-09-09 22:20:10.474611 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 5,21] READING: |||CALL 307 DURATION 503||| 2009-09-09 22:20:11.474456 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 5,21] READING: |||CALL 307 DURATION 504||| 2009-09-09 22:20:12.411664 [DEBUG] switch_rtp.c:2239 RTP RECV DTMF D:2000 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:633 rev 14771[(nil)|37 ][DEBUG_SKYPE 633 ][interface1][-1, 5,21] interface1 CHANNEL SEND_DTMF 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:634 rev 14771[(nil)|37 ][DEBUG_SKYPE 634 ][interface1][-1, 5,21] DTMF: D 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:882 rev 14707[(nil)|37 ][DEBUG_SKYPE 882 ][interface1][-1, 5,21] DIGIT received: D 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:1352 rev 14707[(nil)|37 ][DEBUG_SKYPE 1352 ][interface1][-1, 5,21] SENDING: |||SET CALL 307 DTMF D|||| 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:1530 rev 14707[(nil)|37 ][DEBUG_SKYPE 1530 ][interface1][-1, 5,21] Got a 'continue' XAtom without a previous 'begin'. It's value (between vertical bars) is=|||allowed call prop||| 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 5,21] READING: |||ERROR 21 Unknown/dis||| 2009-09-09 22:20:12.411664 [ERR] skypiax_protocol.c:144 rev 14707[(nil)|37 ][ERRORA 144 ][interface1][-1, 5,21] Skype got ERROR: |||ERROR 21 Unknown/dis||| 2009-09-09 22:20:12.411664 [ERR] skypiax_protocol.c:146 rev 14707[(nil)|37 ][ERRORA 146 ][interface1][-1, 5,16] skype_call now is DOWN 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:1011 rev 14771[(nil)|37 ][DEBUG_SKYPE 1011 ][interface1][-1, 1,16] skype call ended 2009-09-09 22:20:12.411664 [NOTICE] mod_skypiax.c:1022 Hangup skypiax/interface1/user2 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-09-09 22:20:12.411664 [DEBUG] switch_channel.c:1715 Send signal skypiax/interface1/user2 [KILL] 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:566 rev 14771[(nil)|37 ][DEBUG_SKYPE 566 ][interface1][-1, 1,16] interface1 CHANNEL KILL_CHANNEL 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:569 rev 14771[(nil)|37 ][DEBUG_SKYPE 569 ][interface1][-1, 1,16] skypiax/interface1/user2 CHANNEL got SWITCH_SIG_KILL 2009-09-09 22:20:12.411664 [DEBUG] switch_core_session.c:932 Send signal skypiax/interface1/user2 [BREAK] 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:566 rev 14771[(nil)|37 ][DEBUG_SKYPE 566 ][interface1][-1, 1,16] interface1 CHANNEL KILL_CHANNEL 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:589 rev 14771[(nil)|37 ][DEBUG_SKYPE 589 ][interface1][-1, 1,16] skypiax/interface1/user2 CHANNEL got SWITCH_SIG_BREAK 2009-09-09 22:20:12.428590 [DEBUG] skypiax_protocol.c:670 rev 14707[(nil)|37 ][DEBUG_SKYPE 670 ][interface1][-1, 1,16] Skype incoming audio GONE 2009-09-09 22:20:12.428590 [DEBUG] mod_skypiax.c:702 rev 14771[(nil)|37 ][DEBUG_SKYPE 702 ][interface1][-1, 1,16] CHANNEL READ FALSE 2009-09-09 22:20:12.428590 [DEBUG] switch_ivr_bridge.c:377 skypiax/interface1/user2 ending bridge by request from read function 2009-09-09 22:20:12.428590 [DEBUG] switch_ivr_bridge.c:452 BRIDGE THREAD DONE [skypiax/interface1/user2] 2009-09-09 22:20:12.428590 [DEBUG] switch_ivr_bridge.c:454 Send signal sofia/internal/1002 at 192.168.121.66 [BREAK] 2009-09-09 22:20:12.428590 [DEBUG] switch_core_state_machine.c:497 (skypiax/interface1/user2) State EXCHANGE_MEDIA going to sleep 2009-09-09 22:20:12.428590 [DEBUG] switch_core_state_machine.c:398 (skypiax/interface1/user2) Running State Change CS_HANGUP 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:434 (skypiax/interface1/user2) State HANGUP 2009-09-09 22:20:12.429654 [DEBUG] mod_skypiax.c:506 rev 14771[(nil)|37 ][DEBUG_SKYPE 506 ][interface1][-1, 1,16] interface1 CHANNEL HANGUP 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:46 skypiax/interface1/user2 Standard HANGUP, cause: NORMAL_CLEARING 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:434 (skypiax/interface1/user2) State HANGUP going to sleep 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:479 (skypiax/interface1/user2) State Change CS_HANGUP -> CS_REPORTING 2009-09-09 22:20:12.429654 [DEBUG] switch_core_session.c:932 Send signal skypiax/interface1/user2 [BREAK] 2009-09-09 22:20:12.429654 [DEBUG] mod_skypiax.c:566 rev 14771[(nil)|37 ][DEBUG_SKYPE 566 ][interface1][-1, 0, 0] interface1 CHANNEL KILL_CHANNEL 2009-09-09 22:20:12.429654 [DEBUG] mod_skypiax.c:589 rev 14771[(nil)|37 ][DEBUG_SKYPE 589 ][interface1][-1, 0, 0] skypiax/interface1/user2 CHANNEL got SWITCH_SIG_BREAK 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:398 (skypiax/interface1/user2) Running State Change CS_REPORTING 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:616 (skypiax/interface1/user2) State REPORTING 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:53 skypiax/interface1/user2 Standard REPORTING, cause: NORMAL_CLEARING 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:616 (skypiax/interface1/user2) State REPORTING going to sleep 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:411 (skypiax/interface1/user2) State Change CS_REPORTING -> CS_DESTROY 2009-09-09 22:20:12.429654 [DEBUG] switch_core_session.c:932 Send signal skypiax/interface1/user2 [BREAK] 2009-09-09 22:20:12.429654 [DEBUG] mod_skypiax.c:566 rev 14771[(nil)|37 ][DEBUG_SKYPE 566 ][interface1][-1, 0, 0] interface1 CHANNEL KILL_CHANNEL 2009-09-09 22:20:12.429654 [DEBUG] mod_skypiax.c:589 rev 14771[(nil)|37 ][DEBUG_SKYPE 589 ][interface1][-1, 0, 0] skypiax/interface1/user2 CHANNEL got SWITCH_SIG_BREAK 2009-09-09 22:20:12.429654 [DEBUG] switch_core_session.c:1068 Session 8 (skypiax/interface1/user2) Locked, Waiting on external entities 2009-09-09 22:20:12.439620 [DEBUG] skypiax_protocol.c:849 rev 14707[(nil)|37 ][DEBUG_SKYPE 849 ][interface1][-1, 0, 0] Skype outbound audio GONE 2009-09-09 22:20:12.458666 [DEBUG] switch_ivr_bridge.c:426 sofia/internal/1002 at 192.168.121.66 receive message [UNBRIDGE] 2009-09-09 22:20:12.458666 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/1002 at 192.168.121.66 [BREAK] 2009-09-09 22:20:12.458666 [DEBUG] switch_ivr_bridge.c:452 BRIDGE THREAD DONE [sofia/internal/1002 at 192.168.121.66 ] 2009-09-09 22:20:12.458666 [DEBUG] switch_ivr_bridge.c:454 Send signal skypiax/interface1/user2 [BREAK] 2009-09-09 22:20:12.458666 [DEBUG] mod_skypiax.c:566 rev 14771[(nil)|37 ][DEBUG_SKYPE 566 ][interface1][-1, 0, 0] interface1 CHANNEL KILL_CHANNEL 2009-09-09 22:20:12.458666 [DEBUG] mod_skypiax.c:589 rev 14771[(nil)|37 ][DEBUG_SKYPE 589 ][interface1][-1, 0, 0] skypiax/interface1/user2 CHANNEL got SWITCH_SIG_BREAK 2009-09-09 22:20:12.458666 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal/1002 at 192.168.121.66 [CS_EXECUTE] [NORMAL_CLEARING] 2009-09-09 22:20:12.458666 [DEBUG] switch_channel.c:1715 Send signal sofia/internal/1002 at 192.168.121.66 [KILL] 2009-09-09 22:20:12.458666 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1002 at 192.168.121.66 [BREAK] 2009-09-09 22:20:12.458666 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1002 at 192.168.121.66) State EXECUTE going to sleep 2009-09-09 22:20:12.458666 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1002 at 192.168.121.66) Running State Change CS_HANGUP 2009-09-09 22:20:12.459723 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/1002 at 192.168.121.66) State HANGUP 2009-09-09 22:20:12.459723 [DEBUG] mod_sofia.c:338 Channel sofia/internal/1002 at 192.168.121.66 hanging up, cause: NORMAL_CLEARING 2009-09-09 22:20:12.459723 [DEBUG] mod_sofia.c:376 Sending BYE to sofia/internal/1002 at 192.168.121.66 (...) 2009-09-09 22:20:12.712461 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 0, 0] READING: |||CALL 307 DURATION 505||| 2009-09-09 22:20:12.712461 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 0, 0] READING: |||CALL 307 VAA_INPUT_STATUS FALSE||| 2009-09-09 22:20:13.473652 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 0, 0] READING: |||CALL 307 DURATION 506||| 2009-09-09 22:20:14.473723 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 0, 0] READING: |||CALL 307 DURATION 507||| (...) 2009-09-09 22:26:47.775793 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 0, 0] READING: |||CALL 307 STATUS FINISHED||| 2009-09-09 22:26:47.775793 [DEBUG] skypiax_protocol.c:361 rev 14707[(nil)|37 ][DEBUG_SKYPE 361 ][interface1][-1, 0, 0] skype_call 307 is NOT MY call, ignoring 2009-09-09 22:48:08.274202 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 0, 0] READING: |||USER user1 ONLINESTATUS OFFLINE||| 2009-09-09 22:48:08.274202 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 0, 0] READING: |||USERSTATUS OFFLINE||| 2009-09-09 22:48:08.274202 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 0, 0] READING: |||CONNSTATUS CONNECTING||| 2009-09-09 22:48:31.374348 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 0, 0] READING: |||USER user1 ONLINESTATUS ONLINE||| 2009-09-09 22:48:31.374348 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 0, 0] READING: |||USERSTATUS ONLINE||| 2009-09-09 22:48:31.375500 [DEBUG] skypiax_protocol.c:104 rev 14707[(nil)|37 ][DEBUG_SKYPE 104 ][interface1][-1, 0, 0] READING: |||CONNSTATUS ONLINE||| Is this expected behavior? Of course, the main problem is probably in hardware but does skypiax do its job right? - Dmitry Bely From msc at freeswitch.org Wed Sep 9 13:36:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Sep 2009 13:36:53 -0700 Subject: [Freeswitch-users] filter in fs_cli In-Reply-To: <8ccbff060909091017u5c9ec05t8fcf01532618ef2@mail.gmail.com> References: <8ccbff060909091017u5c9ec05t8fcf01532618ef2@mail.gmail.com> Message-ID: <87f2f3b90909091336u19fac5fei71ef59a70bd6e990@mail.gmail.com> On Wed, Sep 9, 2009 at 10:17 AM, Dome Charoenyost wrote: > Dear All, > I'm looking for document,example for /filter command. > where to get it ? > This is a handy way to add filters to what you see on the fs_cli. Event sockets allow for filters and the "/filter" command lets you add them to your fs_cli session. Check this page for specifics: http://wiki.freeswitch.org/wiki/Mod_event_socket#filter -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/aeab6979/attachment.html From msc at freeswitch.org Wed Sep 9 13:40:56 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Sep 2009 13:40:56 -0700 Subject: [Freeswitch-users] filter in fs_cli In-Reply-To: <87f2f3b90909091336u19fac5fei71ef59a70bd6e990@mail.gmail.com> References: <8ccbff060909091017u5c9ec05t8fcf01532618ef2@mail.gmail.com> <87f2f3b90909091336u19fac5fei71ef59a70bd6e990@mail.gmail.com> Message-ID: <87f2f3b90909091340p218fe5f7h29c27cdd3d1c30d4@mail.gmail.com> On Wed, Sep 9, 2009 at 1:36 PM, Michael Collins wrote: > > > On Wed, Sep 9, 2009 at 10:17 AM, Dome Charoenyost wrote: > >> Dear All, >> I'm looking for document,example for /filter command. >> where to get it ? >> > > This is a handy way to add filters to what you see on the fs_cli. Event > sockets allow for filters and the "/filter" command lets you add them to > your fs_cli session. > > Check this page for specifics: > http://wiki.freeswitch.org/wiki/Mod_event_socket#filter > > -MC > > Also, I forgot to mention that this is used in conjunction with the "/event' command. Open fs_cli and execute these commands: /log 0 /event plain all At this point you will get no log messages and just events. Now you can filter them as needed. Example: /filter Event-Name CHANNEL_EXECUTE Have fun! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/a606e593/attachment-0001.html From josh at radianttiger.com Wed Sep 9 13:40:56 2009 From: josh at radianttiger.com (Josh Rivers) Date: Wed, 9 Sep 2009 13:40:56 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <63A023F4-1AA8-4A56-8029-91F5A5C74084@avgs.ca> Message-ID: kernel32.dll!77e4bef7() Here's that call stack. [Frames below may be incorrect and/or missing, no symbols loaded for kernel32.dll] kernel32.dll!77e4bef7() msvcr80.dll!78158e89() mscorwks.dll!79e7a17a() mscorwks.dll!79ea0fa8() mscorwks.dll!79ea0eff() mscorwks.dll!79e976cc() mscorwks.dll!79e976b3() mscorwks.dll!79e9e3bd() mscorwks.dll!79e970c8() mscorwks.dll!79f782f1() mscorwks.dll!79eaa5c5() mscorwks.dll!79eaad29() mscorwks.dll!79e9a15d() mscorwks.dll!79e9a15d() mscorwks.dll!79e7a1f1() mscorwks.dll!79e7a1f1() mscorwks.dll!79e7a17a() mscorwks.dll!79e88cca() mscorwks.dll!79e96571() mscorwks.dll!79e965a4() mscorwks.dll!79e965c2() mscorwks.dll!79f59330() mscorwks.dll!79f59492() mscorlib.ni.dll!792d5348() mscorlib.ni.dll!792d514f() mscorlib.ni.dll!792d4fde() mscorlib.ni.dll!79799714() mscorwks.dll!79e813e4() mscorwks.dll!79e813ec() > FreeSwitch.dll!switch_loadable_module_load_file(char * path=0x01181250, char * filename=0x01181240, switch_bool_t global=SWITCH_FALSE, switch_loadable_module * * new_module=0x0012d9e0) Line 846 + 0xd bytes C FreeSwitch.dll!switch_loadable_module_load_module_ex(char * dir=0x003994a8, char * fname=0x01081d59, switch_bool_t runtime=SWITCH_FALSE, switch_bool_t global=SWITCH_FALSE, const char * * err=0x0012da5c) Line 942 + 0x15 bytes C FreeSwitch.dll!switch_loadable_module_init() Line 1174 + 0x23 bytes C FreeSwitch.dll!switch_core_init_and_modload(unsigned int flags=129, switch_bool_t console=SWITCH_TRUE, const char * * err=0x0012fdec) Line 1469 + 0x5 bytes C FreeSwitch.exe!main(int argc=1, char * * argv=0x00394f80) Line 748 + 0x23 bytes C FreeSwitch.exe!__tmainCRTStartup() Line 586 + 0x19 bytes C FreeSwitch.exe!mainCRTStartup() Line 403 C kernel32.dll!77e6f23b() The breakpoint is: status = load_func_ptr(&module_interface, pool); Line 846 in switch_loadable_module.c --Josh On Tue, Sep 8, 2009 at 10:50 PM, Josh Rivers wrote: > I'm running of the binary release, so I don't have debug symbols for the > freeswitch core. I can do a build...but does somebody else already have one > handy? -Josh > > > On Tue, Sep 8, 2009 at 10:33 PM, Mathieu Rene wrote: > >> Click Break, then go in Window, Debug, Stack Trace (or something similar, >> I don't have any VS nearby), then copy paste that. >> Mathieu Rene >> Avant-Garde Solutions Inc >> Office: + 1 (514) 664-1044 x100 >> Cell: +1 (514) 664-1044 x200 >> mrene at avgs.ca >> >> >> >> >> On 8-Sep-09, at 10:30 PM, Josh Rivers wrote: >> >> Here is the error I get with the loop I mentioned. -Josh >> >> >> On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo wrote: >> >>> Hi, >>> >>> >>> Can you please elaborate on the crash you receive when >>> you queue a thread during load? >>> >>> >>> Thanks, >>> >>> Michael >>> >>> >>> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh >>> Rivers >>> *Sent:* Tuesday, September 08, 2009 12:22 AM >>> *To:* freeswitch-users at lists.freeswitch.org >>> *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / >>> .NET >>> >>> >>> Thanks for the response! >>> >>> >>> I have tried putting a long-running loop here, but then it blocks >>> anything else managed from happening: >>> >>> >>> public class TestLoop : ILoadNotificationPlugin >>> >>> { >>> >>> public bool Load() >>> >>> { >>> >>> EventConsumer con = new EventConsumer("all", ""); >>> >>> while (true) >>> >>> { >>> >>> Event ev = con.pop(0); >>> >>> Log.WriteLine(LogLevel.Notice, "Event: " + >>> ev.serialized_string); >>> >>> freeswitch.msleep(100); >>> >>> } >>> >>> } >>> >>> } >>> >>> >>> However, if I fork off a thread here, freeswitch crashes: >>> >>> public class TestLoop : ILoadNotificationPlugin >>> >>> { >>> >>> public bool Load() >>> >>> { >>> >>> ThreadPool.QueueUserWorkItem((o) => >>> >>> { >>> >>> Log.WriteLine(LogLevel.Notice, "Thread Starting. "); >>> >>> EventConsumer con = new EventConsumer("all", ""); >>> >>> while (true) >>> >>> { >>> >>> Event ev = con.pop(0); >>> >>> Log.WriteLine(LogLevel.Notice, "Event: " + >>> ev.serialized_string); >>> >>> freeswitch.msleep(100); >>> >>> } >>> >>> }); >>> >>> return true; >>> >>> } >>> >>> } >>> >>> >>> It doesn't look like this is a good place to start a long-running >>> process? >>> >>> >>> Thanks! >>> >>> Josh >>> >>> >>> On Mon, Sep 7, 2009 at 11:05 PM, Raffaele P. Guidi < >>> raffaele.p.guidi at gmail.com> wrote: >>> >>> Yes! >>> >>> >>> public class LoadDemo : ILoadNotificationPlugin { >>> >>> public bool Load() { >>> >>> Log.WriteLine(LogLevel.Notice, "LoadDemo running."); >>> >>> return true; >>> >>> } >>> >>> } >>> >>> >>> this example is from Michael Giagnocavo's Demo.csx which you can find >>> into the mod_managed svn. >>> >>> >>> And let me add that works like a charm :) >>> >>> >>> Ciao, >>> >>> Raffaele >>> >>> >>> On Sun, Sep 6, 2009 at 22:50, Josh Rivers wrote: >>> >>> Is there a way to start this when FreeSWITCH starts? The lua and perl >>> modules have a 'startup-script' configuration preference. Is there something >>> similar in mod_managed? Or is there a way to have an api command executed at >>> a startup? >>> >>> >>> >>> >>> Exactly what I was after - thank you! >>> >>> >>> On Thu, Sep 3, 2009 at 1:54 PM, Jeff Lenk wrote: >>> >>> >>> > >>> >>> > try something like this >>> >>> > >>> >>> > EventConsumer con = new EventConsumer("all", ""); >>> >>> > Event ev = con.pop(0); >>> >>> > >>> >>> > see lua sample - >>> >>> > http://wiki.freeswitch.org/wiki/Lua#freeswitch.EventConsumer >>> >>> > >>> >>> > >>> >>> > Phillip Jones-2 wrote: >>> >>> > > >>> >>> > > Hi there, >>> >>> > > >>> >>> > > mod_managed exposes EventReceivedFunction such that: >>> >>> > > >>> >>> > > Session.EventReceivedFunction = (e) => >>> >>> > > { >>> >>> > > Log.WriteLine(LogLevel.Alert, "Received Event {0}", >>> e.ToString()); >>> >>> > > return ""; >>> >>> > > }; >>> >>> > > >>> >>> > > should trap all events to which i subscribe. >>> >>> > > >>> >>> > > >>> >>> > > But how do I subscribe to events? What is the .NET / managed >>> equivalent >>> >>> > > of: >>> >>> > > >>> >>> > > switch_event_bind(const char *id, switch_event_types_t event, const >>> char >>> >>> > > *subclass_name, switch_event_callback_t callback, void *user_data); >>> >>> > > >>> >>> > > >>> >>> > > >>> >>> > > Thank you! >>> >>> > > >>> >>> > > >>> >>> > > >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/f4aeee18/attachment-0001.html From msc at freeswitch.org Wed Sep 9 13:54:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Sep 2009 13:54:19 -0700 Subject: [Freeswitch-users] Call Forwarding Question In-Reply-To: References: <7d0bfd8c0909061654r30e6f906o5c9b25f1b27720cc@mail.gmail.com> Message-ID: <87f2f3b90909091354n1caa4953ocd170beedf14774f@mail.gmail.com> On Tue, Sep 8, 2009 at 1:20 PM, Nikolai Geordzhev wrote: > I`ve already tried the legs variable in cdr_csv.conf.xml, I have also tried > to use the loopback endpoint and to bridge the call to the internal > interface(so it can go out and in again generating the 2cdr-s I need) and > still haven`t achieved any success. Can anyone please share some experience > in doing CallForwarding in FreeSwitch. I beleive I`m not the only guy tryiig > to achieve this, what`s the Best Practices for this task? > > Nik, Can you pastebin your dialplan where you do this? I'd like to see what you're doing and perhaps see if I can duplicate your scenario for testing. Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/f8649761/attachment.html From josh at radianttiger.com Wed Sep 9 14:01:09 2009 From: josh at radianttiger.com (Josh Rivers) Date: Wed, 9 Sep 2009 14:01:09 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <1252526228140-3613195.post@n2.nabble.com> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> Message-ID: A new discovery: public bool Load() { ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); if (ev == null) continue; Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); } }); return true; } Does not crash. (Adding the null check prevents crash.) The backgrounded loop runs fine. Looks like the event object goes straight to pinvokes, so a null result just crashes? I like the idea of a 'startup-script' for mod_managed. It would also be excellent if there was an event or message informing the background code to terminate nicely when the module reloads. --Josh On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk wrote: > > I think the problem here is that the loader only keeps this method in scope > until completion then it drops the remoted connection. Therefore you should > not use threads in this method. Michael please correct me if I am wrong > here. > > As an example of the failure simply just put a Sleep(10000) call in the > thread and you will see the failure. > > As Michael said this method was only designed to allow the option to opt > out > of being loaded. > > In order to support this perhaps a configuration flag simular to the lua > "startup-script" should be added. > > > > Here is the error I get with the loop I mentioned. -Josh > [image: Capture.PNG] > > On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo > wrote: > > > Hi, > > > > > > > > Can you please elaborate on the crash you receive when > you > > queue a thread during load? > > > > > > > > Thanks, > > > > Michael > > > > > > -- > View this message in context: > http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/b3c9a271/attachment.html From rob4manhere at gmail.com Wed Sep 9 14:01:30 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Wed, 9 Sep 2009 16:01:30 -0500 Subject: [Freeswitch-users] Mod fax best practices In-Reply-To: <2A6CC956-A18F-4631-BB73-38E9EB6D2123@gmail.com> References: <2A6CC956-A18F-4631-BB73-38E9EB6D2123@gmail.com> Message-ID: Hi all, Made some progress on the success rate of faxes and question about mod_fax default settings that might help others just getting starting. ECM is turned on by default- which proved problematic, at least for my setup. ECM, if I understand correctly, is the Error Correction Mode which allows fax machines to detect errors in small blocks of data on a page (via check sums) and request that just those blocks be re- transmitted (http://en.wikipedia.org/wiki/Error_correction_mode). Unfortunately, I was seeing a build up of resend requests in the debug logs which eventually, on 10+ page faxes, exceeded the limit and caused the fax to fail. Disabling ECM prevented this failure, though, as I understand it, possibly at the expense of quality. Certain blocks could have errors that aren't corrected. I haven't seen any visible loss in quality though and now am transmitting and receiving with a very good rate of success. I guess the next question is why there were so many resend requests. Is that just part of the fax-over-ip territory? Or is there more than can be tuned? Thoughts anyone? Cheers, Rob On Sep 8, 2009, at 10:29 AM, Rob Forman wrote: > Hi all, > > I built a mod_fax setup which is working well. Hats off to the > authors for the module. I receive faxes regularly without issue. I > can also send faxes, but seem to have a higher failure rate, > especially with faxes over 10 pages, usually with error 21: No > response after sending a page. > > I know fax-over-IP can be dicey sometimes (don't know if T.38 would > help but see that mod_fax doesn't support yet anyway), but here's my > question: > > Are there any best practices when using mod_fax? Codecs to use or > avoid, jitter settings, OS tuning, etc? Things that you guys have > learned through live use. Sorry if its a newbie question but I've > read through the documentation and wiki but haven't seen much in this. > > About the system: Dedicated hosted server, Debian 5, Freeswitch > 1.0.4. Nothing else running on the system. > > Any thoughts or lessons learned would be greatly appreciated. > > Cheers, > Rob Forman From msc at freeswitch.org Wed Sep 9 14:23:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Sep 2009 14:23:15 -0700 Subject: [Freeswitch-users] mod_opal segmentation fault error In-Reply-To: <5BC8E502-1F0D-4490-A01A-A2251E3F4C36@gmail.com> References: <5BC8E502-1F0D-4490-A01A-A2251E3F4C36@gmail.com> Message-ID: <87f2f3b90909091423l6eb37df8h23d61bf5af957735@mail.gmail.com> On Tue, Sep 8, 2009 at 8:59 PM, Rogelio Perez wrote: > Hi guys, > > My FS setup was working smoothly with mod_opal enabled until I had to > rebuild everything from scratch. > Now I have compiled everything following the same procedure (I even > have a script for that) and mod_opal stopped working. > The SVN commands are: > > 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 > > ...and the compilation commands follow the documentation isntructions > and there are no output errors. > I start FS with mod_opal disabled and then when I run "load mod_opal"I > get the error: Segmentation fault (core dumped). > The log output shows nothing, and I see there are core.xxxxx files on > the FS directory but I dont know how to read them. > Any ideas? > > Visit these pages for information on how to collect a backtrace and open a JIRA bug report: http://wiki.freeswitch.org/wiki/Reporting_Bugs http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Loading_FreeSWITCH_in_GDB Before you file a bug report make sure that you update FS to latest SVN and also use the buildopal.sh file to rebuild opal and ptlib. Use "make current" to get your FS updated cleanly to the latest SVN and then try opal. If it still segs then open the bug report in the MODOPAL section. ( http://jira.freeswitch.org/browse/MODOPAL) Join us on IRC if you have more questions on all this. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/ffacc629/attachment.html From msc at freeswitch.org Wed Sep 9 14:25:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 9 Sep 2009 14:25:21 -0700 Subject: [Freeswitch-users] Sangoma A500 - dial out from specific port group? In-Reply-To: <8a9b664c0909082331x3bbb5510g970bec0167213ac4@mail.gmail.com> References: <8a9b664c0909082320l3be47aedk8d67c2c83d5b411d@mail.gmail.com> <8a9b664c0909082331x3bbb5510g970bec0167213ac4@mail.gmail.com> Message-ID: <87f2f3b90909091425r22c6409j10fb9670bed4f1f2@mail.gmail.com> What is the output of "oz list" and "oz dump"? Put them in pastebin.freeswitch.org and link here in the mailing list. -MC On Tue, Sep 8, 2009 at 11:31 PM, Vassil Panayotov wrote: > Sorry I hit 'send' by mistake... > > Hi, > > Is it possible to originate calls from specific A500 ports with FreeSWITCH? > I am using a A504 (8 BRI interfaces), and I want some outbound calls to be > made from specific BRI interfaces. > > I tried to modify OpenZAP config as follows: > > conf/openzap.conf > > [span wanpipe boostbri1] > trunk_type => bri > b-channel => 1:1-2 > b-channel => 2:1-2 > b-channel => 3:1-2 > b-channel => 4:1-2 > b-channel => 5:1-2 > b-channel => 6:1-2 > > [span wanpipe boostbri2] > trunk_type => bri > b-channel => 7:1-2 > b-channel => 8:1-2 > > conf/autoload_configs/openzap. > conf.xml: > > > > > > > > > > > > > > > When I try to originate call I am getting errors: > > freeswitch at emo-voip> originate openzap/2/a/123456 music > API CALL [originate(openzap/2/a/123456 music)] output: > -ERR NORMAL_CIRCUIT_CONGESTION > > 2009-09-04 09:23:34.87253 [CRIT] ozmod_ss7_boost.c:244 SPAN is not online. > 2009-09-04 09:23:34.87253 [ERR] mod_openzap.c:1043 No channels available > 2009-09-04 09:23:34.87253 [ERR] switch_ivr_originate.c:1510 Cannot create > outgoing channel of type [openzap] cause: [NORMAL_CIRCUIT_CONGESTION] > > Then I tried to modify the /etc/wanpipe/smg_bri.conf: > > ;Sangoma AFT-A500 port 11 [slot:8 bus:1 span:7] > group=2 > country=europe > operator=etsi > connection_type=point_to_point > signalling=bri_nt > spans=7 > > ;Sangoma AFT-A500 port 12 [slot:8 bus:1 span:8] > group=2 > country=europe > operator=etsi > connection_type=point_to_point > signalling=bri_nt > spans=8 > > i.e. changed 'group' to 2, but this doesn't help either. > > Marc Celsie from Sangoma's techdesk told me that I should ' dial X at gY with > X being the number and Y being the group number'. > > How originate command should look like in this case? > originate openzap/1/a/123456 at g2 someExt ? > > I tried this syntax but with no effect. Marc also told me that there is a > bug in FS which prevents groups from working. > > Should I fill bug report or feature request? > > Best regards, > Vassil Panayotov > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/44bcaf01/attachment-0001.html From jlenk at frontiernet.net Wed Sep 9 14:38:33 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Wed, 9 Sep 2009 16:38:33 -0500 (CDT) Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> Message-ID: <1252532313906-3614845.post@n2.nabble.com> Yeah I noticed that but the thread was still terminating after a few seconds anyway for me. Does it stay running for you? Josh Rivers-2 wrote: > > A new discovery: public bool Load() > { > ThreadPool.QueueUserWorkItem((o) => > { > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > EventConsumer con = new EventConsumer("all", ""); > while (true) > { > Event ev = con.pop(0); > if (ev == null) continue; > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > } > }); > return true; > } > Does not crash. (Adding the null check prevents crash.) The backgrounded > loop runs fine. Looks like the event object goes straight to pinvokes, so > a > null result just crashes? > > I like the idea of a 'startup-script' for mod_managed. It would also be > excellent if there was an event or message informing the background code > to > terminate nicely when the module reloads. > > --Josh > > On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk wrote: > >> >> I think the problem here is that the loader only keeps this method in >> scope >> until completion then it drops the remoted connection. Therefore you >> should >> not use threads in this method. Michael please correct me if I am wrong >> here. >> >> As an example of the failure simply just put a Sleep(10000) call in the >> thread and you will see the failure. >> >> As Michael said this method was only designed to allow the option to opt >> out >> of being loaded. >> >> In order to support this perhaps a configuration flag simular to the lua >> "startup-script" should be added. >> >> >> >> Here is the error I get with the loop I mentioned. -Josh >> [image: Capture.PNG] >> >> On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo >> wrote: >> >> > Hi, >> > >> > >> > >> > Can you please elaborate on the crash you receive when >> you >> > queue a thread during load? >> > >> > >> > >> > Thanks, >> > >> > Michael >> > >> > >> >> -- >> View this message in context: >> http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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/Subscribing-to-events-in-managed-C-NET-tp3573619p3614845.html Sent from the freeswitch-users mailing list archive at Nabble.com. From moises.silva at gmail.com Wed Sep 9 15:07:34 2009 From: moises.silva at gmail.com (Moises Silva) Date: Wed, 9 Sep 2009 18:07:34 -0400 Subject: [Freeswitch-users] Sangoma A500 - dial out from specific port group? In-Reply-To: <87f2f3b90909091425r22c6409j10fb9670bed4f1f2@mail.gmail.com> References: <8a9b664c0909082320l3be47aedk8d67c2c83d5b411d@mail.gmail.com> <8a9b664c0909082331x3bbb5510g970bec0167213ac4@mail.gmail.com> <87f2f3b90909091425r22c6409j10fb9670bed4f1f2@mail.gmail.com> Message-ID: > > Hi, >> >> Is it possible to originate calls from specific A500 ports with >> FreeSWITCH? >> I am using a A504 (8 BRI interfaces), and I want some outbound calls to be >> made from specific BRI interfaces. >> > Hello Vassil, Unless you are using openzap trunk (and that probably means FreeSWITCH trunk as well) this is unlikely to work. Revision 825 in openzap should fix the bug in trunk group selection. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/577084ef/attachment.html From josh at radianttiger.com Wed Sep 9 15:10:43 2009 From: josh at radianttiger.com (Josh Rivers) Date: Wed, 9 Sep 2009 15:10:43 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <1252532313906-3614845.post@n2.nabble.com> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> <1252532313906-3614845.post@n2.nabble.com> Message-ID: It does from a fresh start of FreeSWITCH. I've noticed, although not really confirmed, a race condition between the unload and reload of managed code. It seems that threads started in the newly submodule are terminated along with the threads for the old, unloading submodule. Is that what you are seeing as well? On Wed, Sep 9, 2009 at 2:38 PM, Jeff Lenk wrote: > > Yeah I noticed that but the thread was still terminating after a few > seconds > anyway for me. Does it stay running for you? > > > Josh Rivers-2 wrote: > > > > A new discovery: public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => > > { > > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > if (ev == null) continue; > > Log.WriteLine(LogLevel.Notice, "Event: " + > > ev.serialized_string); > > } > > }); > > return true; > > } > > Does not crash. (Adding the null check prevents crash.) The backgrounded > > loop runs fine. Looks like the event object goes straight to pinvokes, so > > a > > null result just crashes? > > > > I like the idea of a 'startup-script' for mod_managed. It would also be > > excellent if there was an event or message informing the background code > > to > > terminate nicely when the module reloads. > > > > --Josh > > > > On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk > wrote: > > > >> > >> I think the problem here is that the loader only keeps this method in > >> scope > >> until completion then it drops the remoted connection. Therefore you > >> should > >> not use threads in this method. Michael please correct me if I am wrong > >> here. > >> > >> As an example of the failure simply just put a Sleep(10000) call in the > >> thread and you will see the failure. > >> > >> As Michael said this method was only designed to allow the option to opt > >> out > >> of being loaded. > >> > >> In order to support this perhaps a configuration flag simular to the lua > >> "startup-script" should be added. > >> > >> > >> > >> Here is the error I get with the loop I mentioned. -Josh > >> [image: Capture.PNG] > >> > >> On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo > >> wrote: > >> > >> > Hi, > >> > > >> > > >> > > >> > Can you please elaborate on the crash you receive when > >> you > >> > queue a thread during load? > >> > > >> > > >> > > >> > Thanks, > >> > > >> > Michael > >> > > >> > > >> > >> -- > >> View this message in context: > >> > http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html > >> Sent from the freeswitch-users mailing list archive at Nabble.com. > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at 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/Subscribing-to-events-in-managed-C-NET-tp3573619p3614845.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/b8ed69ce/attachment.html From tacvbo at tacvbo.net Wed Sep 9 15:58:25 2009 From: tacvbo at tacvbo.net (Octavio Ruiz) Date: Wed, 9 Sep 2009 17:58:25 -0500 Subject: [Freeswitch-users] Sangoma A500 - dial out from specific port group? In-Reply-To: <8a9b664c0909082320l3be47aedk8d67c2c83d5b411d@mail.gmail.com> References: <8a9b664c0909082320l3be47aedk8d67c2c83d5b411d@mail.gmail.com> Message-ID: On Wed, Sep 9, 2009 at 01:20, Vassil Panayotov wrote: > Hi, > > Is it possible to originate calls from specific A500 ports with FreeSWITCH? > I am using a A504 (8 BRI interfaces), and I want some outbound calls to be > made from specific BRI interfaces. You can't define several spans in openzap.conf for boost, the sangoma_brid config file is where you define groups, so your config should look like this: /// smg_bri.conf ...... group=1 spans=1 group=2 spans=2 group=3 spans=3 ...... /// openzap.conf [span wanpipe BoostBRI] trunk_type => bri b-channel => 1:1-2 b-channel => 2:1-2 b-channel => 3:1-2 b-channel => 4:1-2 b-channel => 5:1-2 b-channel => 6:1-2 b-channel => 7:1-2 b-channel => 8:1-2 /// openzap.conf.xml Then, you can Dial to your span/group number 3 with: freeswitch> originate openzap/1/a/12345 at g3 |&() freeswitch> originate openzap/1/a/12345 at G3 |&() freeswitch> originate openzap/1/a/12345 at r3 |&() freeswitch> originate openzap/1/a/12345 at R3 |&() If you are using FS 1.0.4, there is a bug, you can fix it with this -already in trunk- patch. Index: src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c =================================================================== --- libs/openzap/src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c.orig +++ libs/openzap/src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c @@ -282,6 +282,8 @@ } ss7bc_call_init(&event, caller_data->cid_num.digits, ani, r); + //ss7_bc_call_init will clear the trunk_group val so we need to set it again + event.trunk_group=tg; if (gr && *(gr+1)) { Best regards, -- Octavio H. Ruiz Cervera Tel.: (+52 55) 8590-9000 Ext. 7016 Mobile: (+52 1 55) 4358-4565 Sent from Mexico City, DF, Mexico From mgg at giagnocavo.net Wed Sep 9 16:50:59 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Wed, 9 Sep 2009 19:50:59 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <1252526228140-3613195.post@n2.nabble.com> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DB2B5981@mse17be1.mse17.exchange.ms> The ILoadNotifcationPlugin is run in the appdomain created for the plugin, so it should only get unloaded when the plugin gets reloaded. Spawning threads here should work, it's definitely the intention that if you need a long-running process, you can fire it up on load and have it work. As to the race condition on reload, mod_managed should do this: - Load the new plugin into a new appdomain - Remove the entry points to the old appdomain, add entries to the new one - Old appdomain now stays alive until foreground API and APP calls finish So, you can have many versions of the same plugin active in memory. I probably need to go break compatibility and make ILoadWhateverPlugin be something like IPluginController and allow it to return loading options to control the mod_managed behavior, as well as allow it to delay shutdown of the appdomain. Part of the question is: how many people out there need compatibility, or can we go breaking all of you and make you recompile? :) Although, IIRC, if you handle AppDomain.Unload (or whatever it is), it will stay alive until your event handler completes. Hope that helps a bit. -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Wednesday, September 09, 2009 1:57 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET I think the problem here is that the loader only keeps this method in scope until completion then it drops the remoted connection. Therefore you should not use threads in this method. Michael please correct me if I am wrong here. As an example of the failure simply just put a Sleep(10000) call in the thread and you will see the failure. As Michael said this method was only designed to allow the option to opt out of being loaded. In order to support this perhaps a configuration flag simular to the lua "startup-script" should be added. Here is the error I get with the loop I mentioned. -Josh [image: Capture.PNG] On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo wrote: > Hi, > > > > Can you please elaborate on the crash you receive when you > queue a thread during load? > > > > Thanks, > > Michael > > -- View this message in context: http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From mgg at giagnocavo.net Wed Sep 9 16:51:54 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Wed, 9 Sep 2009 19:51:54 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DB2B5982@mse17be1.mse17.exchange.ms> >Looks like the event object goes straight to pinvokes, so a null result just crashes? If it's null, you should get a NullReferenceException. The C# compiler should callvirt the property getter and that'll do a null check. If that isn't happening, that'd be an interesting optimization somewhere along the line. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Wednesday, September 09, 2009 3:01 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET A new discovery: public bool Load() { ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); if (ev == null) continue; Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); } }); return true; } Does not crash. (Adding the null check prevents crash.) The backgrounded loop runs fine. Looks like the event object goes straight to pinvokes, so a null result just crashes? I like the idea of a 'startup-script' for mod_managed. It would also be excellent if there was an event or message informing the background code to terminate nicely when the module reloads. --Josh On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk > wrote: I think the problem here is that the loader only keeps this method in scope until completion then it drops the remoted connection. Therefore you should not use threads in this method. Michael please correct me if I am wrong here. As an example of the failure simply just put a Sleep(10000) call in the thread and you will see the failure. As Michael said this method was only designed to allow the option to opt out of being loaded. In order to support this perhaps a configuration flag simular to the lua "startup-script" should be added. Here is the error I get with the loop I mentioned. -Josh [image: Capture.PNG] On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo >wrote: > Hi, > > > > Can you please elaborate on the crash you receive when you > queue a thread during load? > > > > Thanks, > > Michael > > -- View this message in context: http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/37439aab/attachment.html From josh at radianttiger.com Wed Sep 9 17:38:57 2009 From: josh at radianttiger.com (Josh Rivers) Date: Wed, 9 Sep 2009 17:38:57 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DB2B5982@mse17be1.mse17.exchange.ms> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5982@mse17be1.mse17.exchange.ms> Message-ID: I have a new thought on the crashes...I'm able to crash FreeSWITCH any time I like, just by having an exception in a thread. public class CrashFreeSWITCH : ILoadNotificationPlugin { public bool Load() { ThreadPool.QueueUserWorkItem((o) => { throw new NotImplementedException(); }); return true; } } Perhaps Application.ThreadException or AppDomain.UnhandledException need to be trapped? On Wed, Sep 9, 2009 at 4:51 PM, Michael Giagnocavo wrote: > >Looks like the event object goes straight to pinvokes, so a null result > just crashes? > > > > If it?s null, you should get a NullReferenceException. The C# compiler > should callvirt the property getter and that?ll do a null check. If that > isn?t happening, that?d be an interesting optimization somewhere along the > line. > > > > -Michael > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers > *Sent:* Wednesday, September 09, 2009 3:01 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > A new discovery: > > public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => > > { > > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > if (ev == null) continue; > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > } > > }); > > return true; > > } > > Does not crash. (Adding the null check prevents crash.) The backgrounded > loop runs fine. Looks like the event object goes straight to pinvokes, so a > null result just crashes? > > > > I like the idea of a 'startup-script' for mod_managed. It would also be > excellent if there was an event or message informing the background code to > terminate nicely when the module reloads. > > > > --Josh > > > > On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk wrote: > > > I think the problem here is that the loader only keeps this method in scope > until completion then it drops the remoted connection. Therefore you should > not use threads in this method. Michael please correct me if I am wrong > here. > > As an example of the failure simply just put a Sleep(10000) call in the > thread and you will see the failure. > > As Michael said this method was only designed to allow the option to opt > out > of being loaded. > > In order to support this perhaps a configuration flag simular to the lua > "startup-script" should be added. > > > > > Here is the error I get with the loop I mentioned. -Josh > [image: Capture.PNG] > > On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo > wrote: > > > Hi, > > > > > > > > Can you please elaborate on the crash you receive when > you > > queue a thread during load? > > > > > > > > Thanks, > > > > Michael > > > > > > -- > View this message in context: > http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/fe257f72/attachment-0001.html From mgg at giagnocavo.net Wed Sep 9 18:30:05 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Wed, 9 Sep 2009 21:30:05 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5982@mse17be1.mse17.exchange.ms> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DB2B598E@mse17be1.mse17.exchange.ms> That's by design. If a thread fails, and there's no handler, then the application could be in a corrupted state, so the CLR takes down the process. I think there is a .NET 1.0 compat switch you can enable in the config if you like exceptions to be silently ignored :). -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Wednesday, September 09, 2009 6:39 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET I have a new thought on the crashes...I'm able to crash FreeSWITCH any time I like, just by having an exception in a thread. public class CrashFreeSWITCH : ILoadNotificationPlugin { public bool Load() { ThreadPool.QueueUserWorkItem((o) => { throw new NotImplementedException(); }); return true; } } Perhaps Application.ThreadException or AppDomain.UnhandledException need to be trapped? On Wed, Sep 9, 2009 at 4:51 PM, Michael Giagnocavo > wrote: >Looks like the event object goes straight to pinvokes, so a null result just crashes? If it's null, you should get a NullReferenceException. The C# compiler should callvirt the property getter and that'll do a null check. If that isn't happening, that'd be an interesting optimization somewhere along the line. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Wednesday, September 09, 2009 3:01 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET A new discovery: public bool Load() { ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); if (ev == null) continue; Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); } }); return true; } Does not crash. (Adding the null check prevents crash.) The backgrounded loop runs fine. Looks like the event object goes straight to pinvokes, so a null result just crashes? I like the idea of a 'startup-script' for mod_managed. It would also be excellent if there was an event or message informing the background code to terminate nicely when the module reloads. --Josh On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk > wrote: I think the problem here is that the loader only keeps this method in scope until completion then it drops the remoted connection. Therefore you should not use threads in this method. Michael please correct me if I am wrong here. As an example of the failure simply just put a Sleep(10000) call in the thread and you will see the failure. As Michael said this method was only designed to allow the option to opt out of being loaded. In order to support this perhaps a configuration flag simular to the lua "startup-script" should be added. Here is the error I get with the loop I mentioned. -Josh [image: Capture.PNG] On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo >wrote: > Hi, > > > > Can you please elaborate on the crash you receive when you > queue a thread during load? > > > > Thanks, > > Michael > > -- View this message in context: http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/9cc2a4c8/attachment.html From grae at digilord.net Wed Sep 9 20:54:36 2009 From: grae at digilord.net (Dan) Date: Wed, 09 Sep 2009 20:54:36 -0700 Subject: [Freeswitch-users] OpenZAP No Audio In Outbound FXO for 8-10 Seconds Message-ID: <4AA8787C.2050105@digilord.net> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello all! I am having the following issue. When I dial out over a FXO port (analog) for the first 8-10 seconds I get no audio. If I wait I will eventually hear something. On inbound calls audio works great in both directions. I used ztmonitor to record from the channel and there was in fact audio there. In the recording you can clearly tell when audio starts flowing as you can hear me twice in the recording (I was calling to my cell from a phone on my desk). Any help here would be greatly appreciated! FS is current SVN OpenZAP is current SVN. Hardware: Rhino 8 port w/6 FXO and 2 FXS ports installed. Daniel Morrigan "He who says it cannot be done is interrupting the one doing it." - Chinese Proverb -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkqoeHoACgkQ3JaPN6smlEXhdQCfRf5GBKwVrtZWsCS4J1fug2e7 SEEAn1FkyhBPKiXnfUiXgvd2ggqUW87w =OFH/ -----END PGP SIGNATURE----- From josh at radianttiger.com Wed Sep 9 21:41:17 2009 From: josh at radianttiger.com (Josh Rivers) Date: Wed, 9 Sep 2009 21:41:17 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DB2B598E@mse17be1.mse17.exchange.ms> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5982@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B598E@mse17be1.mse17.exchange.ms> Message-ID: The question is whether the CLR should take down the whole phone server due to an unhandled exception...definitely the CLR should terminate...but shouldn't it just log the exception to the console, not crash the core? On Wed, Sep 9, 2009 at 6:30 PM, Michael Giagnocavo wrote: > That?s by design. If a thread fails, and there?s no handler, then the > application could be in a corrupted state, so the CLR takes down the > process. > > > > I think there is a .NET 1.0 compat switch you can enable in the config if > you like exceptions to be silently ignored J. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers > *Sent:* Wednesday, September 09, 2009 6:39 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > I have a new thought on the crashes...I'm able to crash FreeSWITCH any time > I like, just by having an exception in a thread. > > > > public class CrashFreeSWITCH : ILoadNotificationPlugin > > { > > public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => { throw new > NotImplementedException(); }); > > return true; > > } > > } > > > > Perhaps Application.ThreadException or AppDomain.UnhandledException need > to be trapped? > > > > On Wed, Sep 9, 2009 at 4:51 PM, Michael Giagnocavo > wrote: > > >Looks like the event object goes straight to pinvokes, so a null result > just crashes? > > > > If it?s null, you should get a NullReferenceException. The C# compiler > should callvirt the property getter and that?ll do a null check. If that > isn?t happening, that?d be an interesting optimization somewhere along the > line. > > > > -Michael > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers > *Sent:* Wednesday, September 09, 2009 3:01 PM > > > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > A new discovery: > > public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => > > { > > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > if (ev == null) continue; > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > } > > }); > > return true; > > } > > Does not crash. (Adding the null check prevents crash.) The backgrounded > loop runs fine. Looks like the event object goes straight to pinvokes, so a > null result just crashes? > > > > I like the idea of a 'startup-script' for mod_managed. It would also be > excellent if there was an event or message informing the background code to > terminate nicely when the module reloads. > > > > --Josh > > > > On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk wrote: > > > I think the problem here is that the loader only keeps this method in scope > until completion then it drops the remoted connection. Therefore you should > not use threads in this method. Michael please correct me if I am wrong > here. > > As an example of the failure simply just put a Sleep(10000) call in the > thread and you will see the failure. > > As Michael said this method was only designed to allow the option to opt > out > of being loaded. > > In order to support this perhaps a configuration flag simular to the lua > "startup-script" should be added. > > > > > Here is the error I get with the loop I mentioned. -Josh > [image: Capture.PNG] > > On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo > wrote: > > > Hi, > > > > > > > > Can you please elaborate on the crash you receive when > you > > queue a thread during load? > > > > > > > > Thanks, > > > > Michael > > > > > > -- > View this message in context: > http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/0fb843b9/attachment-0001.html From mike at jerris.com Wed Sep 9 22:21:26 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 10 Sep 2009 01:21:26 -0400 Subject: [Freeswitch-users] Mod_fifo posision in queue In-Reply-To: <8ccbff060909010737v4a8f17ep4d6175c2a2a2f866@mail.gmail.com> References: <8ccbff060909010737v4a8f17ep4d6175c2a2a2f866@mail.gmail.com> Message-ID: You can use a phrase macro but I am not sure that we set the position in a way that you can expand it for the macro. Mike On Sep 1, 2009, at 10:37 AM, Dome Charoenyost wrote: > 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. From ahmedmunir007 at gmail.com Wed Sep 9 22:16:24 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Thu, 10 Sep 2009 11:16:24 +0600 Subject: [Freeswitch-users] Implementing h extension in FS Message-ID: HI, I'm newbie in FS, I want to know how to implement h extension of asterisk to FS. As I listed down below; h => { NOOP("Call Completed with Carrier ${CARRIER}"); goto add_cdr|h|1; }; My other question is, which application/function/class is use in mod_perl to check the channel status? i.e. busy, answer,hangup,ringing,etc. Kindly advice me soon. -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/35adc306/attachment.html From mike at jerris.com Wed Sep 9 22:26:17 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 10 Sep 2009 01:26:17 -0400 Subject: [Freeswitch-users] outbould PHP ESL In-Reply-To: <8ccbff060909020843v425cea9ey45fce1d56154fbf5@mail.gmail.com> References: <8ccbff060909020539n7931ca6av6f7d9bab5ee9cdf4@mail.gmail.com> <4A9E7022.4000800@telemaque.fr> <8ccbff060909020756t167cd564qde77e849227a4f7@mail.gmail.com> <87f2f3b90909020825x516c2054l7a54dc6a1b61691c@mail.gmail.com> <8ccbff060909020843v425cea9ey45fce1d56154fbf5@mail.gmail.com> Message-ID: It should be the same, except using php syntax instead of perl. Mike On Sep 2, 2009, at 11:43 AM, Dome Charoenyost wrote: > 2009/9/2 Michael Collins : >> Are you trying to get a channel variable or capture DTMF input from >> the >> caller? > > i try to make IVR by php outbound socket. in XML dialplan we can get > DTMF by read application (store in channel variable) > I found it's success in perl outbound (IVR.pm) but for php how do i ? > > > Dome C. > >> -MC >> >> On Wed, Sep 2, 2009 at 7:56 AM, Dome Charoenyost >> wrote: >>> >>> I follow >>> http://wiki.freeswitch.org/wiki/PHP_ESL#ivrd >>> >>> how to get from php ? >>> >>> >>> Dome C. >>> ------------------------------------------------------ >>> #!/usr/bin/php -q >>> >>> >> >>> // set a couple of things so we dont kill the system >>> ob_implicit_flush(true); >>> set_time_limit(30); >>> >>> // Open stdin so we can read the AGI data in >>> $in = fopen("php://stdin", "r"); >>> >>> // Connect >>> echo "connect\n\n"; >>> >>> // Answer >>> echo "sendmsg\n"; >>> echo "call-command: execute\n"; >>> echo "execute-app-name: answer\n\n"; >>> >>> echo "sendmsg\n"; >>> echo "call-command: execute\n"; >>> echo "execute-app-name: read\n"; >>> echo "execute-app-arg: 0 20 >>> /opt/freeswitch/sounds/th/tuxza/welcome.wav res 5000 #\n\n"; >>> >>> // Wait >>> sleep(5); >>> >>> // Hangup >>> echo "sendmsg\n"; >>> echo "call-command: hangup\n\n"; >>> >>> fclose($in); >>> >>> ?> >>> >>> >>> 2009/9/2 Brian West : >>>> uuid_getvar >>>> >>>> /b >>>> >>>> On Sep 2, 2009, at 8:16 AM, Tristan Mah? wrote: >>>> >>>>> Hi, >>>>> >>>>> just a fast 2cent: >>>>> >>>>> get var via channel status ? ( variable_res ) >>>> From mike at jerris.com Wed Sep 9 22:33:48 2009 From: mike at jerris.com (Michael Jerris) Date: Thu, 10 Sep 2009 01:33:48 -0400 Subject: [Freeswitch-users] "make install" failure on Solaris 10 In-Reply-To: <4A9F9DAA.8060702@blueface.ie> References: <4A9F9DAA.8060702@blueface.ie> Message-ID: somewhere in that mess of my commands solaris is not liking something. I tested this a lot on solaris and had it working on every box i was in, so not sure what this could be. If you can get me into a box in this state via ssh I can take a look. Mike On Sep 3, 2009, at 6:42 AM, Bruce McAlister wrote: > Hi, > > I have just managed to complete a build of FreeSWITCH 1.0.4 on > Solaris 10. The problem I am now having is that it fails on the > "make install" part of the installation. > > I have attached the complete output of the "make install". > > A snippet of the failure is below: > > --- > Installing freeswitch > *** Error code 1 > The following command caused the error: > for htdocsfile in `find htdocs -name \* | grep -v .svn` ; do \ > dir=`echo $htdocsfile | sed -e 's|/[^/]*$||'`; \ > filename=`echo $htdocsfile | sed -e 's|^.*/||'`; \ > test -d /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/ > freeswitch/$dir || /export/home/user/packages/BUILD/freeswitch-1.0.4/ > build/config/install-sh -d /var/tmp/pkgbuild-user/freeswitch-1.0.4- > build/opt/freeswitch/$dir ; \ > test -f /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/ > freeswitch/$dir/$filename || /opt/jdsbld/bin/ginstall -c -m 644 $dir/ > $filename /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/ > freeswitch/$dir 2>/dev/null; \ > done > make: Fatal error: Command failed for target `samples-htdocs' > Current working directory /export/home/user/packages/BUILD/ > freeswitch-1.0.4 > *** Error code 1 > The following command caused the error: > test -d /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/ > htdocs || make samples-htdocs > make: Fatal error: Command failed for target `install-data-local' > Current working directory /export/home/user/packages/BUILD/ > freeswitch-1.0.4 > *** Error code 1 > The following command caused the error: > make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed - > e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i > in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" > "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed - > e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i > in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" > "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v > "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf > | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$ > (for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" > "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v > "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf > | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$ > (for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" > "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/ > packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed - > e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo > $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$ > (tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/ > freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | > sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; > done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$ > (grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/ > modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; > mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo > $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep > -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in > $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" > || echo -s` install-exec-am install-data-am > make: Fatal error: Command failed for target `install-am' > Current working directory /export/home/user/packages/BUILD/ > freeswitch-1.0.4 > *** Error code 1 > The following command caused the error: > failcom='exit 1'; \ > for f in x $MAKEFLAGS; do \ > case $f in \ > *=* | --[!k]*);; \ > *k*) failcom='fail=yes';; \ > esac; \ > done; \ > dot_seen=no; \ > target=`echo install-recursive | sed s/-recursive//`; \ > list='. src build'; for subdir in $list; do \ > echo "Making $target in $subdir"; \ > if test "$subdir" = "."; then \ > dot_seen=yes; \ > local_target="$target-am"; \ > else \ > local_target="$target"; \ > fi; \ > (cd $subdir && make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$ > (grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/ > modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else > tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; > done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then > tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/ > freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; > else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i- > clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z > "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/ > freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; > else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i- > install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z > "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/ > freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; > else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i- > uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$ > (grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/ > modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; > mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" > "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "\#" /export/home/ > user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | > sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do > echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES= > $(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/ > freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | > sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; > done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$ > (grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/ > modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; > mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo > $mods )" `test -n "" || echo -s` $local_target) \ > || eval $failcom; \ > done; \ > if test "$dot_seen" = "no"; then \ > make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed - > e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i > in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" > "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed - > e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i > in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" > "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v > "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf > | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$ > (for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" > "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v > "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf > | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$ > (for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" > "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/ > packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed - > e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo > $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$ > (tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/ > freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | > sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; > done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$ > (grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/ > modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; > mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo > $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep > -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in > $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" > || echo -s` "$target-am" || exit 1; \ > fi; test -z "$fail" > make: Fatal error: Command failed for target `install-recursive' > Current working directory /export/home/user/packages/BUILD/ > freeswitch-1.0.4 > *** Error code 1 > make: Fatal error: Command failed for target `install' > > --- > > Any pointers would be greatly appreciated. > > Thanks > Bruce > make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed - > e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i > in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" > "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed - > e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i > in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" > "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v > "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf > | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$ > (for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" > "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v > "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf > | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$ > (for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" > "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/ > packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed - > e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo > $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$ > (tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/ > freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | > sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; > done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$ > (grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/ > modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; > mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo > $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep > -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in > $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" > || echo -s` install-recursive > Making install in . > /bin/sh /export/home/user/packages/BUILD/freeswitch-1.0.4/ > quiet_libtool --mode=install /opt/jdsbld/bin/ginstall -c > 'libfreeswitch.la' '/var/tmp/pkgbuild-user/freeswitch-1.0.4-build/ > opt/freeswitch/lib/libfreeswitch.la' > /opt/jdsbld/bin/ginstall -c .libs/libfreeswitch.so.1.0.0 /var/tmp/ > pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/lib/ > libfreeswitch.so.1.0.0 > (cd /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/lib > && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so.1 || { rm -f > libfreeswitch.so.1 && ln -s libfreeswitch.so.1.0.0 libfreeswitch.so. > 1; }; }) > (cd /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/lib > && { ln -s -f libfreeswitch.so.1.0.0 libfreeswitch.so || { rm -f > libfreeswitch.so && ln -s libfreeswitch.so.1.0.0 > libfreeswitch.so; }; }) > chmod +x /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/ > freeswitch/lib/libfreeswitch.so.1.0.0 > /opt/jdsbld/bin/ginstall -c .libs/libfreeswitch.lai /var/tmp/ > pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/lib/ > libfreeswitch.la > /opt/jdsbld/bin/ginstall -c .libs/libfreeswitch.a /var/tmp/pkgbuild- > user/freeswitch-1.0.4-build/opt/freeswitch/lib/libfreeswitch.a > chmod 644 /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/ > freeswitch/lib/libfreeswitch.a > ranlib /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/ > lib/libfreeswitch.a > quiet_libtool: install: warning: remember to run `quiet_libtool -- > finish /opt/freeswitch/lib' > /bin/sh /export/home/user/packages/BUILD/freeswitch-1.0.4/ > quiet_libtool --mode=install /opt/jdsbld/bin/ginstall -c > 'freeswitch' '/var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/ > freeswitch/bin/freeswitch' > quiet_libtool: install: warning: `libfreeswitch.la' has not been > installed in `/opt/freeswitch/lib' > /opt/jdsbld/bin/ginstall -c .libs/freeswitch /var/tmp/pkgbuild-user/ > freeswitch-1.0.4-build/opt/freeswitch/bin/freeswitch > /bin/sh /export/home/user/packages/BUILD/freeswitch-1.0.4/ > quiet_libtool --mode=install /opt/jdsbld/bin/ginstall -c 'fs_cli' '/ > var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/bin/ > fs_cli' > /opt/jdsbld/bin/ginstall -c fs_cli /var/tmp/pkgbuild-user/ > freeswitch-1.0.4-build/opt/freeswitch/bin/fs_cli > /bin/sh /export/home/user/packages/BUILD/freeswitch-1.0.4/ > quiet_libtool --mode=install /opt/jdsbld/bin/ginstall -c 'fs_ivrd' '/ > var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/bin/ > fs_ivrd' > /opt/jdsbld/bin/ginstall -c fs_ivrd /var/tmp/pkgbuild-user/ > freeswitch-1.0.4-build/opt/freeswitch/bin/fs_ivrd > /opt/jdsbld/bin/ginstall -c 'scripts/gentls_cert' '/var/tmp/pkgbuild- > user/freeswitch-1.0.4-build/opt/freeswitch/bin/gentls_cert' > /opt/jdsbld/bin/ginstall -c 'scripts/fsxs' '/var/tmp/pkgbuild-user/ > freeswitch-1.0.4-build/opt/freeswitch/bin/fsxs' > Installing freeswitch > *** Error code 1 > The following command caused the error: > for htdocsfile in `find htdocs -name \* | grep -v .svn` ; do \ > dir=`echo $htdocsfile | sed -e 's|/[^/]*$||'`; \ > filename=`echo $htdocsfile | sed -e 's|^.*/||'`; \ > test -d /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/ > freeswitch/$dir || /export/home/user/packages/BUILD/freeswitch-1.0.4/ > build/config/install-sh -d /var/tmp/pkgbuild-user/freeswitch-1.0.4- > build/opt/freeswitch/$dir ; \ > test -f /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/ > freeswitch/$dir/$filename || /opt/jdsbld/bin/ginstall -c -m 644 $dir/ > $filename /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/ > freeswitch/$dir 2>/dev/null; \ > done > make: Fatal error: Command failed for target `samples-htdocs' > Current working directory /export/home/user/packages/BUILD/ > freeswitch-1.0.4 > *** Error code 1 > The following command caused the error: > test -d /var/tmp/pkgbuild-user/freeswitch-1.0.4-build/opt/freeswitch/ > htdocs || make samples-htdocs > make: Fatal error: Command failed for target `install-data-local' > Current working directory /export/home/user/packages/BUILD/ > freeswitch-1.0.4 > *** Error code 1 > The following command caused the error: > make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed - > e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i > in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" > "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed - > e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i > in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" > "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v > "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf > | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$ > (for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" > "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v > "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf > | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$ > (for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" > "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/ > packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed - > e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo > $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$ > (tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/ > freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | > sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; > done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$ > (grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/ > modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; > mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo > $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep > -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in > $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" > || echo -s` install-exec-am install-data-am > make: Fatal error: Command failed for target `install-am' > Current working directory /export/home/user/packages/BUILD/ > freeswitch-1.0.4 > *** Error code 1 > The following command caused the error: > failcom='exit 1'; \ > for f in x $MAKEFLAGS; do \ > case $f in \ > *=* | --[!k]*);; \ > *k*) failcom='fail=yes';; \ > esac; \ > done; \ > dot_seen=no; \ > target=`echo install-recursive | sed s/-recursive//`; \ > list='. src build'; for subdir in $list; do \ > echo "Making $target in $subdir"; \ > if test "$subdir" = "."; then \ > dot_seen=yes; \ > local_target="$target-am"; \ > else \ > local_target="$target"; \ > fi; \ > (cd $subdir && make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$ > (grep -v "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/ > modules.conf | sed -e "s|^.*/||" | sort | uniq )"; else > tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i-all ; > done )"; echo $mods )" "OUR_CLEAN_MODULES=$(if test -z "" ; then > tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/ > freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; > else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i- > clean ; done )"; echo $mods )" "OUR_INSTALL_MODULES=$(if test -z > "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/ > freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; > else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i- > install ; done)"; echo $mods )" "OUR_UNINSTALL_MODULES=$(if test -z > "" ; then tmp_mods="$(grep -v "\#" /export/home/user/packages/BUILD/ > freeswitch-1.0.4/modules.conf | sed -e "s|^.*/||" | sort | uniq )"; > else tmp_mods="" ; fi ; mods="$(for i in $tmp_mods ; do echo $i- > uninstall ; done)"; echo $mods )" "OUR_DISABLED_MODULES=$(tmp_mods="$ > (grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/ > modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; > mods="$(for i in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" > "OUR_DISABLED_CLEAN_MODULES=$(tmp_mods="$(grep "\#" /export/home/ > user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | > sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do > echo $i-clean ; done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES= > $(tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/ > freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | > sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-install ; > done)"; echo $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$ > (grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/ > modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; > mods="$(for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo > $mods )" `test -n "" || echo -s` $local_target) \ > || eval $failcom; \ > done; \ > if test "$dot_seen" = "no"; then \ > make "OUR_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed - > e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i > in $tmp_mods ; do echo $i-all ; done )"; echo $mods )" > "OUR_CLEAN_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | sed - > e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$(for i > in $tmp_mods ; do echo $i-clean ; done )"; echo $mods )" > "OUR_INSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v > "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf > | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$ > (for i in $tmp_mods ; do echo $i-install ; done)"; echo $mods )" > "OUR_UNINSTALL_MODULES=$(if test -z "" ; then tmp_mods="$(grep -v > "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf > | sed -e "s|^.*/||" | sort | uniq )"; else tmp_mods="" ; fi ; mods="$ > (for i in $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" > "OUR_DISABLED_MODULES=$(tmp_mods="$(grep "\#" /export/home/user/ > packages/BUILD/freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed - > e "s|^.*/||" | sort | uniq )"; mods="$(for i in $tmp_mods ; do echo > $i-all ; done )"; echo $mods )" "OUR_DISABLED_CLEAN_MODULES=$ > (tmp_mods="$(grep "\#" /export/home/user/packages/BUILD/ > freeswitch-1.0.4/modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | > sort | uniq )"; mods="$(for i in $tmp_mods ; do echo $i-clean ; > done )"; echo $mods )" "OUR_DISABLED_INSTALL_MODULES=$(tmp_mods="$ > (grep "\#" /export/home/user/packages/BUILD/freeswitch-1.0.4/ > modules.conf | grep -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; > mods="$(for i in $tmp_mods ; do echo $i-install ; done)"; echo > $mods )" "OUR_DISABLED_UNINSTALL_MODULES=$(tmp_mods="$(grep "\#" / > export/home/user/packages/BUILD/freeswitch-1.0.4/modules.conf | grep > -v "\#\#" | sed -e "s|^.*/||" | sort | uniq )"; mods="$(for i in > $tmp_mods ; do echo $i-uninstall ; done)"; echo $mods )" `test -n "" > || echo -s` "$target-am" || exit 1; \ > fi; test -z "$fail" > make: Fatal error: Command failed for target `install-recursive' > Current working directory /export/home/user/packages/BUILD/ > freeswitch-1.0.4 > *** Error code 1 > make: Fatal error: Command failed for target `install' > _______________________________________________ From josh at radianttiger.com Wed Sep 9 22:44:33 2009 From: josh at radianttiger.com (Josh Rivers) Date: Wed, 9 Sep 2009 22:44:33 -0700 Subject: [Freeswitch-users] Implementing h extension in FS In-Reply-To: References: Message-ID: You should be able to handle hangups in one of two ways:1) Register a hangup handler in your script or dialplan. This will execute a script on the hangup of the call. 2) Use the Event Socket Layer(ESL) to listen to hangup events and then perform your actions there. You can find more about these options in the wiki. On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir wrote: > HI, > > I'm newbie in FS, I want to know how to implement h extension of asterisk > to FS. As I listed down below; > > h => > { > NOOP("Call Completed with Carrier ${CARRIER}"); > goto add_cdr|h|1; > }; > > My other question is, which application/function/class is use in mod_perl > to check the channel status? i.e. busy, answer,hangup,ringing,etc. > > > Kindly advice me soon. > > -- > Regards, > > Ahmed Munir > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090909/151479f9/attachment.html From mgg at giagnocavo.net Wed Sep 9 23:09:28 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 10 Sep 2009 02:09:28 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5982@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B598E@mse17be1.mse17.exchange.ms> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DB2B59B3@mse17be1.mse17.exchange.ms> Well, a segfault in voicemail would do the same thing. Suppose your plugin runs a thread that does something important, like billing or so on. That thread fails - do you really want it to go on? Anyways, the solution is simple enough, handle your exceptions :). Every plugin can decide what it wants to do here. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Wednesday, September 09, 2009 10:41 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET The question is whether the CLR should take down the whole phone server due to an unhandled exception...definitely the CLR should terminate...but shouldn't it just log the exception to the console, not crash the core? On Wed, Sep 9, 2009 at 6:30 PM, Michael Giagnocavo > wrote: That's by design. If a thread fails, and there's no handler, then the application could be in a corrupted state, so the CLR takes down the process. I think there is a .NET 1.0 compat switch you can enable in the config if you like exceptions to be silently ignored :). -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Wednesday, September 09, 2009 6:39 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET I have a new thought on the crashes...I'm able to crash FreeSWITCH any time I like, just by having an exception in a thread. public class CrashFreeSWITCH : ILoadNotificationPlugin { public bool Load() { ThreadPool.QueueUserWorkItem((o) => { throw new NotImplementedException(); }); return true; } } Perhaps Application.ThreadException or AppDomain.UnhandledException need to be trapped? On Wed, Sep 9, 2009 at 4:51 PM, Michael Giagnocavo > wrote: >Looks like the event object goes straight to pinvokes, so a null result just crashes? If it's null, you should get a NullReferenceException. The C# compiler should callvirt the property getter and that'll do a null check. If that isn't happening, that'd be an interesting optimization somewhere along the line. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Wednesday, September 09, 2009 3:01 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET A new discovery: public bool Load() { ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); if (ev == null) continue; Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); } }); return true; } Does not crash. (Adding the null check prevents crash.) The backgrounded loop runs fine. Looks like the event object goes straight to pinvokes, so a null result just crashes? I like the idea of a 'startup-script' for mod_managed. It would also be excellent if there was an event or message informing the background code to terminate nicely when the module reloads. --Josh On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk > wrote: I think the problem here is that the loader only keeps this method in scope until completion then it drops the remoted connection. Therefore you should not use threads in this method. Michael please correct me if I am wrong here. As an example of the failure simply just put a Sleep(10000) call in the thread and you will see the failure. As Michael said this method was only designed to allow the option to opt out of being loaded. In order to support this perhaps a configuration flag simular to the lua "startup-script" should be added. Here is the error I get with the loop I mentioned. -Josh [image: Capture.PNG] On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo >wrote: > Hi, > > > > Can you please elaborate on the crash you receive when you > queue a thread during load? > > > > Thanks, > > Michael > > -- View this message in context: http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/9c0f979d/attachment.html From dome at tel.co.th Wed Sep 9 23:29:45 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Thu, 10 Sep 2009 13:29:45 +0700 Subject: [Freeswitch-users] filter in fs_cli In-Reply-To: <87f2f3b90909091340p218fe5f7h29c27cdd3d1c30d4@mail.gmail.com> References: <8ccbff060909091017u5c9ec05t8fcf01532618ef2@mail.gmail.com> <87f2f3b90909091336u19fac5fei71ef59a70bd6e990@mail.gmail.com> <87f2f3b90909091340p218fe5f7h29c27cdd3d1c30d4@mail.gmail.com> Message-ID: <8ccbff060909092329r7cc20db2ldbb4b22efe98dd51@mail.gmail.com> How to use filter with sofia trace on ? Like Asterisk we can debug sip by sip set debug ip xx.xx.xx.xx. BG Dome C. 2009/9/10 Michael Collins : > > > On Wed, Sep 9, 2009 at 1:36 PM, Michael Collins wrote: >> >> >> On Wed, Sep 9, 2009 at 10:17 AM, Dome Charoenyost wrote: >>> >>> Dear All, >>> ? ? ? ? ? ?I'm looking for document,example for /filter command. >>> where to get it ? >> >> This is a handy way to add filters to what you see on the fs_cli. Event >> sockets allow for filters and the "/filter" command lets you add them to >> your fs_cli session. >> >> Check this page for specifics: >> http://wiki.freeswitch.org/wiki/Mod_event_socket#filter >> >> -MC >> > > Also, I forgot to mention that this is used in conjunction with the "/event' > command. Open fs_cli and execute these commands: > > /log 0 > /event plain all > > At this point you will get no log messages and just events. Now you can > filter them as needed. Example: > > /filter Event-Name CHANNEL_EXECUTE > > Have fun! > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dome at tel.co.th Wed Sep 9 23:32:42 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Thu, 10 Sep 2009 13:32:42 +0700 Subject: [Freeswitch-users] Mod_fifo posision in queue In-Reply-To: References: <8ccbff060909010737v4a8f17ep4d6175c2a2a2f866@mail.gmail.com> Message-ID: <8ccbff060909092332q1d06e74m8d5ca3a39ca4f572@mail.gmail.com> 2009/9/10 Michael Jerris : > You can use a phrase macro but I am not sure that we set the position > in a way that you can expand it for the macro. fifo don't have queue position variable ? Dome C. > > Mike > > On Sep 1, 2009, at 10:37 AM, Dome Charoenyost wrote: > >> 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From panayotov.vd at gmail.com Wed Sep 9 23:54:18 2009 From: panayotov.vd at gmail.com (Vassil Panayotov) Date: Thu, 10 Sep 2009 09:54:18 +0300 Subject: [Freeswitch-users] Sangoma A500 - dial out from specific port group? In-Reply-To: References: <8a9b664c0909082320l3be47aedk8d67c2c83d5b411d@mail.gmail.com> Message-ID: <8a9b664c0909092354v62cc84b8mcf525d46e91c3e72@mail.gmail.com> Michael, Moises and Octavio thank you for your replies! The server will be shipped to another site today and I can't test thoroughly now. When it is installed I will update this thread. Best regards, Vassil On Thu, Sep 10, 2009 at 1:58 AM, Octavio Ruiz wrote: > On Wed, Sep 9, 2009 at 01:20, Vassil Panayotov wrote: >> Hi, >> >> Is it possible to originate calls from specific A500 ports with FreeSWITCH? >> I am using a A504 (8 BRI interfaces), and I want some outbound calls to be >> made from specific BRI interfaces. > > You can't define several spans in openzap.conf for boost, the > sangoma_brid config file is where you define groups, so your config > should look like this: > > /// smg_bri.conf > ...... > > group=1 > spans=1 > > group=2 > spans=2 > > group=3 > spans=3 > > ...... > > /// openzap.conf > > ?[span wanpipe BoostBRI] > ?trunk_type => bri > ?b-channel => 1:1-2 > ?b-channel => 2:1-2 > ?b-channel => 3:1-2 > ?b-channel => 4:1-2 > ?b-channel => 5:1-2 > ?b-channel => 6:1-2 > ?b-channel => 7:1-2 > ?b-channel => 8:1-2 > > /// openzap.conf.xml > > ? > ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? ? > ? ? > ? > > > Then, you can Dial to your span/group number 3 with: > > freeswitch> ? ?originate openzap/1/a/12345 at g3 > |&() > freeswitch> ? ?originate openzap/1/a/12345 at G3 > |&() > freeswitch> ? ?originate openzap/1/a/12345 at r3 > |&() > freeswitch> ? ?originate openzap/1/a/12345 at R3 > |&() > > > If you are using FS 1.0.4, there is a bug, you can fix it with this > -already in trunk- patch. > > Index: src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c > =================================================================== > --- libs/openzap/src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c.orig > +++ libs/openzap/src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c > @@ -282,6 +282,8 @@ > ? ? ? ?} > > ? ? ? ?ss7bc_call_init(&event, caller_data->cid_num.digits, ani, r); > + ? ? ? //ss7_bc_call_init will clear the trunk_group val so we need to set it again > + ? ? ? event.trunk_group=tg; > > ? ? ? ?if (gr && *(gr+1)) { > > Best regards, > > -- > Octavio H. Ruiz Cervera > Tel.: (+52 55) 8590-9000 Ext. 7016 > Mobile: (+52 1 55) 4358-4565 > Sent from Mexico City, DF, Mexico > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jmesquita at freeswitch.org Wed Sep 9 23:57:36 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 10 Sep 2009 03:57:36 -0300 Subject: [Freeswitch-users] Implementing h extension in FS In-Reply-To: References: Message-ID: Try to explain a little bit better what add_cdr does right here. Unlike Asterisk, FreeSWITCH do have lots on information on CDR and it feels like you are trying to do things on the wrong place. If you want to understand where I am going with this, take a look at this example XML CDR that can be posted by FreeSWITCH to a webserver at the end of a call: http://wiki.freeswitch.org/wiki/Example_XML_cdr Also you might want to check this referece here: http://wiki.freeswitch.org/wiki/Mod_xml_cdr jmesquita On Thu, Sep 10, 2009 at 2:44 AM, Josh Rivers wrote: > You should be able to handle hangups in one of two ways:1) Register a > hangup handler in your script or dialplan. This will execute a script on the > hangup of the call. > 2) Use the Event Socket Layer(ESL) to listen to hangup events and then > perform your actions there. > > You can find more about these options in the wiki. > > On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir wrote: > >> HI, >> >> I'm newbie in FS, I want to know how to implement h extension of asterisk >> to FS. As I listed down below; >> >> h => >> { >> NOOP("Call Completed with Carrier ${CARRIER}"); >> goto add_cdr|h|1; >> }; >> >> My other question is, which application/function/class is use in mod_perl >> to check the channel status? i.e. busy, answer,hangup,ringing,etc. >> >> >> Kindly advice me soon. >> >> -- >> Regards, >> >> Ahmed Munir >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/0080665e/attachment.html From jmesquita at freeswitch.org Wed Sep 9 23:59:44 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 10 Sep 2009 03:59:44 -0300 Subject: [Freeswitch-users] filter in fs_cli In-Reply-To: <8ccbff060909092329r7cc20db2ldbb4b22efe98dd51@mail.gmail.com> References: <8ccbff060909091017u5c9ec05t8fcf01532618ef2@mail.gmail.com> <87f2f3b90909091336u19fac5fei71ef59a70bd6e990@mail.gmail.com> <87f2f3b90909091340p218fe5f7h29c27cdd3d1c30d4@mail.gmail.com> <8ccbff060909092329r7cc20db2ldbb4b22efe98dd51@mail.gmail.com> Message-ID: No can do. There are better tools to do that. tshark, wireshark and all other variants can do that for you. jmesquita On Thu, Sep 10, 2009 at 3:29 AM, Dome Charoenyost wrote: > How to use filter with sofia trace on ? > Like Asterisk we can debug sip by > > sip set debug ip xx.xx.xx.xx. > > BG > > Dome C. > > 2009/9/10 Michael Collins : > > > > > > On Wed, Sep 9, 2009 at 1:36 PM, Michael Collins > wrote: > >> > >> > >> On Wed, Sep 9, 2009 at 10:17 AM, Dome Charoenyost > wrote: > >>> > >>> Dear All, > >>> I'm looking for document,example for /filter command. > >>> where to get it ? > >> > >> This is a handy way to add filters to what you see on the fs_cli. Event > >> sockets allow for filters and the "/filter" command lets you add them to > >> your fs_cli session. > >> > >> Check this page for specifics: > >> http://wiki.freeswitch.org/wiki/Mod_event_socket#filter > >> > >> -MC > >> > > > > Also, I forgot to mention that this is used in conjunction with the > "/event' > > command. Open fs_cli and execute these commands: > > > > /log 0 > > /event plain all > > > > At this point you will get no log messages and just events. Now you can > > filter them as needed. Example: > > > > /filter Event-Name CHANNEL_EXECUTE > > > > Have fun! > > -MC > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/b2c47e29/attachment.html From msc at freeswitch.org Thu Sep 10 00:21:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Sep 2009 00:21:05 -0700 Subject: [Freeswitch-users] filter in fs_cli In-Reply-To: References: <8ccbff060909091017u5c9ec05t8fcf01532618ef2@mail.gmail.com> <87f2f3b90909091336u19fac5fei71ef59a70bd6e990@mail.gmail.com> <87f2f3b90909091340p218fe5f7h29c27cdd3d1c30d4@mail.gmail.com> <8ccbff060909092329r7cc20db2ldbb4b22efe98dd51@mail.gmail.com> Message-ID: <87f2f3b90909100021x6bf0a362t1fd5ffb929171a3c@mail.gmail.com> 2009/9/9 Jo?o Mesquita > No can do. There are better tools to do that. tshark, wireshark and all > other variants can do that for you. > > JM is correct here. The filter is to filter FreeSWITCH events, not log information hitting the console. I highly recommend this page for SIP traces: http://wiki.freeswitch.org/wiki/Packet_Capture -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/c69810ae/attachment.html From msc at freeswitch.org Thu Sep 10 00:37:39 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Sep 2009 00:37:39 -0700 Subject: [Freeswitch-users] Mod_fifo posision in queue In-Reply-To: <8ccbff060909092332q1d06e74m8d5ca3a39ca4f572@mail.gmail.com> References: <8ccbff060909010737v4a8f17ep4d6175c2a2a2f866@mail.gmail.com> <8ccbff060909092332q1d06e74m8d5ca3a39ca4f572@mail.gmail.com> Message-ID: <87f2f3b90909100037m45b93a9ei7ec62b4d1f84e986@mail.gmail.com> On Wed, Sep 9, 2009 at 11:32 PM, Dome Charoenyost wrote: > 2009/9/10 Michael Jerris : > > You can use a phrase macro but I am not sure that we set the position > > in a way that you can expand it for the macro. > > fifo don't have queue position variable ? > > I have a feeling that this functionality is possible but it will probably need to be added to mod_fifo.c as a feature request. Announcing the caller's position in queue is really a function of the queue system itself and not some third-party script. That being said, a question for the devs is this: if I know the uuid of a call that's in a fifo queue can I displace the audio (be it MOH, fifo_chime, etc.) and playback the caller's position (or any other audio for that matter) without disrupting the queue system? Just curious. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/92d00719/attachment-0001.html From jason at jasonjgw.net Thu Sep 10 00:48:23 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 10 Sep 2009 17:48:23 +1000 Subject: [Freeswitch-users] filter in fs_cli In-Reply-To: References: <8ccbff060909091017u5c9ec05t8fcf01532618ef2@mail.gmail.com> <87f2f3b90909091336u19fac5fei71ef59a70bd6e990@mail.gmail.com> <87f2f3b90909091340p218fe5f7h29c27cdd3d1c30d4@mail.gmail.com> <8ccbff060909092329r7cc20db2ldbb4b22efe98dd51@mail.gmail.com> Message-ID: <20090910074823.GA15797@jdc.jasonjgw.net> Jo?o Mesquita wrote: > No can do. There are better tools to do that. tshark, wireshark and all > other variants can do that for you. I would recommend learning about read filters in tshark/wireshark, which support a very flexible filtering language that is suitable for capturing SIP traffic. From msc at freeswitch.org Thu Sep 10 00:55:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Sep 2009 00:55:02 -0700 Subject: [Freeswitch-users] Implementing h extension in FS In-Reply-To: References: Message-ID: <87f2f3b90909100055v7e644bcbk62deb0d8e2201b11@mail.gmail.com> On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir wrote: > HI, > > I'm newbie in FS, I want to know how to implement h extension of asterisk > to FS. As I listed down below; > > h => > { > NOOP("Call Completed with Carrier ${CARRIER}"); > goto add_cdr|h|1; > }; > > My other question is, which application/function/class is use in mod_perl > to check the channel status? i.e. busy, answer,hangup,ringing,etc. > > > Kindly advice me soon. > > -- > Regards, > > Ahmed Munir > > It depends on what you are trying to accomplish, but the closest thing you'll find in FS to the 'h' extension is the channel variable api_hangup_hook which lets you launch an API at the end of the call. It sounds like you are working on CDR processing. Please tell us more about your application and we'll do our best to offer advice. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/cfaf65ce/attachment.html From msc at freeswitch.org Thu Sep 10 01:02:02 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Sep 2009 01:02:02 -0700 Subject: [Freeswitch-users] ANNOUNCEMENT: Friday Meeting Agenda - 2009-09-11 Message-ID: <87f2f3b90909100102rdb6c67do6193af9db718eac3@mail.gmail.com> FYI, The Friday meeting agenda is posted here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11 Please add agenda topics as needed. We invite all to join us this coming Friday at 11AM CST. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/36cbd032/attachment.html From krice at freeswitch.org Thu Sep 10 01:06:57 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 10 Sep 2009 03:06:57 -0500 Subject: [Freeswitch-users] Implementing h extension in FS In-Reply-To: <87f2f3b90909100055v7e644bcbk62deb0d8e2201b11@mail.gmail.com> Message-ID: If you are just doing CDR processing the easiest ways are using the event socket to trigger this on the hangup event and you will get all the data you want or use mod_xml_cdr which will either drop a file or fire a web request. You can use either of these methods to trigger a billing update From: Michael Collins Reply-To: Date: Thu, 10 Sep 2009 00:55:02 -0700 To: Subject: Re: [Freeswitch-users] Implementing h extension in FS On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir wrote: > HI, > > I'm newbie in FS, I want to know how to implement h extension of asterisk to > FS. As I listed down below; > > h => > ??? { > ??? ??? NOOP("Call Completed with Carrier ${CARRIER}"); > ??? ??? goto add_cdr|h|1; > ??? }; > > My other question is, which application/function/class is use in mod_perl to > check the channel status? i.e. busy, answer,hangup,ringing,etc. > > > Kindly advice me soon. > > -- > Regards, > > Ahmed Munir > It depends on what you are trying to accomplish, but the closest thing you'll find in FS to the 'h' extension is the channel variable api_hangup_hook which lets you launch an API at the end of the call. It sounds like you are working on CDR processing. Please tell us more about your application and we'll do our best to offer advice. -MC _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/1237959e/attachment.html From gmaruzz at celliax.org Thu Sep 10 01:32:38 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 10 Sep 2009 10:32:38 +0200 Subject: [Freeswitch-users] Skypiax false DTMF event In-Reply-To: <90823c940909091304j2e2c7ad2m50f80e1e605fb9f9@mail.gmail.com> References: <90823c940909091304j2e2c7ad2m50f80e1e605fb9f9@mail.gmail.com> Message-ID: <7b197bef0909100132x7605c21fm19374333be6ca009@mail.gmail.com> On Wed, Sep 9, 2009 at 10:04 PM, Dmitry Bely wrote: > I have a problem. After 10-20 minutes of Skype talk via cordless phone > connected to ATA the latter erroneously generated DTMF 'D' ?event. > Then skypiax looses connection while the call remain active in Skype > client. The only way to terminate it is to ask another party to hang > up: Ciao Dmitry, could you please fill a Jira with the same infos? http://wiki.freeswitch.org/wiki/Skypiax#How_To_Report_BUGS_and_Feature_Requests That is the standard and correct procedure for bugs, so the devs can follow up on it. -giovanni > > (...) > > 2009-09-09 22:20:07.474051 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ][interface1][-1, 5,21] READING: > |||CALL 307 DURATION 500||| > 2009-09-09 22:20:08.473755 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ][interface1][-1, 5,21] READING: > |||CALL 307 DURATION 501||| > 2009-09-09 22:20:09.474247 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ][interface1][-1, 5,21] READING: > |||CALL 307 DURATION 502||| > 2009-09-09 22:20:10.474611 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ][interface1][-1, 5,21] READING: > |||CALL 307 DURATION 503||| > 2009-09-09 22:20:11.474456 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ][interface1][-1, 5,21] READING: > |||CALL 307 DURATION 504||| > 2009-09-09 22:20:12.411664 [DEBUG] switch_rtp.c:2239 RTP RECV DTMF D:2000 > 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:633 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?633 ?][interface1][-1, 5,21] > interface1 CHANNEL SEND_DTMF > 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:634 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?634 ?][interface1][-1, 5,21] DTMF: D > 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:882 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?882 ?][interface1][-1, 5,21] DIGIT > received: D > 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:1352 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?1352 ][interface1][-1, 5,21] > SENDING: |||SET CALL 307 DTMF D|||| > 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:1530 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?1530 ][interface1][-1, 5,21] Got a > 'continue' XAtom without a previous 'begin'. It's value (between > vertical bars) is=|||allowed call prop||| > 2009-09-09 22:20:12.411664 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ?][interface1][-1, 5,21] > READING: |||ERROR 21 Unknown/dis||| > 2009-09-09 22:20:12.411664 [ERR] skypiax_protocol.c:144 rev > 14707[(nil)|37 ? ? ][ERRORA ?144 ?][interface1][-1, 5,21] Skype got > ERROR: |||ERROR 21 Unknown/dis||| > 2009-09-09 22:20:12.411664 [ERR] skypiax_protocol.c:146 rev > 14707[(nil)|37 ? ? ][ERRORA ?146 ?][interface1][-1, 5,16] skype_call > now is DOWN > 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:1011 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?1011 ][interface1][-1, 1,16] skype > call ended > 2009-09-09 22:20:12.411664 [NOTICE] mod_skypiax.c:1022 Hangup > skypiax/interface1/user2 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2009-09-09 22:20:12.411664 [DEBUG] switch_channel.c:1715 Send signal > skypiax/interface1/user2 [KILL] > 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:566 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?566 ?][interface1][-1, 1,16] > interface1 CHANNEL KILL_CHANNEL > 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:569 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?569 ?][interface1][-1, 1,16] > skypiax/interface1/user2 CHANNEL got SWITCH_SIG_KILL > 2009-09-09 22:20:12.411664 [DEBUG] switch_core_session.c:932 Send > signal skypiax/interface1/user2 [BREAK] > 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:566 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?566 ?][interface1][-1, 1,16] > interface1 CHANNEL KILL_CHANNEL > 2009-09-09 22:20:12.411664 [DEBUG] mod_skypiax.c:589 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?589 ?][interface1][-1, 1,16] > skypiax/interface1/user2 CHANNEL got SWITCH_SIG_BREAK > 2009-09-09 22:20:12.428590 [DEBUG] skypiax_protocol.c:670 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?670 ?][interface1][-1, 1,16] Skype > incoming audio GONE > 2009-09-09 22:20:12.428590 [DEBUG] mod_skypiax.c:702 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?702 ?][interface1][-1, 1,16] CHANNEL > READ FALSE > 2009-09-09 22:20:12.428590 [DEBUG] switch_ivr_bridge.c:377 > skypiax/interface1/user2 ending bridge by request from read function > 2009-09-09 22:20:12.428590 [DEBUG] switch_ivr_bridge.c:452 BRIDGE > THREAD DONE [skypiax/interface1/user2] > 2009-09-09 22:20:12.428590 [DEBUG] switch_ivr_bridge.c:454 Send signal > sofia/internal/1002 at 192.168.121.66 ? ? ? ?[BREAK] > 2009-09-09 22:20:12.428590 [DEBUG] switch_core_state_machine.c:497 > (skypiax/interface1/user2) State EXCHANGE_MEDIA going to sleep > 2009-09-09 22:20:12.428590 [DEBUG] switch_core_state_machine.c:398 > (skypiax/interface1/user2) Running State Change CS_HANGUP > 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:434 > (skypiax/interface1/user2) State HANGUP > 2009-09-09 22:20:12.429654 [DEBUG] mod_skypiax.c:506 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?506 ?][interface1][-1, 1,16] > interface1 CHANNEL HANGUP > 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:46 > skypiax/interface1/user2 Standard HANGUP, cause: NORMAL_CLEARING > 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:434 > (skypiax/interface1/user2) State HANGUP going to sleep > 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:479 > (skypiax/interface1/user2) State Change CS_HANGUP -> CS_REPORTING > 2009-09-09 22:20:12.429654 [DEBUG] switch_core_session.c:932 Send > signal skypiax/interface1/user2 [BREAK] > 2009-09-09 22:20:12.429654 [DEBUG] mod_skypiax.c:566 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?566 ?][interface1][-1, 0, 0] > interface1 CHANNEL KILL_CHANNEL > 2009-09-09 22:20:12.429654 [DEBUG] mod_skypiax.c:589 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?589 ?][interface1][-1, 0, 0] > skypiax/interface1/user2 CHANNEL got SWITCH_SIG_BREAK > 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:398 > (skypiax/interface1/user2) Running State Change CS_REPORTING > 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:616 > (skypiax/interface1/user2) State REPORTING > 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:53 > skypiax/interface1/user2 Standard REPORTING, cause: NORMAL_CLEARING > 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:616 > (skypiax/interface1/user2) State REPORTING going to sleep > 2009-09-09 22:20:12.429654 [DEBUG] switch_core_state_machine.c:411 > (skypiax/interface1/user2) State Change CS_REPORTING -> CS_DESTROY > 2009-09-09 22:20:12.429654 [DEBUG] switch_core_session.c:932 Send > signal skypiax/interface1/user2 [BREAK] > 2009-09-09 22:20:12.429654 [DEBUG] mod_skypiax.c:566 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?566 ?][interface1][-1, 0, 0] > interface1 CHANNEL KILL_CHANNEL > 2009-09-09 22:20:12.429654 [DEBUG] mod_skypiax.c:589 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?589 ?][interface1][-1, 0, 0] > skypiax/interface1/user2 CHANNEL got SWITCH_SIG_BREAK > 2009-09-09 22:20:12.429654 [DEBUG] switch_core_session.c:1068 Session > 8 (skypiax/interface1/user2) Locked, Waiting on external entities > 2009-09-09 22:20:12.439620 [DEBUG] skypiax_protocol.c:849 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?849 ?][interface1][-1, 0, 0] Skype > outbound audio GONE > 2009-09-09 22:20:12.458666 [DEBUG] switch_ivr_bridge.c:426 > sofia/internal/1002 at 192.168.121.66 ? ?receive message [UNBRIDGE] > 2009-09-09 22:20:12.458666 [DEBUG] switch_core_session.c:630 Send > signal sofia/internal/1002 at 192.168.121.66 ? ? ?[BREAK] > 2009-09-09 22:20:12.458666 [DEBUG] switch_ivr_bridge.c:452 BRIDGE > THREAD DONE [sofia/internal/1002 at 192.168.121.66 ? ? ? ] > 2009-09-09 22:20:12.458666 [DEBUG] switch_ivr_bridge.c:454 Send signal > skypiax/interface1/user2 [BREAK] > 2009-09-09 22:20:12.458666 [DEBUG] mod_skypiax.c:566 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?566 ?][interface1][-1, 0, 0] > interface1 CHANNEL KILL_CHANNEL > 2009-09-09 22:20:12.458666 [DEBUG] mod_skypiax.c:589 rev > 14771[(nil)|37 ? ? ][DEBUG_SKYPE ?589 ?][interface1][-1, 0, 0] > skypiax/interface1/user2 CHANNEL got SWITCH_SIG_BREAK > 2009-09-09 22:20:12.458666 [NOTICE] switch_core_state_machine.c:179 > Hangup sofia/internal/1002 at 192.168.121.66 ? ?[CS_EXECUTE] > [NORMAL_CLEARING] > 2009-09-09 22:20:12.458666 [DEBUG] switch_channel.c:1715 Send signal > sofia/internal/1002 at 192.168.121.66 ?[KILL] > 2009-09-09 22:20:12.458666 [DEBUG] switch_core_session.c:932 Send > signal sofia/internal/1002 at 192.168.121.66 ? ? ?[BREAK] > 2009-09-09 22:20:12.458666 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/1002 at 192.168.121.66) State EXECUTE going to sleep > 2009-09-09 22:20:12.458666 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/1002 at 192.168.121.66) Running State Change CS_HANGUP > 2009-09-09 22:20:12.459723 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/1002 at 192.168.121.66) State HANGUP > 2009-09-09 22:20:12.459723 [DEBUG] mod_sofia.c:338 Channel > sofia/internal/1002 at 192.168.121.66 ? ?hanging up, cause: > NORMAL_CLEARING > 2009-09-09 22:20:12.459723 [DEBUG] mod_sofia.c:376 Sending BYE to > sofia/internal/1002 at 192.168.121.66 > > (...) > > 2009-09-09 22:20:12.712461 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ?][interface1][-1, 0, 0] > READING: |||CALL 307 DURATION 505||| > 2009-09-09 22:20:12.712461 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ?][interface1][-1, 0, 0] > READING: |||CALL 307 VAA_INPUT_STATUS FALSE||| > 2009-09-09 22:20:13.473652 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ?][interface1][-1, 0, 0] > READING: |||CALL 307 DURATION 506||| > 2009-09-09 22:20:14.473723 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ?][interface1][-1, 0, 0] > READING: |||CALL 307 DURATION 507||| > > (...) > > 2009-09-09 22:26:47.775793 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ?][interface1][-1, 0, 0] > READING: |||CALL 307 STATUS FINISHED||| > 2009-09-09 22:26:47.775793 [DEBUG] skypiax_protocol.c:361 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?361 ?][interface1][-1, 0, 0] > skype_call 307 is NOT MY call, ignoring > 2009-09-09 22:48:08.274202 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ?][interface1][-1, 0, 0] > READING: |||USER user1 ONLINESTATUS OFFLINE||| > 2009-09-09 22:48:08.274202 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ?][interface1][-1, 0, 0] > READING: |||USERSTATUS OFFLINE||| > 2009-09-09 22:48:08.274202 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ?][interface1][-1, 0, 0] > READING: |||CONNSTATUS CONNECTING||| > 2009-09-09 22:48:31.374348 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ?][interface1][-1, 0, 0] > READING: |||USER user1 ONLINESTATUS ONLINE||| > 2009-09-09 22:48:31.374348 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ?][interface1][-1, 0, 0] > READING: |||USERSTATUS ONLINE||| > 2009-09-09 22:48:31.375500 [DEBUG] skypiax_protocol.c:104 rev > 14707[(nil)|37 ? ? ][DEBUG_SKYPE ?104 ?][interface1][-1, 0, 0] > READING: |||CONNSTATUS ONLINE||| > > Is this expected behavior? Of course, the main problem is probably in > hardware but does skypiax do its job right? > > - Dmitry Bely > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nandy1925 at gmail.com Thu Sep 10 02:30:37 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Thu, 10 Sep 2009 17:30:37 +0800 Subject: [Freeswitch-users] example configs for FS outside of NAT? In-Reply-To: References: <938ad7be0909090316p2f85f104w8aa3c95b3c8a6ae8@mail.gmail.com> <938ad7be0909090343x29263d6cq3eeee6bbafbb799c@mail.gmail.com> <938ad7be0909090416y1d2986ddyceb017c188235d05@mail.gmail.com> Message-ID: <7d0bfd8c0909100230s4f913f32o1f4958db4a75672f@mail.gmail.com> hi brian, for outside clients to register w/ the internal profile, the router has to forward port 5060 to FS. am i correct? /nandy On Wed, Sep 9, 2009 at 10:28 PM, Brian West wrote: > Those configs will still work. > /b > > On Sep 9, 2009, at 6:16 AM, J?rg Hartmann wrote: > > Hi there, > > the internal.xml and external.xml examples are for situations where FS is > running inside a company's private network, behind a NAT router. So > internal.xml connects the clients to FS without crossing a NAT, within the > same private network, while external.xml connects SIP providers through the > NAT router. > > But what if FS is running with a public IP (and DNS entry) outside the > private network, so that the clients have to pass the NAT router to connect > with FS, while FS can connect to SIP providers directly? Are there any > example configs for such a configuration? > > Thanks in advance, > Cheers, > JH > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/7a6236dc/attachment-0001.html From jason at jasonjgw.net Thu Sep 10 02:52:23 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 10 Sep 2009 19:52:23 +1000 Subject: [Freeswitch-users] example configs for FS outside of NAT? In-Reply-To: <7d0bfd8c0909100230s4f913f32o1f4958db4a75672f@mail.gmail.com> References: <938ad7be0909090316p2f85f104w8aa3c95b3c8a6ae8@mail.gmail.com> <938ad7be0909090343x29263d6cq3eeee6bbafbb799c@mail.gmail.com> <938ad7be0909090416y1d2986ddyceb017c188235d05@mail.gmail.com> <7d0bfd8c0909100230s4f913f32o1f4958db4a75672f@mail.gmail.com> Message-ID: <20090910095223.GA20076@jdc.jasonjgw.net> Nandy Dagondon wrote: > for outside clients to register w/ the internal profile, the router has to > forward port 5060 to FS. am i correct? Yes, but by default the internal profile doesn't handle nat, which is why (if I recall correctly) it has been recommended that the external profile be used to register clients that are not on the local network when nat is involved. From nandy1925 at gmail.com Thu Sep 10 03:57:24 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Thu, 10 Sep 2009 18:57:24 +0800 Subject: [Freeswitch-users] example configs for FS outside of NAT? In-Reply-To: <20090910095223.GA20076@jdc.jasonjgw.net> References: <938ad7be0909090316p2f85f104w8aa3c95b3c8a6ae8@mail.gmail.com> <938ad7be0909090343x29263d6cq3eeee6bbafbb799c@mail.gmail.com> <938ad7be0909090416y1d2986ddyceb017c188235d05@mail.gmail.com> <7d0bfd8c0909100230s4f913f32o1f4958db4a75672f@mail.gmail.com> <20090910095223.GA20076@jdc.jasonjgw.net> Message-ID: <7d0bfd8c0909100357q4715f2b4l6176ed3289313d62@mail.gmail.com> hi jason, yes, we're aware of the external profile. but the sample profile shows only how to register FS to SIP gateways - not external clients registering to FS. the directory/default/*xml belongs to the internal profile. how can we create another directory for external clients? we like to see sample configs in the distribution. tks for the clarification, /nandy On Thu, Sep 10, 2009 at 5:52 PM, Jason White wrote: > Nandy Dagondon wrote: > > for outside clients to register w/ the internal profile, the router has > to > > forward port 5060 to FS. am i correct? > > Yes, but by default the internal profile doesn't handle nat, which is why > (if > I recall correctly) it has been recommended that the external profile be > used > to register clients that are not on the local network when nat is involved. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/7df2a0ff/attachment.html From mcampbellsmith at gmail.com Thu Sep 10 04:39:45 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 10 Sep 2009 21:39:45 +1000 Subject: [Freeswitch-users] Dialplan Context Message-ID: <33c87fa30909100439i59214eb9tf5b57c88a6614bb4@mail.gmail.com> Hi! Where in the dialplan does FS decide which context is used for processing.. I am dialing an outbound call but the call is being processed in context public and not default? mod_dialplan_xml.c:315 Processing Extension1000-> in context public Why is FS choosing the public context for an outbound call instead of the normal default context? Thanks! From tculjaga at gmail.com Thu Sep 10 05:09:47 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 10 Sep 2009 14:09:47 +0200 Subject: [Freeswitch-users] Dialplan Context In-Reply-To: <33c87fa30909100439i59214eb9tf5b57c88a6614bb4@mail.gmail.com> References: <33c87fa30909100439i59214eb9tf5b57c88a6614bb4@mail.gmail.com> Message-ID: <65d96fc80909100509y1df8da6cob5dc9252918c9368@mail.gmail.com> check your sip profiles /usr/local/freeswitch/conf/sip_profiles/external.xml /usr/local/freeswitch/conf/sip_profiles/internal.xml /usr/local/freeswitch/conf/vars.xml It is simple :P On Thu, Sep 10, 2009 at 1:39 PM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > Where in the dialplan does FS decide which context is used for > processing.. I am dialing an outbound call but the call is being > processed in context public and not default? > > mod_dialplan_xml.c:315 Processing Extension1000-> in context > public > > Why is FS choosing the public context for an outbound call instead of > the normal default context? > > Thanks! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/595637ad/attachment.html From mcampbellsmith at gmail.com Thu Sep 10 05:24:50 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 10 Sep 2009 22:24:50 +1000 Subject: [Freeswitch-users] Dialplan Context In-Reply-To: <65d96fc80909100509y1df8da6cob5dc9252918c9368@mail.gmail.com> References: <33c87fa30909100439i59214eb9tf5b57c88a6614bb4@mail.gmail.com> <65d96fc80909100509y1df8da6cob5dc9252918c9368@mail.gmail.com> Message-ID: <33c87fa30909100524q7a2b889k98c74acf4fa16e63@mail.gmail.com> Hi! Actually I think the problem was with the acl list.. I had put commented line below in. How did this cause the internal profile be executed in the public extension? <- when removed it works again Thanks On Thu, Sep 10, 2009 at 10:09 PM, Tihomir Culjaga wrote: > check your sip profiles > > /usr/local/freeswitch/conf/sip_profiles/external.xml > > > > > /usr/local/freeswitch/conf/sip_profiles/internal.xml > > > > > > /usr/local/freeswitch/conf/vars.xml > > ? > ? > ? > ? > ? > ? > > ? > ? > ? > ? > ? > ? > > > > > It is simple :P > > > > > On Thu, Sep 10, 2009 at 1:39 PM, Mark Campbell-Smith > wrote: >> >> Hi! >> >> Where in the dialplan does FS decide which context is used for >> processing.. ?I am dialing an outbound call but the call is being >> processed in context public and not default? >> >> mod_dialplan_xml.c:315 Processing Extension1000-> in context >> public >> >> Why is FS choosing the public context for an outbound call instead of >> the normal default context? >> >> Thanks! >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From tzury.by at reguluslabs.com Thu Sep 10 06:26:02 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Thu, 10 Sep 2009 16:26:02 +0300 Subject: [Freeswitch-users] No audio on caller side when both side support speex/8000 only In-Reply-To: References: <10128ef10909090419g196d2280p5d851728fb8c6309@mail.gmail.com> Message-ID: <10128ef10909100626l12ce8528u3f5b8710cecc1ae7@mail.gmail.com> > This looks and sounds like a case where pjsip isn't listening to our > SDP. ?If we 200 OK with speex on 102 and the far end starts sending it > on 98 then I suspect the client is broken if I'm not mistaken. > > /b Brian, The original "a=rtpmap:98 SPEEX/8000" is generated by FS and NOT by the client. From juanbackson at gmail.com Thu Sep 10 06:28:17 2009 From: juanbackson at gmail.com (Juan Backson) Date: Thu, 10 Sep 2009 21:28:17 +0800 Subject: [Freeswitch-users] Need help enabling some variables in b-leg Message-ID: <27c25bc40909100628p2e84a3d3h6619375932637962@mail.gmail.com> Hi, I need to have local_media_ip, local_media_port, remote_media_ip, and remote_port to be available in the b-leg. Is there anyway to do that? Since these variables are only created after the call has been answered, I can't do .... In Kamailio, I'm executing the sleep function (from cfgutils module) in the failure_route block: failure_route[2] { ...Get your new ruri only for some reply codes... avp_pushto("$ruri", "$(avp(i:1511))"); append_branch(); t_on_failure("2"); #In case of a new failure this block will be executed again sleep("3"); t_relay(); return; } May be it's not the best solution but it could be helpful for someone else. Best regards, Humberto _________________________________________________________________ Click less, chat more: Messenger on MSN.ca http://go.microsoft.com/?linkid=9677404 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/03c61309/attachment-0001.html From brian at freeswitch.org Thu Sep 10 06:37:42 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Sep 2009 08:37:42 -0500 Subject: [Freeswitch-users] example configs for FS outside of NAT? In-Reply-To: <20090910095223.GA20076@jdc.jasonjgw.net> References: <938ad7be0909090316p2f85f104w8aa3c95b3c8a6ae8@mail.gmail.com> <938ad7be0909090343x29263d6cq3eeee6bbafbb799c@mail.gmail.com> <938ad7be0909090416y1d2986ddyceb017c188235d05@mail.gmail.com> <7d0bfd8c0909100230s4f913f32o1f4958db4a75672f@mail.gmail.com> <20090910095223.GA20076@jdc.jasonjgw.net> Message-ID: <264B1A07-9B35-4B89-8E0D-3B54F666FF13@freeswitch.org> This does now... /b On Sep 10, 2009, at 4:52 AM, Jason White wrote: > > Yes, but by default the internal profile doesn't handle nat, which > is why (if > I recall correctly) it has been recommended that the external > profile be used > to register clients that are not on the local network when nat is > involved. From jlenk at frontiernet.net Thu Sep 10 07:33:44 2009 From: jlenk at frontiernet.net (Jeff Lenk) Date: Thu, 10 Sep 2009 09:33:44 -0500 (CDT) Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DB2B5981@mse17be1.mse17.exchange.ms> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5981@mse17be1.mse17.exchange.ms> Message-ID: <1252593224328-3617527.post@n2.nabble.com> Thanks Michael, I was seeing some strange behavior in a static compiled dll versus dynamically loaded and compiled scripts. This code when used in a dll was blowing up when a thread was spawned in the ILoadNotificationPlugin.Load but is not when used in a dynamically loaded script(csx). Code: ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); while (true) { Thread.Sleep(10000); Log.WriteLine(LogLevel.Notice, "Thread Running. "); } }); I will look into this more later. Jeff Michael Giagnocavo wrote: > > The ILoadNotifcationPlugin is run in the appdomain created for the plugin, > so it should only get unloaded when the plugin gets reloaded. Spawning > threads here should work, it's definitely the intention that if you need a > long-running process, you can fire it up on load and have it work. > > As to the race condition on reload, mod_managed should do this: > > - Load the new plugin into a new appdomain > - Remove the entry points to the old appdomain, add entries to the new > one > - Old appdomain now stays alive until foreground API and APP calls finish > > So, you can have many versions of the same plugin active in memory. > > I probably need to go break compatibility and make ILoadWhateverPlugin be > something like IPluginController and allow it to return loading options to > control the mod_managed behavior, as well as allow it to delay shutdown of > the appdomain. Part of the question is: how many people out there need > compatibility, or can we go breaking all of you and make you recompile? :) > > Although, IIRC, if you handle AppDomain.Unload (or whatever it is), it > will stay alive until your event handler completes. > > Hope that helps a bit. > > -Michael > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff > Lenk > Sent: Wednesday, September 09, 2009 1:57 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET > > > I think the problem here is that the loader only keeps this method in > scope > until completion then it drops the remoted connection. Therefore you > should > not use threads in this method. Michael please correct me if I am wrong > here. > > As an example of the failure simply just put a Sleep(10000) call in the > thread and you will see the failure. > > As Michael said this method was only designed to allow the option to opt > out > of being loaded. > > In order to support this perhaps a configuration flag simular to the lua > "startup-script" should be added. > > > > Here is the error I get with the loop I mentioned. -Josh > [image: Capture.PNG] > > On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo > wrote: > >> Hi, >> >> >> >> Can you please elaborate on the crash you receive when >> you >> queue a thread during load? >> >> >> >> Thanks, >> >> Michael >> >> > > -- > View this message in context: > http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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/Subscribing-to-events-in-managed-C-NET-tp3573619p3617527.html Sent from the freeswitch-users mailing list archive at Nabble.com. From carolinabc28 at gmail.com Thu Sep 10 07:51:42 2009 From: carolinabc28 at gmail.com (Carolina Benavides Cabrera) Date: Thu, 10 Sep 2009 09:51:42 -0500 Subject: [Freeswitch-users] FreeSWITCH configuration Message-ID: <9a28ddc30909100751v699e988ew96d7778ac4596542@mail.gmail.com> Hi. I'm looking for any function similar to "rewritehostport" function of OpenSIPS. I need to route the SIP signalling toward Sailfin AS, replacing the RequestURI. Thanks for your help Carolina -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/276ebda0/attachment.html From krice at freeswitch.org Thu Sep 10 08:12:07 2009 From: krice at freeswitch.org (Ken Rice) Date: Thu, 10 Sep 2009 10:12:07 -0500 Subject: [Freeswitch-users] FreeSWITCH configuration In-Reply-To: <9a28ddc30909100751v699e988ew96d7778ac4596542@mail.gmail.com> Message-ID: Ummm bridge... ie: This will rewrite the request URI on an invite... However keep in mind the FreeSWITCH is a B2BUA and not a proxy so this might not be exactly what you are looking for From: Carolina Benavides Cabrera Reply-To: Date: Thu, 10 Sep 2009 09:51:42 -0500 To: Subject: [Freeswitch-users] FreeSWITCH configuration Hi. I'm looking for any function similar to "rewritehostport" function of OpenSIPS. I need to route the SIP signalling toward Sailfin AS, replacing the RequestURI. Thanks for your help Carolina _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/5967489b/attachment.html From mgg at giagnocavo.net Thu Sep 10 08:14:45 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 10 Sep 2009 11:14:45 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <1252593224328-3617527.post@n2.nabble.com> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5981@mse17be1.mse17.exchange.ms> <1252593224328-3617527.post@n2.nabble.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DB2B5A03@mse17be1.mse17.exchange.ms> Were both of them actually loading completely? As in, they had an app or api plugin? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Thursday, September 10, 2009 8:34 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET Thanks Michael, I was seeing some strange behavior in a static compiled dll versus dynamically loaded and compiled scripts. This code when used in a dll was blowing up when a thread was spawned in the ILoadNotificationPlugin.Load but is not when used in a dynamically loaded script(csx). Code: ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); while (true) { Thread.Sleep(10000); Log.WriteLine(LogLevel.Notice, "Thread Running. "); } }); I will look into this more later. Jeff Michael Giagnocavo wrote: > > The ILoadNotifcationPlugin is run in the appdomain created for the plugin, > so it should only get unloaded when the plugin gets reloaded. Spawning > threads here should work, it's definitely the intention that if you need a > long-running process, you can fire it up on load and have it work. > > As to the race condition on reload, mod_managed should do this: > > - Load the new plugin into a new appdomain > - Remove the entry points to the old appdomain, add entries to the new > one > - Old appdomain now stays alive until foreground API and APP calls finish > > So, you can have many versions of the same plugin active in memory. > > I probably need to go break compatibility and make ILoadWhateverPlugin be > something like IPluginController and allow it to return loading options to > control the mod_managed behavior, as well as allow it to delay shutdown of > the appdomain. Part of the question is: how many people out there need > compatibility, or can we go breaking all of you and make you recompile? :) > > Although, IIRC, if you handle AppDomain.Unload (or whatever it is), it > will stay alive until your event handler completes. > > Hope that helps a bit. > > -Michael > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff > Lenk > Sent: Wednesday, September 09, 2009 1:57 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET > > > I think the problem here is that the loader only keeps this method in > scope > until completion then it drops the remoted connection. Therefore you > should > not use threads in this method. Michael please correct me if I am wrong > here. > > As an example of the failure simply just put a Sleep(10000) call in the > thread and you will see the failure. > > As Michael said this method was only designed to allow the option to opt > out > of being loaded. > > In order to support this perhaps a configuration flag simular to the lua > "startup-script" should be added. > > > > Here is the error I get with the loop I mentioned. -Josh > [image: Capture.PNG] > > On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo > wrote: > >> Hi, >> >> >> >> Can you please elaborate on the crash you receive when >> you >> queue a thread during load? >> >> >> >> Thanks, >> >> Michael >> >> > > -- > View this message in context: > http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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/Subscribing-to-events-in-managed-C-NET-tp3573619p3617527.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nik.middleton at noblesolutions.co.uk Thu Sep 10 09:11:45 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 10 Sep 2009 17:11:45 +0100 Subject: [Freeswitch-users] Getting core dump from last night's build Message-ID: Hi Guys, I'm getting a core dump when running an lua script that's been fine for months In Freeswitch_lua.cpp line 92 is being called, but it's not clear what exactly this is doing lua_State *Session::getLUA() { if (!L) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Doh!\n"); } return L; } Anyone shed a light on this? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/ad6abff3/attachment.html From rob4manhere at gmail.com Thu Sep 10 09:28:42 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 10 Sep 2009 11:28:42 -0500 Subject: [Freeswitch-users] FS Crashes After Second Call To Python Application Message-ID: <2BFD2BCE-22B1-4A77-8BC4-6A702A500D99@gmail.com> Hi all, I wrote a small 10-line python script wrapping "txfax" (http://pastebin.freeswitch.org/10274 ). Basically it originates a call with Session(), calls mod_fax txfax application, then hangs up. The weird thing is that this works fine the first I run it from fs_cli. When I run a second time however - it still works and sends the fax - but FS seg faults after the python script exits. Gdb shows something with Session destroy: #0 PYTHON::Session::destroy (this=0x83cd308) at freeswitch_python.cpp: 60 60 Py_DECREF(Self); Here is the backtrace from the core: http://pastebin.freeswitch.org/10275 Is this a bug- or am I not setting up or cleaning up the session properly? I was going to open a jira but thought I'd see if I'm missing something obvious first. Thanks! Rob From anthony.minessale at gmail.com Thu Sep 10 09:54:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Sep 2009 11:54:21 -0500 Subject: [Freeswitch-users] Getting core dump from last night's build In-Reply-To: References: Message-ID: <191c3a030909100954r31aec836o4838070d9abc89f9@mail.gmail.com> I wonder if maybe you have a build issue with an older mod_lua with a newer FreeSWITCH did you update via make current? On Thu, Sep 10, 2009 at 11:11 AM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Hi Guys, > > > > I?m getting a core dump when running an lua script that?s been fine for > months > > > > In Freeswitch_lua.cpp line 92 is being called, but it?s not clear what > exactly this is doing > > > > > > lua_State *Session::getLUA() > > { > > if (!L) { > > switch_log_printf(SWITCH_CHANNEL_LOG, > SWITCH_LOG_ERROR, "Doh!\n"); > > } > > return L; > > } > > > > Anyone shed a light on this? > > > > Regards > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/9860a04a/attachment.html From n.geordzhev at gmail.com Thu Sep 10 09:58:39 2009 From: n.geordzhev at gmail.com (Nikolai Geordzhev) Date: Thu, 10 Sep 2009 19:58:39 +0300 Subject: [Freeswitch-users] FS Crashes After Second Call To Python Application In-Reply-To: <2BFD2BCE-22B1-4A77-8BC4-6A702A500D99@gmail.com> References: <2BFD2BCE-22B1-4A77-8BC4-6A702A500D99@gmail.com> Message-ID: FYI, I got the same/similar issue with a 10-line perl script, starting FS without -hp option worked for me. On Thu, Sep 10, 2009 at 7:28 PM, Rob Forman wrote: > Hi all, > > I wrote a small 10-line python script wrapping "txfax" ( > http://pastebin.freeswitch.org/10274 > ). Basically it originates a call with Session(), calls mod_fax txfax > application, then hangs up. > > The weird thing is that this works fine the first I run it from > fs_cli. When I run a second time however - it still works and sends > the fax - but FS seg faults after the python script exits. > > Gdb shows something with Session destroy: > > #0 PYTHON::Session::destroy (this=0x83cd308) at freeswitch_python.cpp: > 60 > 60 Py_DECREF(Self); > > Here is the backtrace from the core: > http://pastebin.freeswitch.org/10275 > > > Is this a bug- or am I not setting up or cleaning up the session > properly? I was going to open a jira but thought I'd see if I'm > missing something obvious first. > > Thanks! > Rob > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/7e7f7ce9/attachment.html From rob4manhere at gmail.com Thu Sep 10 10:07:11 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Thu, 10 Sep 2009 12:07:11 -0500 Subject: [Freeswitch-users] FS Crashes After Second Call To Python Application In-Reply-To: References: <2BFD2BCE-22B1-4A77-8BC4-6A702A500D99@gmail.com> Message-ID: <3D1335FB-3287-464F-ABA0-29C117C88CDC@gmail.com> Hi Nikolai, Thanks for the response. Unfortunately I am already running without - hp. Just plain ol' ./bin/freeswitch. Also, I didn't mention it but I'm current and running off the build (14805). Rob On Sep 10, 2009, at 11:58 AM, Nikolai Geordzhev wrote: > FYI, I got the same/similar issue with a 10-line perl script, > starting FS without -hp option worked for me. > > On Thu, Sep 10, 2009 at 7:28 PM, Rob Forman > wrote: > Hi all, > > I wrote a small 10-line python script wrapping "txfax" (http://pastebin.freeswitch.org/10274 > ). Basically it originates a call with Session(), calls mod_fax txfax > application, then hangs up. > > The weird thing is that this works fine the first I run it from > fs_cli. When I run a second time however - it still works and sends > the fax - but FS seg faults after the python script exits. > > Gdb shows something with Session destroy: > > #0 PYTHON::Session::destroy (this=0x83cd308) at > freeswitch_python.cpp: > 60 > 60 Py_DECREF(Self); > > Here is the backtrace from the core: > http://pastebin.freeswitch.org/10275 > > > Is this a bug- or am I not setting up or cleaning up the session > properly? I was going to open a jira but thought I'd see if I'm > missing something obvious first. > > Thanks! > Rob > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/4ce7dbf8/attachment.html From diego.viola at gmail.com Thu Sep 10 10:08:57 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 10 Sep 2009 17:08:57 +0000 Subject: [Freeswitch-users] FS Crashes After Second Call To Python Application In-Reply-To: References: <2BFD2BCE-22B1-4A77-8BC4-6A702A500D99@gmail.com> Message-ID: <86a32abc0909101008t54d2c9a0w674f83fef90f97d4@mail.gmail.com> Well, that shouldn't happen, get a trace and open a jira or contact the developers to resolve the issue. Best regards, Diego On Thu, Sep 10, 2009 at 4:58 PM, Nikolai Geordzhev wrote: > FYI, I got the same/similar issue with a 10-line perl script, starting FS > without -hp option worked for me. > > > On Thu, Sep 10, 2009 at 7:28 PM, Rob Forman wrote: > >> Hi all, >> >> I wrote a small 10-line python script wrapping "txfax" ( >> http://pastebin.freeswitch.org/10274 >> ). Basically it originates a call with Session(), calls mod_fax txfax >> application, then hangs up. >> >> The weird thing is that this works fine the first I run it from >> fs_cli. When I run a second time however - it still works and sends >> the fax - but FS seg faults after the python script exits. >> >> Gdb shows something with Session destroy: >> >> #0 PYTHON::Session::destroy (this=0x83cd308) at freeswitch_python.cpp: >> 60 >> 60 Py_DECREF(Self); >> >> Here is the backtrace from the core: >> http://pastebin.freeswitch.org/10275 >> >> >> Is this a bug- or am I not setting up or cleaning up the session >> properly? I was going to open a jira but thought I'd see if I'm >> missing something obvious first. >> >> Thanks! >> Rob >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/a01a3827/attachment.html From diego.viola at gmail.com Thu Sep 10 10:38:09 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 10 Sep 2009 17:38:09 +0000 Subject: [Freeswitch-users] Implementing h extension in FS In-Reply-To: References: <87f2f3b90909100055v7e644bcbk62deb0d8e2201b11@mail.gmail.com> Message-ID: <86a32abc0909101038ycc36f71vda41ef82ec1dd606@mail.gmail.com> Here you have an example of how to get the CDR vars from the hangup event and send it to the db. http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/callcard/cdr.rb Regards, Diego On Thu, Sep 10, 2009 at 8:06 AM, Ken Rice wrote: > If you are just doing CDR processing the easiest ways are using the event > socket to trigger this on the hangup event and you will get all the data you > want or use mod_xml_cdr which will either drop a file or fire a web request. > You can use either of these methods to trigger a billing update > > > ------------------------------ > *From: *Michael Collins > *Reply-To: * > *Date: *Thu, 10 Sep 2009 00:55:02 -0700 > *To: * > *Subject: *Re: [Freeswitch-users] Implementing h extension in FS > > > > > On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir > wrote: > > HI, > > I'm newbie in FS, I want to know how to implement h extension of asterisk > to FS. As I listed down below; > > h => > { > NOOP("Call Completed with Carrier ${CARRIER}"); > goto add_cdr|h|1; > }; > > My other question is, which application/function/class is use in mod_perl > to check the channel status? i.e. busy, answer,hangup,ringing,etc. > > > Kindly advice me soon. > > -- > Regards, > > Ahmed Munir > > > It depends on what you are trying to accomplish, but the closest thing > you'll find in FS to the 'h' extension is the channel variable > api_hangup_hook which lets you launch an API at the end of the call. It > sounds like you are working on CDR processing. Please tell us more about > your application and we'll do our best to offer advice. > -MC > > > ------------------------------ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/11f9cc96/attachment-0001.html From pjintheusa at gmail.com Thu Sep 10 10:39:38 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 10 Sep 2009 10:39:38 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> Message-ID: <367751820909101039u73941f70mfb449273cc7bbdfc@mail.gmail.com> Surely this is because you are calling ev.serialized_string and ev is null if there is not event waiting? i.e. con.pop(0); returns null if no event is waiting. Therefore if(ev != null) Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); would work, alternatively calling Event ev = con.pop(1); would block until an event is available - and therefore ev cannot be null.... On Wed, Sep 9, 2009 at 2:01 PM, Josh Rivers wrote: > A new discovery: public bool Load() > { > ThreadPool.QueueUserWorkItem((o) => > { > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > EventConsumer con = new EventConsumer("all", ""); > while (true) > { > Event ev = con.pop(0); > if (ev == null) continue; > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > } > }); > return true; > } > Does not crash. (Adding the null check prevents crash.) The backgrounded > loop runs fine. Looks like the event object goes straight to pinvokes, so a > null result just crashes? > > I like the idea of a 'startup-script' for mod_managed. It would also be > excellent if there was an event or message informing the background code to > terminate nicely when the module reloads. > > --Josh > > On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk wrote: > >> >> I think the problem here is that the loader only keeps this method in >> scope >> until completion then it drops the remoted connection. Therefore you >> should >> not use threads in this method. Michael please correct me if I am wrong >> here. >> >> As an example of the failure simply just put a Sleep(10000) call in the >> thread and you will see the failure. >> >> As Michael said this method was only designed to allow the option to opt >> out >> of being loaded. >> >> In order to support this perhaps a configuration flag simular to the lua >> "startup-script" should be added. >> >> >> >> Here is the error I get with the loop I mentioned. -Josh >> [image: Capture.PNG] >> >> On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo >> wrote: >> >> > Hi, >> > >> > >> > >> > Can you please elaborate on the crash you receive when >> you >> > queue a thread during load? >> > >> > >> > >> > Thanks, >> > >> > Michael >> > >> > >> >> -- >> View this message in context: >> http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/50577281/attachment.html From nik.middleton at noblesolutions.co.uk Thu Sep 10 10:43:26 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Thu, 10 Sep 2009 18:43:26 +0100 Subject: [Freeswitch-users] Getting core dump from last night's build In-Reply-To: <191c3a030909100954r31aec836o4838070d9abc89f9@mail.gmail.com> References: <191c3a030909100954r31aec836o4838070d9abc89f9@mail.gmail.com> Message-ID: Yes I did a make current, and make sure, I've now reverted back to the latest release and all's well. Is there anything else I could try so as to ensure I've not got any issue like you suggest? Regards, ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: 10 September 2009 17:54 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Getting core dump from last night's build I wonder if maybe you have a build issue with an older mod_lua with a newer FreeSWITCH did you update via make current? On Thu, Sep 10, 2009 at 11:11 AM, Nik Middleton wrote: Hi Guys, I'm getting a core dump when running an lua script that's been fine for months In Freeswitch_lua.cpp line 92 is being called, but it's not clear what exactly this is doing lua_State *Session::getLUA() { if (!L) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Doh!\n"); } return L; } Anyone shed a light on this? Regards _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/7d7d4bee/attachment.html From anthony.minessale at gmail.com Thu Sep 10 10:53:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Sep 2009 12:53:52 -0500 Subject: [Freeswitch-users] Getting core dump from last night's build In-Reply-To: References: <191c3a030909100954r31aec836o4838070d9abc89f9@mail.gmail.com> Message-ID: <191c3a030909101053j44d716fehf7ee987e87d34cc4@mail.gmail.com> could you produce a minimal offending script we could test with on our box? On Thu, Sep 10, 2009 at 12:43 PM, Nik Middleton < nik.middleton at noblesolutions.co.uk> wrote: > Yes I did a make current, and make sure, > > > > I?ve now reverted back to the latest release and all?s well. > > > > Is there anything else I could try so as to ensure I?ve not got any issue > like you suggest? > > > > Regards, > > > > > > > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* 10 September 2009 17:54 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Getting core dump from last night's > build > > > > I wonder if maybe you have a build issue with an older mod_lua with a newer > FreeSWITCH > did you update via make current? > > On Thu, Sep 10, 2009 at 11:11 AM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > > Hi Guys, > > > > I?m getting a core dump when running an lua script that?s been fine for > months > > > > In Freeswitch_lua.cpp line 92 is being called, but it?s not clear what > exactly this is doing > > > > > > lua_State *Session::getLUA() > > { > > if (!L) { > > switch_log_printf(SWITCH_CHANNEL_LOG, > SWITCH_LOG_ERROR, "Doh!\n"); > > } > > return L; > > } > > > > Anyone shed a light on this? > > > > Regards > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/b56d212e/attachment-0001.html From anthony.minessale at gmail.com Thu Sep 10 11:23:23 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Sep 2009 13:23:23 -0500 Subject: [Freeswitch-users] Mod_fifo posision in queue In-Reply-To: <87f2f3b90909100037m45b93a9ei7ec62b4d1f84e986@mail.gmail.com> References: <8ccbff060909010737v4a8f17ep4d6175c2a2a2f866@mail.gmail.com> <8ccbff060909092332q1d06e74m8d5ca3a39ca4f572@mail.gmail.com> <87f2f3b90909100037m45b93a9ei7ec62b4d1f84e986@mail.gmail.com> Message-ID: <191c3a030909101123k6ccbf949q74fa22197f44517e@mail.gmail.com> I added a var for it (I hope properly) to r14806 On Thu, Sep 10, 2009 at 2:37 AM, Michael Collins wrote: > > > On Wed, Sep 9, 2009 at 11:32 PM, Dome Charoenyost wrote: > >> 2009/9/10 Michael Jerris : >> > You can use a phrase macro but I am not sure that we set the position >> > in a way that you can expand it for the macro. >> >> fifo don't have queue position variable ? >> >> > I have a feeling that this functionality is possible but it will probably > need to be added to mod_fifo.c as a feature request. Announcing the caller's > position in queue is really a function of the queue system itself and not > some third-party script. That being said, a question for the devs is this: > if I know the uuid of a call that's in a fifo queue can I displace the audio > (be it MOH, fifo_chime, etc.) and playback the caller's position (or any > other audio for that matter) without disrupting the queue system? Just > curious. > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/b0be1621/attachment.html From josh at radianttiger.com Thu Sep 10 11:47:39 2009 From: josh at radianttiger.com (Josh Rivers) Date: Thu, 10 Sep 2009 11:47:39 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DB2B59B3@mse17be1.mse17.exchange.ms> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5982@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B598E@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B59B3@mse17be1.mse17.exchange.ms> Message-ID: I'm only concerned with the difference in treatment. public class CrashFreeSWITCH : ILoadNotificationPlugin { public bool Load() { ThreadPool.QueueUserWorkItem((o) => { throw new NotImplementedException(); }); return true; } } Crashes the entire switch, terminating all calls and disconnecting from the PSTN. public class CrashFreeSWITCH : ILoadNotificationPlugin { public bool Load() { throw new NotImplementedException(); return true; } } Logs a message to the console and doesn't load the module, while leaving the switch operating. In my experience, exceptions in multi-threaded code: a) happen, b) are hard to diagnose. Is the best behavior for the environment to crash, providing no diagnostic information? That's hard in development, and even harder in production. I suppose 'terminate switch on fault' should be an option, to allow the operating system to restart the whole process on fault conditions, but if that is the intended result, shouldn't any fault do the same thing? What if the billing was happening in my second code block? Normally, I'd trap the ThreadException and UnhandledExceptions in my application, so that my code could choose the correct actions to perform should the application get into an unknown state. This can include terminating the whole application, but also logging diagnostic information, trying to save uncommitted data, and sending notifications of the failure. Is 'crash if it's a thread, but not if it's not' good because it's the way the module works now, or is it a better design for a reason I'm not understanding? On Wed, Sep 9, 2009 at 11:09 PM, Michael Giagnocavo wrote: > Well, a segfault in voicemail would do the same thing. > > > > Suppose your plugin runs a thread that does something important, like > billing or so on. That thread fails ? do you really want it to go on? > > > > Anyways, the solution is simple enough, handle your exceptions J. Every > plugin can decide what it wants to do here. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers > *Sent:* Wednesday, September 09, 2009 10:41 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > The question is whether the CLR should take down the whole phone server due > to an unhandled exception...definitely the CLR should terminate...but > shouldn't it just log the exception to the console, not crash the core? > > On Wed, Sep 9, 2009 at 6:30 PM, Michael Giagnocavo > wrote: > > That?s by design. If a thread fails, and there?s no handler, then the > application could be in a corrupted state, so the CLR takes down the > process. > > > > I think there is a .NET 1.0 compat switch you can enable in the config if > you like exceptions to be silently ignored J. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers > *Sent:* Wednesday, September 09, 2009 6:39 PM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > I have a new thought on the crashes...I'm able to crash FreeSWITCH any time > I like, just by having an exception in a thread. > > > > public class CrashFreeSWITCH : ILoadNotificationPlugin > > { > > public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => { throw new > NotImplementedException(); }); > > return true; > > } > > } > > > > Perhaps Application.ThreadException or AppDomain.UnhandledException need to > be trapped? > > > > On Wed, Sep 9, 2009 at 4:51 PM, Michael Giagnocavo > wrote: > > >Looks like the event object goes straight to pinvokes, so a null result > just crashes? > > > > If it?s null, you should get a NullReferenceException. The C# compiler > should callvirt the property getter and that?ll do a null check. If that > isn?t happening, that?d be an interesting optimization somewhere along the > line. > > > > -Michael > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers > *Sent:* Wednesday, September 09, 2009 3:01 PM > > > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > A new discovery: > > public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => > > { > > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > if (ev == null) continue; > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > } > > }); > > return true; > > } > > Does not crash. (Adding the null check prevents crash.) The backgrounded > loop runs fine. Looks like the event object goes straight to pinvokes, so a > null result just crashes? > > > > I like the idea of a 'startup-script' for mod_managed. It would also be > excellent if there was an event or message informing the background code to > terminate nicely when the module reloads. > > > > --Josh > > > > On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk wrote: > > > I think the problem here is that the loader only keeps this method in scope > until completion then it drops the remoted connection. Therefore you should > not use threads in this method. Michael please correct me if I am wrong > here. > > As an example of the failure simply just put a Sleep(10000) call in the > thread and you will see the failure. > > As Michael said this method was only designed to allow the option to opt > out > of being loaded. > > In order to support this perhaps a configuration flag simular to the lua > "startup-script" should be added. > > > > > Here is the error I get with the loop I mentioned. -Josh > [image: Capture.PNG] > > On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo > wrote: > > > Hi, > > > > > > > > Can you please elaborate on the crash you receive when > you > > queue a thread during load? > > > > > > > > Thanks, > > > > Michael > > > > > > -- > View this message in context: > http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/e8927667/attachment-0001.html From msc at freeswitch.org Thu Sep 10 11:50:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Sep 2009 11:50:23 -0700 Subject: [Freeswitch-users] Mod_fifo posision in queue In-Reply-To: <191c3a030909101123k6ccbf949q74fa22197f44517e@mail.gmail.com> References: <8ccbff060909010737v4a8f17ep4d6175c2a2a2f866@mail.gmail.com> <8ccbff060909092332q1d06e74m8d5ca3a39ca4f572@mail.gmail.com> <87f2f3b90909100037m45b93a9ei7ec62b4d1f84e986@mail.gmail.com> <191c3a030909101123k6ccbf949q74fa22197f44517e@mail.gmail.com> Message-ID: <87f2f3b90909101150w394c338m1b625d030c107508@mail.gmail.com> On Thu, Sep 10, 2009 at 11:23 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I added a var for it (I hope properly) to r14806 > > > Dome, Look for the fifo_position chan var in the latest SVN. Please try it out and see if it works, and make sure that it gets updated as the person's position in queue changes, etc. Please report back on your success. Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/083218f1/attachment.html From msc at freeswitch.org Thu Sep 10 12:01:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Sep 2009 12:01:29 -0700 Subject: [Freeswitch-users] 482 Request merged, in serial forking - Solved In-Reply-To: References: Message-ID: <87f2f3b90909101201q4e4908cdwb14f1ee5f2003d35@mail.gmail.com> Would you mind creating a wiki page on this? It would be good to document why you need this functionality, that is, what your business case is and how this solution helps you resolve your issue. Create your own page and then link to it from this page: http://wiki.freeswitch.org/wiki/Examples Thanks! -MC On Thu, Sep 10, 2009 at 6:31 AM, Humberto Quintana wrote: > Hi, > > I want to share my findings in making work Freeswitch as SBC when Kamailio > is doing serial forking. FS doesnt take any routing decision, it receives > the R-URI from Kamailio. > > After the 1st route failed, I was receiving "482 Request merged" for the > 2nd route. That was because the SIP transaction "lives" for 4 seconds in FS > after receiving the ACK for a 4xx, 5xx, 6xx reply (only when using UDP). > Also, Kamailio takes less than 1 second to try a new route for the same > call, thus making FS generates the 482 message. > > > To adapt both to my needs. I set in external.xml the timer T4 to 2500 > (default is 4000): > > > .... > > > > > In Kamailio, I'm executing the sleep function (from cfgutils module) in the > failure_route block: > > > failure_route[2] { > > ...Get your new ruri only for some reply codes... > > avp_pushto("$ruri", "$(avp(i:1511))"); > append_branch(); > t_on_failure("2"); #In case of a new failure this block will be > executed again > sleep("3"); > t_relay(); > return; > } > > > > May be it's not the best solution but it could be helpful for someone else. > > Best regards, > > Humberto > > > ------------------------------ > Faster Hotmail access now on the new MSN homepage. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/e33e1588/attachment.html From dome at tel.co.th Thu Sep 10 12:03:20 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 11 Sep 2009 02:03:20 +0700 Subject: [Freeswitch-users] Mod_fifo posision in queue In-Reply-To: <191c3a030909101123k6ccbf949q74fa22197f44517e@mail.gmail.com> References: <8ccbff060909010737v4a8f17ep4d6175c2a2a2f866@mail.gmail.com> <8ccbff060909092332q1d06e74m8d5ca3a39ca4f572@mail.gmail.com> <87f2f3b90909100037m45b93a9ei7ec62b4d1f84e986@mail.gmail.com> <191c3a030909101123k6ccbf949q74fa22197f44517e@mail.gmail.com> Message-ID: <8ccbff060909101203m5625a991y32138e32913959c5@mail.gmail.com> 2009/9/11 Anthony Minessale : > I added a var for it (I hope properly) to r14806 Thanks. I'll try Dome C. > > > On Thu, Sep 10, 2009 at 2:37 AM, Michael Collins wrote: >> >> >> On Wed, Sep 9, 2009 at 11:32 PM, Dome Charoenyost wrote: >>> >>> 2009/9/10 Michael Jerris : >>> > You can use a phrase macro but I am not sure that we set the position >>> > in a way that you can expand it for the macro. >>> >>> fifo don't have queue position variable ? >>> >> >> I have a feeling that this functionality is possible but it will probably >> need to be added to mod_fifo.c as a feature request. Announcing the caller's >> position in queue is really a function of the queue system itself and not >> some third-party script. That being said, a question for the devs is this: >> if I know the uuid of a call that's in a fifo queue can I displace the audio >> (be it MOH, fifo_chime, etc.) and playback the caller's position (or any >> other audio for that matter) without disrupting the queue system? Just >> curious. >> -MC >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From diego.viola at gmail.com Thu Sep 10 12:32:17 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 10 Sep 2009 19:32:17 +0000 Subject: [Freeswitch-users] Mod_fifo posision in queue In-Reply-To: <8ccbff060909101203m5625a991y32138e32913959c5@mail.gmail.com> References: <8ccbff060909010737v4a8f17ep4d6175c2a2a2f866@mail.gmail.com> <8ccbff060909092332q1d06e74m8d5ca3a39ca4f572@mail.gmail.com> <87f2f3b90909100037m45b93a9ei7ec62b4d1f84e986@mail.gmail.com> <191c3a030909101123k6ccbf949q74fa22197f44517e@mail.gmail.com> <8ccbff060909101203m5625a991y32138e32913959c5@mail.gmail.com> Message-ID: <86a32abc0909101232k656be844qfa19bb7a679a2e02@mail.gmail.com> Lets make sure we add it on the wiki too =D. Diego On Thu, Sep 10, 2009 at 7:03 PM, Dome Charoenyost wrote: > 2009/9/11 Anthony Minessale : > > I added a var for it (I hope properly) to r14806 > > Thanks. I'll try > > Dome C. > > > > > > > On Thu, Sep 10, 2009 at 2:37 AM, Michael Collins > wrote: > >> > >> > >> On Wed, Sep 9, 2009 at 11:32 PM, Dome Charoenyost > wrote: > >>> > >>> 2009/9/10 Michael Jerris : > >>> > You can use a phrase macro but I am not sure that we set the position > >>> > in a way that you can expand it for the macro. > >>> > >>> fifo don't have queue position variable ? > >>> > >> > >> I have a feeling that this functionality is possible but it will > probably > >> need to be added to mod_fifo.c as a feature request. Announcing the > caller's > >> position in queue is really a function of the queue system itself and > not > >> some third-party script. That being said, a question for the devs is > this: > >> if I know the uuid of a call that's in a fifo queue can I displace the > audio > >> (be it MOH, fifo_chime, etc.) and playback the caller's position (or any > >> other audio for that matter) without disrupting the queue system? Just > >> curious. > >> -MC > >> > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/b7cae2a1/attachment.html From msc at freeswitch.org Thu Sep 10 13:21:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Sep 2009 13:21:35 -0700 Subject: [Freeswitch-users] OpenZAP No Audio In Outbound FXO for 8-10 Seconds In-Reply-To: <4AA8787C.2050105@digilord.net> References: <4AA8787C.2050105@digilord.net> Message-ID: <87f2f3b90909101321s488cde9fj6e0ac7da76bed0a@mail.gmail.com> Is this the same card that had issues and Rhino had to re-flash? -MC On Wed, Sep 9, 2009 at 8:54 PM, Dan wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello all! > I am having the following issue. > > When I dial out over a FXO port (analog) for the first 8-10 seconds I > get no audio. If I wait I will eventually hear something. On inbound > calls audio works great in both directions. I used ztmonitor to record > from the channel and there was in fact audio there. In the recording > you can clearly tell when audio starts flowing as you can hear me twice > in the recording (I was calling to my cell from a phone on my desk). > > Any help here would be greatly appreciated! > > FS is current SVN > OpenZAP is current SVN. > Hardware: Rhino 8 port w/6 FXO and 2 FXS ports installed. > > Daniel Morrigan > > "He who says it cannot be done is interrupting the one doing it." - > Chinese Proverb > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.9 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org > > iEYEARECAAYFAkqoeHoACgkQ3JaPN6smlEXhdQCfRf5GBKwVrtZWsCS4J1fug2e7 > SEEAn1FkyhBPKiXnfUiXgvd2ggqUW87w > =OFH/ > -----END PGP SIGNATURE----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/83c5448b/attachment-0001.html From msc at freeswitch.org Thu Sep 10 13:22:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Sep 2009 13:22:23 -0700 Subject: [Freeswitch-users] Mod_fifo posision in queue In-Reply-To: <86a32abc0909101232k656be844qfa19bb7a679a2e02@mail.gmail.com> References: <8ccbff060909010737v4a8f17ep4d6175c2a2a2f866@mail.gmail.com> <8ccbff060909092332q1d06e74m8d5ca3a39ca4f572@mail.gmail.com> <87f2f3b90909100037m45b93a9ei7ec62b4d1f84e986@mail.gmail.com> <191c3a030909101123k6ccbf949q74fa22197f44517e@mail.gmail.com> <8ccbff060909101203m5625a991y32138e32913959c5@mail.gmail.com> <86a32abc0909101232k656be844qfa19bb7a679a2e02@mail.gmail.com> Message-ID: <87f2f3b90909101322v68e49f74jbc4cf2052e1811c3@mail.gmail.com> On Thu, Sep 10, 2009 at 12:32 PM, Diego Viola wrote: > Lets make sure we add it on the wiki too =D. > > Yep, as soon as we verify its functionality we'll wikify it. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/302b3d6f/attachment.html From email.list.subscriber at gmail.com Thu Sep 10 13:47:49 2009 From: email.list.subscriber at gmail.com (email lists) Date: Thu, 10 Sep 2009 16:47:49 -0400 Subject: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? Message-ID: <4aa965af.161bf30a.61f8.7c7d@mx.google.com> Hello, Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate RADIUS messages being generated for individual calls (sample messages for one call below). Looking at the "Acct-Unique-Session-Id" and "Acct-Session-Id" fields, it would appear that perhaps each call leg results in a pair of start/stop RADIUS messages; is this the expected behavior? If so, is there a way to disable RADIUS messaging for what I presume is the "ingress" or A leg of the call? Any leads would be appreciated. Thanks in advance. Vladimir Thu Sep 10 10:37:25 2009 Acct-Status-Type = Start Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" User-Name = "8135793256" Freeswitch-Src = "8135793256" Freeswitch-CLID = "sipp" Freeswitch-Dst = "14043297226" Freeswitch-Dialplan = "XML" Framed-IP-Address = 50.46.50.55 Freeswitch-Context = "public" Freeswitch-Source = "mod_sofia" Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = "097c8472ff7bcec7" Timestamp = 1252604245 Request-Authenticator = Verified Thu Sep 10 10:37:25 2009 Acct-Status-Type = Start Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" User-Name = "8135793256" Freeswitch-Src = "8135793256" Freeswitch-CLID = "sipp" Freeswitch-Dst = "14043297226 at x.x.x.x" Freeswitch-Dialplan = "XML" Framed-IP-Address = 50.46.50.55 Freeswitch-Context = "public" Freeswitch-Source = "mod_sofia" Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = "53f729e173e8c0a9" Timestamp = 1252604245 Request-Authenticator = Verified Thu Sep 10 10:37:57 2009 Acct-Status-Type = Stop Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" Freeswitch-Hangupcause = Normal-Unspecified User-Name = "8135793256" Freeswitch-Src = "8135793256" Freeswitch-CLID = "sipp" Freeswitch-Dst = "14043297226" Freeswitch-Dialplan = "XML" Framed-IP-Address = 50.46.50.55 Freeswitch-Context = "public" Freeswitch-Source = "mod_sofia" Freeswitch-Lastapp = "bridge" Freeswitch-Billusec = 32029926 Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" Freeswitch-Callanswerdate = "2009-09-10T10:22:00.319197-0700" Freeswitch-Callenddate = "2009-09-10T10:22:32.349123-0700" Acct-Session-Time = 32 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = "097c8472ff7bcec7" Timestamp = 1252604277 Request-Authenticator = Verified Thu Sep 10 10:38:02 2009 Acct-Status-Type = Stop Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" Freeswitch-Hangupcause = Normal-Clearing User-Name = "8135793256" Freeswitch-Src = "8135793256" Freeswitch-CLID = "sipp" Freeswitch-Dst = "14043297226 at x.x.x.x" Freeswitch-Dialplan = "XML" Framed-IP-Address = 50.46.50.55 Freeswitch-Context = "public" Freeswitch-Source = "mod_sofia" Freeswitch-Billusec = 32049973 Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" Freeswitch-Callanswerdate = "2009-09-10T10:22:00.289136-0700" Freeswitch-Callenddate = "2009-09-10T10:22:32.339109-0700" Acct-Session-Time = 32 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = "53f729e173e8c0a9" Timestamp = 1252604282 Request-Authenticator = Verified -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/b916a1a1/attachment-0001.html From pjintheusa at gmail.com Thu Sep 10 14:24:22 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 10 Sep 2009 17:24:22 -0400 Subject: [Freeswitch-users] uuid_exists - does it still exist? In-Reply-To: <87f2f3b90909031456s76a12874xe0a6ebfa26babb10@mail.gmail.com> References: <367751820909031255l1c6947c6t1bf577c77551f9b3@mail.gmail.com> <87f2f3b90909031456s76a12874xe0a6ebfa26babb10@mail.gmail.com> Message-ID: <367751820909101424h787e7757vd67649562c8d07f8@mail.gmail.com> Strangely - the uuid_getvar uuid workaround does not work for me. This is the result of: apiResult = fsApi.ExecuteString(string.Format("uuid_getvar {0} uuid", call.Uuid)); Log.WriteLine(LogLevel.Alert, "RESULT: uuid_getvar {0} is: {1}", call.Uuid, apiResult); returns a different uuid????? 2009-09-10 17:12:26.953125 [ALERT] switch_cpp.cpp:1130 RESULT: uuid_getvar 54dd24be-b0da-684f-acee-38c7530b4c2b is: 1a0e83db-240c-ac4e-ae45-bf5d5b46f5c3 the passed uuid is vaild however: 2009-09-10 17:12:26.953125 [ALERT] switch_cpp.cpp:1130 RESULT: uuid_kill 54dd24be-b0da-684f-acee-38c7530b4c2b is: +OK Can a call leg have two uuids?? On Thu, Sep 3, 2009 at 5:56 PM, Michael Collins wrote: > > > On Thu, Sep 3, 2009 at 2:23 PM, Benedikt Fraunhofer < > fraunhofer.lists.freeswitch-001 at traced.net> wrote: > >> Hi, >> >> >> > Usage: uuid_exists >> > However when I call via an API call I get: >> > INVALID COMMAND! >> > I also don't see it in MOD_COMMAND.C >> >> As a workaround or if your are unable to upgrade, you can use >> "uuid_getvar [some_uuid] thisVariableDoesNotExist" >> ("thisVariableDoesNotExist" is any variable you can think of, a valid >> one or literally 'thisVariableDoesNotExist' :) >> You'll either get an error that this channel does not exist any longer >> or "undef" for the channel variable. >> The api will return "+OK" in case the channel still exists, and >> "ERROR" in case it does not. >> >> Beni. >> >> > You could even do this: > uuid_getvar uuid > > If it exists then the return will be the uuid. :) > > Although I must say I recommend this instead: > make current > > :) > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/cfa30564/attachment.html From anthony.minessale at gmail.com Thu Sep 10 14:39:01 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 10 Sep 2009 16:39:01 -0500 Subject: [Freeswitch-users] Getting core dump from last night's build In-Reply-To: <191c3a030909101053j44d716fehf7ee987e87d34cc4@mail.gmail.com> References: <191c3a030909100954r31aec836o4838070d9abc89f9@mail.gmail.com> <191c3a030909101053j44d716fehf7ee987e87d34cc4@mail.gmail.com> Message-ID: <191c3a030909101439t1a7a1bafp502e16cfc4d3588@mail.gmail.com> nevermind, I think i fixed it in latest trunk, please test... On Thu, Sep 10, 2009 at 12:53 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > could you produce a minimal offending script we could test with on our box? > > > > On Thu, Sep 10, 2009 at 12:43 PM, Nik Middleton < > nik.middleton at noblesolutions.co.uk> wrote: > >> Yes I did a make current, and make sure, >> >> >> >> I?ve now reverted back to the latest release and all?s well. >> >> >> >> Is there anything else I could try so as to ensure I?ve not got any issue >> like you suggest? >> >> >> >> Regards, >> >> >> >> >> >> >> ------------------------------ >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony >> Minessale >> *Sent:* 10 September 2009 17:54 >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Getting core dump from last night's >> build >> >> >> >> I wonder if maybe you have a build issue with an older mod_lua with a >> newer FreeSWITCH >> did you update via make current? >> >> On Thu, Sep 10, 2009 at 11:11 AM, Nik Middleton < >> nik.middleton at noblesolutions.co.uk> wrote: >> >> Hi Guys, >> >> >> >> I?m getting a core dump when running an lua script that?s been fine for >> months >> >> >> >> In Freeswitch_lua.cpp line 92 is being called, but it?s not clear what >> exactly this is doing >> >> >> >> >> >> lua_State *Session::getLUA() >> >> { >> >> if (!L) { >> >> switch_log_printf(SWITCH_CHANNEL_LOG, >> SWITCH_LOG_ERROR, "Doh!\n"); >> >> } >> >> return L; >> >> } >> >> >> >> Anyone shed a light on this? >> >> >> >> Regards >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/b4ecb6e4/attachment.html From msc at freeswitch.org Thu Sep 10 15:01:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Sep 2009 15:01:23 -0700 Subject: [Freeswitch-users] uuid_exists - does it still exist? In-Reply-To: <367751820909101424h787e7757vd67649562c8d07f8@mail.gmail.com> References: <367751820909031255l1c6947c6t1bf577c77551f9b3@mail.gmail.com> <87f2f3b90909031456s76a12874xe0a6ebfa26babb10@mail.gmail.com> <367751820909101424h787e7757vd67649562c8d07f8@mail.gmail.com> Message-ID: <87f2f3b90909101501k1d459ddcl15b29018f2d46521@mail.gmail.com> As a sanity test, can you bring up a bridged call and run this test directly from the command line and compare the results to what's in your script? -MC On Thu, Sep 10, 2009 at 2:24 PM, Phillip Jones wrote: > Strangely - the uuid_getvar uuid workaround > does not work for me. > > This is the result of: > > apiResult = fsApi.ExecuteString(string.Format("uuid_getvar {0} uuid", > call.Uuid)); > Log.WriteLine(LogLevel.Alert, "RESULT: uuid_getvar {0} is: {1}", call.Uuid, > apiResult); > > returns a different uuid????? > > 2009-09-10 17:12:26.953125 [ALERT] switch_cpp.cpp:1130 RESULT: uuid_getvar > 54dd24be-b0da-684f-acee-38c7530b4c2b is: > 1a0e83db-240c-ac4e-ae45-bf5d5b46f5c3 > > the passed uuid is vaild however: > > 2009-09-10 17:12:26.953125 [ALERT] switch_cpp.cpp:1130 RESULT: uuid_kill > 54dd24be-b0da-684f-acee-38c7530b4c2b is: +OK > > Can a call leg have two uuids?? > > > On Thu, Sep 3, 2009 at 5:56 PM, Michael Collins wrote: > >> >> >> On Thu, Sep 3, 2009 at 2:23 PM, Benedikt Fraunhofer < >> fraunhofer.lists.freeswitch-001 at traced.net> wrote: >> >>> Hi, >>> >>> >>> > Usage: uuid_exists >>> > However when I call via an API call I get: >>> > INVALID COMMAND! >>> > I also don't see it in MOD_COMMAND.C >>> >>> As a workaround or if your are unable to upgrade, you can use >>> "uuid_getvar [some_uuid] thisVariableDoesNotExist" >>> ("thisVariableDoesNotExist" is any variable you can think of, a valid >>> one or literally 'thisVariableDoesNotExist' :) >>> You'll either get an error that this channel does not exist any longer >>> or "undef" for the channel variable. >>> The api will return "+OK" in case the channel still exists, and >>> "ERROR" in case it does not. >>> >>> Beni. >>> >>> >> You could even do this: >> uuid_getvar uuid >> >> If it exists then the return will be the uuid. :) >> >> Although I must say I recommend this instead: >> make current >> >> :) >> -MC >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/c03a77e5/attachment-0001.html From jan.kubr at gmail.com Thu Sep 10 15:14:19 2009 From: jan.kubr at gmail.com (Jan Kubr) Date: Fri, 11 Sep 2009 00:14:19 +0200 Subject: [Freeswitch-users] Attended transfer - no audio Message-ID: <698401620909101514h15214d47r78130a81e3a81c5d@mail.gmail.com> Hi, we have a Freeswitch server on a public IP and a few phones behind NAT. The phones are configured to use STUN and can register and call each other fine. The problem is that after attended transfer (using the mechanism the phones provide - REFER) is finished, the two parties can't hear each other. This problem doesn't occur when the phones are in the same subnet as Freeswitch. I know this isn't enough information to solve the problem, but do you have any hints on how to debug this? Are there any specific Freeswitch settings that could help us? Thanks, Jan From mgg at giagnocavo.net Thu Sep 10 15:19:53 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 10 Sep 2009 18:19:53 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DAFA4B37@mse17be1.mse17.exchange.ms> <1252526228140-3613195.post@n2.nabble.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5982@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B598E@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B59B3@mse17be1.mse17.exchange.ms> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DB2B5B8E@mse17be1.mse17.exchange.ms> Well, we have absolutely no idea what the background thread is doing. It might be critical, and the fix is trivial: put a try/catch on it. This is the model all .NET applications have. Background threads doing bad things should always take down the process. However, that's a good point about Load() failing. The approach taken is more or less how FreeSWITCH handles things in general now. If a module has an error, the switch just logs and goes on. I'm not really in favour of this, and suggested at least a "required" attribute in the modules.conf that would prevent the switch from loading if the module fails. The fix is probably to create an attribute you can apply to the plugin classes that indicate what kind of failure handling you want. For the assembly, we'd add an attribute with an enumeration like: - Default (scan, call ILoadNotificationPlugin, log errors if they occur) - NoLoad (don't load the assembly) - Critical (stop the switch if there's an exception during loading) That'd provide the control you want for loading. We could do something similar for App/Api plugins. I want to move ILoadNotificationPlugin from being this "catch all" thing that controls the entire assembly to something that can be used to fire up code; effectively "OnLoad" and "OnUnload". To dynamically control loading, we'll probably reflect on the individual plugins looking for attributes or perhaps some sort of static load function. How's that sound? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Thursday, September 10, 2009 12:48 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET I'm only concerned with the difference in treatment. public class CrashFreeSWITCH : ILoadNotificationPlugin { public bool Load() { ThreadPool.QueueUserWorkItem((o) => { throw new NotImplementedException(); }); return true; } } Crashes the entire switch, terminating all calls and disconnecting from the PSTN. public class CrashFreeSWITCH : ILoadNotificationPlugin { public bool Load() { throw new NotImplementedException(); return true; } } Logs a message to the console and doesn't load the module, while leaving the switch operating. In my experience, exceptions in multi-threaded code: a) happen, b) are hard to diagnose. Is the best behavior for the environment to crash, providing no diagnostic information? That's hard in development, and even harder in production. I suppose 'terminate switch on fault' should be an option, to allow the operating system to restart the whole process on fault conditions, but if that is the intended result, shouldn't any fault do the same thing? What if the billing was happening in my second code block? Normally, I'd trap the ThreadException and UnhandledExceptions in my application, so that my code could choose the correct actions to perform should the application get into an unknown state. This can include terminating the whole application, but also logging diagnostic information, trying to save uncommitted data, and sending notifications of the failure. Is 'crash if it's a thread, but not if it's not' good because it's the way the module works now, or is it a better design for a reason I'm not understanding? On Wed, Sep 9, 2009 at 11:09 PM, Michael Giagnocavo > wrote: Well, a segfault in voicemail would do the same thing. Suppose your plugin runs a thread that does something important, like billing or so on. That thread fails - do you really want it to go on? Anyways, the solution is simple enough, handle your exceptions :). Every plugin can decide what it wants to do here. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Wednesday, September 09, 2009 10:41 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET The question is whether the CLR should take down the whole phone server due to an unhandled exception...definitely the CLR should terminate...but shouldn't it just log the exception to the console, not crash the core? On Wed, Sep 9, 2009 at 6:30 PM, Michael Giagnocavo > wrote: That's by design. If a thread fails, and there's no handler, then the application could be in a corrupted state, so the CLR takes down the process. I think there is a .NET 1.0 compat switch you can enable in the config if you like exceptions to be silently ignored :). -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Wednesday, September 09, 2009 6:39 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET I have a new thought on the crashes...I'm able to crash FreeSWITCH any time I like, just by having an exception in a thread. public class CrashFreeSWITCH : ILoadNotificationPlugin { public bool Load() { ThreadPool.QueueUserWorkItem((o) => { throw new NotImplementedException(); }); return true; } } Perhaps Application.ThreadException or AppDomain.UnhandledException need to be trapped? On Wed, Sep 9, 2009 at 4:51 PM, Michael Giagnocavo > wrote: >Looks like the event object goes straight to pinvokes, so a null result just crashes? If it's null, you should get a NullReferenceException. The C# compiler should callvirt the property getter and that'll do a null check. If that isn't happening, that'd be an interesting optimization somewhere along the line. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Wednesday, September 09, 2009 3:01 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET A new discovery: public bool Load() { ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); if (ev == null) continue; Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); } }); return true; } Does not crash. (Adding the null check prevents crash.) The backgrounded loop runs fine. Looks like the event object goes straight to pinvokes, so a null result just crashes? I like the idea of a 'startup-script' for mod_managed. It would also be excellent if there was an event or message informing the background code to terminate nicely when the module reloads. --Josh On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk > wrote: I think the problem here is that the loader only keeps this method in scope until completion then it drops the remoted connection. Therefore you should not use threads in this method. Michael please correct me if I am wrong here. As an example of the failure simply just put a Sleep(10000) call in the thread and you will see the failure. As Michael said this method was only designed to allow the option to opt out of being loaded. In order to support this perhaps a configuration flag simular to the lua "startup-script" should be added. Here is the error I get with the loop I mentioned. -Josh [image: Capture.PNG] On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo >wrote: > Hi, > > > > Can you please elaborate on the crash you receive when you > queue a thread during load? > > > > Thanks, > > Michael > > -- View this message in context: http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/6879aa38/attachment-0001.html From msc at freeswitch.org Thu Sep 10 17:15:41 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 10 Sep 2009 17:15:41 -0700 Subject: [Freeswitch-users] REMINDER: Weekly FreeSWITCH Conference Scheduled for Friday, 11AM CST Message-ID: <87f2f3b90909101715lf327c60i51d18452a5f4edd5@mail.gmail.com> FYI, We are on schedule for the weekly conference call. Please be sure to join us! More information is available here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11 Talk to you all tomorrow! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/c2c8da27/attachment.html From brian at freeswitch.org Thu Sep 10 19:13:26 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Sep 2009 21:13:26 -0500 Subject: [Freeswitch-users] uuid_exists - does it still exist? In-Reply-To: <367751820909101424h787e7757vd67649562c8d07f8@mail.gmail.com> References: <367751820909031255l1c6947c6t1bf577c77551f9b3@mail.gmail.com> <87f2f3b90909031456s76a12874xe0a6ebfa26babb10@mail.gmail.com> <367751820909101424h787e7757vd67649562c8d07f8@mail.gmail.com> Message-ID: What puzzles me is why you can't just update to SVN trunk to get uuid_exists in the first place? Its going to be 1.0.5 the more people we have testing the faster that happens. /b On Sep 10, 2009, at 4:24 PM, Phillip Jones wrote: > Strangely - the uuid_getvar uuid > workaround does not work for me. > > This is the result of: > > apiResult = fsApi.ExecuteString(string.Format("uuid_getvar {0} > uuid", call.Uuid)); > Log.WriteLine(LogLevel.Alert, "RESULT: uuid_getvar {0} is: {1}", > call.Uuid, apiResult); > > returns a different uuid????? > > 2009-09-10 17:12:26.953125 [ALERT] switch_cpp.cpp:1130 RESULT: > uuid_getvar 54dd24be-b0da-684f-acee-38c7530b4c2b is: 1a0e83db-240c- > ac4e-ae45-bf5d5b46f5c3 > > the passed uuid is vaild however: > > 2009-09-10 17:12:26.953125 [ALERT] switch_cpp.cpp:1130 RESULT: > uuid_kill 54dd24be-b0da-684f-acee-38c7530b4c2b is: +OK > > Can a call leg have two uuids?? From brian at freeswitch.org Thu Sep 10 19:17:02 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Sep 2009 21:17:02 -0500 Subject: [Freeswitch-users] Implementing h extension in FS In-Reply-To: References: Message-ID: There will NEVER be an easy way to do billing inline and we never support such crazy because its WRONG. That said... You're not the first to approach FreeSWITCH with an Asterisk mentality. Billing inline = NO Billing direct to DB = NO Post Processing = YES, do this. Nibble/Heartbeat billing via event socket = YES, (this is rather powerful if you take the time to learn it.) The above is just my advice... /b PS: see mod_nibblebill On Sep 10, 2009, at 12:16 AM, Ahmed Munir wrote: > HI, > > I'm newbie in FS, I want to know how to implement h extension of > asterisk to FS. As I listed down below; > > h => > { > NOOP("Call Completed with Carrier ${CARRIER}"); > goto add_cdr|h|1; > }; > > My other question is, which application/function/class is use in > mod_perl to check the channel status? i.e. busy, > answer,hangup,ringing,etc. > > > Kindly advice me soon. > > -- > Regards, > > Ahmed Munir > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Thu Sep 10 19:17:39 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 10 Sep 2009 21:17:39 -0500 Subject: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? In-Reply-To: <4aa965af.161bf30a.61f8.7c7d@mx.google.com> References: <4aa965af.161bf30a.61f8.7c7d@mx.google.com> Message-ID: You'll get this for transfers, and each leg... Can you elaborate on the call scenario more? /b On Sep 10, 2009, at 3:47 PM, email lists wrote: > Hello, > > Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate > RADIUS messages being generated for individual calls (sample > messages for one call below). Looking at the "Acct-Unique-Session- > Id" and "Acct-Session-Id" fields, it would appear that perhaps each > call leg results in a pair of start/stop RADIUS messages; is this > the expected behavior? If so, is there a way to disable RADIUS > messaging for what I presume is the "ingress" or A leg of the call? > > Any leads would be appreciated. > > Thanks in advance. > > Vladimir > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090910/c5dee60e/attachment.html From grae at digilord.net Thu Sep 10 20:13:38 2009 From: grae at digilord.net (Dan) Date: Thu, 10 Sep 2009 20:13:38 -0700 Subject: [Freeswitch-users] OpenZAP No Audio In Outbound FXO for 8-10 Seconds In-Reply-To: <87f2f3b90909101321s488cde9fj6e0ac7da76bed0a@mail.gmail.com> References: <4AA8787C.2050105@digilord.net> <87f2f3b90909101321s488cde9fj6e0ac7da76bed0a@mail.gmail.com> Message-ID: <4AA9C062.5000302@digilord.net> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 MC, Rhino ended up not needing to reflash it. I am trying to sort out why the driver is segfaulting with Zaptel 1.4.12.1. One of the guys at Rhino has a FS box with the same card in it. I am going to try to replicate his settings tomorrow. Digilord Michael Collins wrote: > Is this the same card that had issues and Rhino had to re-flash? > -MC > > On Wed, Sep 9, 2009 at 8:54 PM, Dan wrote: > > Hello all! > I am having the following issue. > > When I dial out over a FXO port (analog) for the first 8-10 seconds I > get no audio. If I wait I will eventually hear something. On inbound > calls audio works great in both directions. I used ztmonitor to record > from the channel and there was in fact audio there. In the recording > you can clearly tell when audio starts flowing as you can hear me twice > in the recording (I was calling to my cell from a phone on my desk). > > Any help here would be greatly appreciated! > > FS is current SVN > OpenZAP is current SVN. > Hardware: Rhino 8 port w/6 FXO and 2 FXS ports installed. > > Daniel Morrigan > > "He who says it cannot be done is interrupting the one doing it." - > Chinese Proverb >> _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.9 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iEYEARECAAYFAkqpwGAACgkQ3JaPN6smlEUEMwCeKhK3XSoXZZ3fdlkD2Ul06iOi DwQAoORpwll1zhDsrTm7N2oC1KKDvPqk =cvT3 -----END PGP SIGNATURE----- From ahmedmunir007 at gmail.com Thu Sep 10 21:40:38 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Fri, 11 Sep 2009 10:40:38 +0600 Subject: [Freeswitch-users] Implementing h extension in FS Message-ID: Thanks for reply, well actually I'm doing billing after call hangup. If h extension is interupts I'm sending to it to addcdr context which interupts perl script for billing purpose. As I'm listing down below asterisk configuration; h => { NOOP("Call Completed with Carrier ${CARRIER}"); goto add_cdr|h|1; }; context add_cdr { _X. => { Hangup(); }; h => { Set(CALL_END_TIME=${EPOCH}); //&print_variables(); NOOP("Call Ended: Card:${CARDNUM} Destination:${CALLEDNUM} Caller-ID:${CALLERID(num)}"); if (${DIALEXECUTED}=YES) { NOOP("Dial-Status:${DIALSTATUS}"); }else { NOOP("Dial was not Executed"); }; DeadAGI(/vopium/agi/billing.pl); NOOP(); }; }; Kindly advice me how I pass/translate h extension in FS in this situation i.e. or there is other way around??? ------------------------------ *From: *Michael Collins *Reply-To: * *Date: *Thu, 10 Sep 2009 00:55:02 -0700 *To: * *Subject: *Re: [Freeswitch-users] Implementing h extension in FS On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir wrote: HI, I'm newbie in FS, I want to know how to implement h extension of asterisk to FS. As I listed down below; h => { NOOP("Call Completed with Carrier ${CARRIER}"); goto add_cdr|h|1; }; My other question is, which application/function/class is use in mod_perl to check the channel status? i.e. busy, answer,hangup,ringing,etc. Kindly advice me soon. -- Regards, Ahmed Munir It depends on what you are trying to accomplish, but the closest thing you'll find in FS to the 'h' extension is the channel variable api_hangup_hook which lets you launch an API at the end of the call. It sounds like you are working on CDR processing. Please tell us more about your application and we'll do our best to offer advice. -MC -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/ea5d9add/attachment.html From anatoliy at kounitskiy.com Fri Sep 11 00:26:29 2009 From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy) Date: Fri, 11 Sep 2009 10:26:29 +0300 Subject: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? In-Reply-To: <4aa965af.161bf30a.61f8.7c7d@mx.google.com> References: <4aa965af.161bf30a.61f8.7c7d@mx.google.com> Message-ID: <4AA9FBA5.5090403@kounitskiy.com> It's normal to have to two records for a call - Start and Stop message. From what i see - you have one start and stop for each leg of the call. Regards, AK email lists wrote: > > Hello, > > > > Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate > RADIUS messages being generated for individual calls (sample messages > for one call below). Looking at the "Acct-Unique-Session-Id" and > "Acct-Session-Id" fields, it would appear that perhaps each call leg > results in a pair of start/stop RADIUS messages; is this the expected > behavior? If so, is there a way to disable RADIUS messaging for what > I presume is the "ingress" or A leg of the call? > > > > Any leads would be appreciated. > > > > Thanks in advance. > > > > Vladimir > > > > Thu Sep 10 10:37:25 2009 > > Acct-Status-Type = Start > > Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "097c8472ff7bcec7" > > Timestamp = 1252604245 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:37:25 2009 > > Acct-Status-Type = Start > > Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226 at x.x.x.x" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "53f729e173e8c0a9" > > Timestamp = 1252604245 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:37:57 2009 > > Acct-Status-Type = Stop > > Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" > > Freeswitch-Hangupcause = Normal-Unspecified > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Lastapp = "bridge" > > Freeswitch-Billusec = 32029926 > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" > > Freeswitch-Callanswerdate = "2009-09-10T10:22:00.319197-0700" > > Freeswitch-Callenddate = "2009-09-10T10:22:32.349123-0700" > > Acct-Session-Time = 32 > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "097c8472ff7bcec7" > > Timestamp = 1252604277 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:38:02 2009 > > Acct-Status-Type = Stop > > Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" > > Freeswitch-Hangupcause = Normal-Clearing > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226 at x.x.x.x" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Billusec = 32049973 > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" > > Freeswitch-Callanswerdate = "2009-09-10T10:22:00.289136-0700" > > Freeswitch-Callenddate = "2009-09-10T10:22:32.339109-0700" > > Acct-Session-Time = 32 > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "53f729e173e8c0a9" > > Timestamp = 1252604282 > > Request-Authenticator = Verified** > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From fraunhofer.lists.freeswitch-001 at traced.net Fri Sep 11 00:32:41 2009 From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer) Date: Fri, 11 Sep 2009 09:32:41 +0200 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <191c3a030909020842j61369a45va923080ea0a88b40@mail.gmail.com> <191c3a030909040757y93c105bs8939559f90142b9e@mail.gmail.com> Message-ID: Hello *, "sched_api ... &" works, too. Thx again and looking forward to the next bug :) Beni. From raffaele.p.guidi at gmail.com Fri Sep 11 00:55:42 2009 From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi) Date: Fri, 11 Sep 2009 09:55:42 +0200 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DB2B5B8E@mse17be1.mse17.exchange.ms> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <1252526228140-3613195.post@n2.nabble.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5982@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B598E@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B59B3@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5B8E@mse17be1.mse17.exchange.ms> Message-ID: *I want to move ILoadNotificationPlugin from being this ?catch all? thing that controls the entire assembly to something that can be used to fire up code; effectively ?OnLoad? and ?OnUnload?. To dynamically control loading, we?ll probably reflect on the individual plugins looking for attributes or perhaps some sort of static load function.* I meant to do something like that probably using spring to inject method names to be invoked. Also event listening (wich is I believe a generic need) could be managed this way and benefit from some abstraction. con.pop(1) is probably the most frequently written line by every plugin developer, probably some abstraction (an event started with his thread and the fs event passed as an argument?) could make code more elegant On Fri, Sep 11, 2009 at 00:19, Michael Giagnocavo wrote: > Well, we have absolutely no idea what the background thread is doing. It > might be critical, and the fix is trivial: put a try/catch on it. This is > the model all .NET applications have. Background threads doing bad things > should always take down the process. > > > > However, that?s a good point about Load() failing. The approach taken is > more or less how FreeSWITCH handles things in general now. If a module has > an error, the switch just logs and goes on. I?m not really in favour of > this, and suggested at least a ?required? attribute in the modules.conf that > would prevent the switch from loading if the module fails. > > > > The fix is probably to create an attribute you can apply to the plugin > classes that indicate what kind of failure handling you want. For the > assembly, we?d add an attribute with an enumeration like: > > - Default (scan, call ILoadNotificationPlugin, log errors if they > occur) > > - NoLoad (don?t load the assembly) > > - Critical (stop the switch if there?s an exception during > loading) > > > > That?d provide the control you want for loading. We could do something > similar for App/Api plugins. > > > > I want to move ILoadNotificationPlugin from being this ?catch all? thing > that controls the entire assembly to something that can be used to fire up > code; effectively ?OnLoad? and ?OnUnload?. To dynamically control loading, > we?ll probably reflect on the individual plugins looking for attributes or > perhaps some sort of static load function. > > > > How?s that sound? > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers > *Sent:* Thursday, September 10, 2009 12:48 PM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > I'm only concerned with the difference in treatment. > > public class CrashFreeSWITCH : ILoadNotificationPlugin > { > public bool Load() > { > ThreadPool.QueueUserWorkItem((o) => { throw new > NotImplementedException(); }); > return true; > } > } > > Crashes the entire switch, terminating all calls and disconnecting from the > PSTN. > > public class CrashFreeSWITCH : ILoadNotificationPlugin > { > public bool Load() > { > throw new NotImplementedException(); > return true; > } > } > > Logs a message to the console and doesn't load the module, while leaving > the switch operating. > > > > In my experience, exceptions in multi-threaded code: a) happen, b) are hard > to diagnose. Is the best behavior for the environment to crash, providing no > diagnostic information? That's hard in development, and even harder in > production. I suppose 'terminate switch on fault' should be an option, to > allow the operating system to restart the whole process on fault conditions, > but if that is the intended result, shouldn't any fault do the same thing? > What if the billing was happening in my second code block? > > > > Normally, I'd trap the ThreadException and UnhandledExceptions in my > application, so that my code could choose the correct actions to perform > should the application get into an unknown state. This can include > terminating the whole application, but also logging diagnostic information, > trying to save uncommitted data, and sending notifications of the failure. > > > > Is 'crash if it's a thread, but not if it's not' good because it's the way > the module works now, or is it a better design for a reason I'm not > understanding? > > > > On Wed, Sep 9, 2009 at 11:09 PM, Michael Giagnocavo > wrote: > > Well, a segfault in voicemail would do the same thing. > > > > Suppose your plugin runs a thread that does something important, like > billing or so on. That thread fails ? do you really want it to go on? > > > > Anyways, the solution is simple enough, handle your exceptions J. Every > plugin can decide what it wants to do here. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers > *Sent:* Wednesday, September 09, 2009 10:41 PM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > The question is whether the CLR should take down the whole phone server due > to an unhandled exception...definitely the CLR should terminate...but > shouldn't it just log the exception to the console, not crash the core? > > On Wed, Sep 9, 2009 at 6:30 PM, Michael Giagnocavo > wrote: > > That?s by design. If a thread fails, and there?s no handler, then the > application could be in a corrupted state, so the CLR takes down the > process. > > > > I think there is a .NET 1.0 compat switch you can enable in the config if > you like exceptions to be silently ignored J. > > > > -Michael > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers > *Sent:* Wednesday, September 09, 2009 6:39 PM > > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > I have a new thought on the crashes...I'm able to crash FreeSWITCH any time > I like, just by having an exception in a thread. > > > > public class CrashFreeSWITCH : ILoadNotificationPlugin > > { > > public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => { throw new > NotImplementedException(); }); > > return true; > > } > > } > > > > Perhaps Application.ThreadException or AppDomain.UnhandledException need to > be trapped? > > > > On Wed, Sep 9, 2009 at 4:51 PM, Michael Giagnocavo > wrote: > > >Looks like the event object goes straight to pinvokes, so a null result > just crashes? > > > > If it?s null, you should get a NullReferenceException. The C# compiler > should callvirt the property getter and that?ll do a null check. If that > isn?t happening, that?d be an interesting optimization somewhere along the > line. > > > > -Michael > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers > *Sent:* Wednesday, September 09, 2009 3:01 PM > > > *To:* freeswitch-users at lists.freeswitch.org > > *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / > .NET > > > > A new discovery: > > public bool Load() > > { > > ThreadPool.QueueUserWorkItem((o) => > > { > > Log.WriteLine(LogLevel.Notice, "Thread Starting. "); > > EventConsumer con = new EventConsumer("all", ""); > > while (true) > > { > > Event ev = con.pop(0); > > if (ev == null) continue; > > Log.WriteLine(LogLevel.Notice, "Event: " + > ev.serialized_string); > > } > > }); > > return true; > > } > > Does not crash. (Adding the null check prevents crash.) The backgrounded > loop runs fine. Looks like the event object goes straight to pinvokes, so a > null result just crashes? > > > > I like the idea of a 'startup-script' for mod_managed. It would also be > excellent if there was an event or message informing the background code to > terminate nicely when the module reloads. > > > > --Josh > > > > On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk wrote: > > > I think the problem here is that the loader only keeps this method in scope > until completion then it drops the remoted connection. Therefore you should > not use threads in this method. Michael please correct me if I am wrong > here. > > As an example of the failure simply just put a Sleep(10000) call in the > thread and you will see the failure. > > As Michael said this method was only designed to allow the option to opt > out > of being loaded. > > In order to support this perhaps a configuration flag simular to the lua > "startup-script" should be added. > > > > > Here is the error I get with the loop I mentioned. -Josh > [image: Capture.PNG] > > On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo > wrote: > > > Hi, > > > > > > > > Can you please elaborate on the crash you receive when > you > > queue a thread during load? > > > > > > > > Thanks, > > > > Michael > > > > > > -- > View this message in context: > http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/3b433e5b/attachment-0001.html From yj13535428332 at gmail.com Fri Sep 11 01:48:26 2009 From: yj13535428332 at gmail.com (jun yang) Date: Fri, 11 Sep 2009 16:48:26 +0800 Subject: [Freeswitch-users] How to subscribe to custom event in cli? Message-ID: <6536698d0909110148r215b92d7k621764cdd55d2a26@mail.gmail.com> how can i subscribe to custom event in cli. cli: load mod_event_socket say Module mod_event_socket Already Loaded! but i use cli: event plain CHANNEL_CREATE return event: Command not found! cli: api event plain CHANNEL_CREATE return api: Command not found! then what is the correct command? thanks for some hint! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/eb4de5b4/attachment.html From yj13535428332 at gmail.com Fri Sep 11 04:21:24 2009 From: yj13535428332 at gmail.com (jun yang) Date: Fri, 11 Sep 2009 19:21:24 +0800 Subject: [Freeswitch-users] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip Message-ID: <6536698d0909110421l347575ccp64bd5615854f473c@mail.gmail.com> the os have three ip, one public ipv4 with adsl which is dynamic assigned every time, two lan ip in diffrent scope, 192.169.0.2 ,192.168.5.2. when freeswitch start ,it auto bind to the pubic ip, so the lan user cann't connect to freeswitch use lan ip. i have setting but have no effect, freeswitch also auto bind to the public ip. any help is thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/53e7a3d6/attachment.html From jason at jasonjgw.net Fri Sep 11 04:48:24 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 11 Sep 2009 21:48:24 +1000 Subject: [Freeswitch-users] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip In-Reply-To: <6536698d0909110421l347575ccp64bd5615854f473c@mail.gmail.com> References: <6536698d0909110421l347575ccp64bd5615854f473c@mail.gmail.com> Message-ID: <20090911114824.GA10620@jdc.jasonjgw.net> jun yang wrote: > when freeswitch start ,it auto bind to the pubic ip, so the lan user cann't > connect to freeswitch use lan ip. > i have setting > > but have no effect, freeswitch also auto bind to the public ip. > any help is thanks. Set local_ip_v4 in vars.xml to your desired IP address. From yj13535428332 at gmail.com Fri Sep 11 05:08:37 2009 From: yj13535428332 at gmail.com (jun yang) Date: Fri, 11 Sep 2009 20:08:37 +0800 Subject: [Freeswitch-users] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip In-Reply-To: <20090911114824.GA10620@jdc.jasonjgw.net> References: <6536698d0909110421l347575ccp64bd5615854f473c@mail.gmail.com> <20090911114824.GA10620@jdc.jasonjgw.net> Message-ID: <6536698d0909110508v65b45cbau90df8bf4ba7227f5@mail.gmail.com> i add before and it has no effect all the same. is that something wrong. 2009/9/11 Jason White > jun yang wrote: > > when freeswitch start ,it auto bind to the pubic ip, so the lan user > cann't > > connect to freeswitch use lan ip. > > i have setting > > > > but have no effect, freeswitch also auto bind to the public ip. > > any help is thanks. > > Set local_ip_v4 in vars.xml to your desired IP address. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/3aca8f7e/attachment.html From yj13535428332 at gmail.com Fri Sep 11 05:28:58 2009 From: yj13535428332 at gmail.com (jun yang) Date: Fri, 11 Sep 2009 20:28:58 +0800 Subject: [Freeswitch-users] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip In-Reply-To: <6536698d0909110508v65b45cbau90df8bf4ba7227f5@mail.gmail.com> References: <6536698d0909110421l347575ccp64bd5615854f473c@mail.gmail.com> <20090911114824.GA10620@jdc.jasonjgw.net> <6536698d0909110508v65b45cbau90df8bf4ba7227f5@mail.gmail.com> Message-ID: <6536698d0909110528o2a2f852aw489c88c0400d4836@mail.gmail.com> when i set local_ip_v4 to 0.0.0.0 i see the info below: 2009-09-11 20:22:27.15625 [WARNING] sofia.c:2291 Invalid IP 0.0.0.0 replaced with 218.21.105.133 2009-09-11 20:22:27.15625 [WARNING] sofia.c:2300 Invalid IP 0.0.0.0 replaced with 218.21.105.133 2009-09-11 20:22:27.15625 [NOTICE] sofia.c:1509 Adding Alias [0.0.0.0] for profile [internal] 2009/9/11 jun yang > i add > > before > > and it has no effect all the same. > > is that something wrong. > > 2009/9/11 Jason White > > jun yang wrote: >> > when freeswitch start ,it auto bind to the pubic ip, so the lan user >> cann't >> > connect to freeswitch use lan ip. >> > i have setting >> > >> > but have no effect, freeswitch also auto bind to the public ip. >> > any help is thanks. >> >> Set local_ip_v4 in vars.xml to your desired IP address. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/2d4ad568/attachment.html From yj13535428332 at gmail.com Fri Sep 11 05:51:14 2009 From: yj13535428332 at gmail.com (jun yang) Date: Fri, 11 Sep 2009 20:51:14 +0800 Subject: [Freeswitch-users] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip In-Reply-To: <6536698d0909110528o2a2f852aw489c88c0400d4836@mail.gmail.com> References: <6536698d0909110421l347575ccp64bd5615854f473c@mail.gmail.com> <20090911114824.GA10620@jdc.jasonjgw.net> <6536698d0909110508v65b45cbau90df8bf4ba7227f5@mail.gmail.com> <6536698d0909110528o2a2f852aw489c88c0400d4836@mail.gmail.com> Message-ID: <6536698d0909110551o27d76fch6ff55b1e3d97b8d8@mail.gmail.com> i found one solution on web: http://forum.pfsense.org/index.php?topic=18200.0 and past below: Hi all, I want to share my configuration of Freeswitch with Dynamic IP WAN. I obviously have to thank Mcrane not only for his great job porting FS to pfSense but also for the huge amount of time spent with me, my configuration and my poor knowdlege about FS (that caused a lot of headhaches to him). When you have a DHCP or PPPOE connection the annoying issue is your are left without PBX when the internet connection is down. It happens very often to me, I'm in countryside and the quality of DSL is terrible. I have PPPoA half-bridge so when the connection goes down my WAN IP di 0.0.0.0. With this setup you will always be able to use internal extensions, I also have a Sipura-3102 with PSTN configured as gateway so I can always call outside also when WAN is down. You'll be able to register local extensions to lan profile and external extensions to internal profiles and they can call each others. Optionally you can also enable SSLv23 for encrypted calls (but I'm not sure this way only SIP messaging or voice data too is encrypted). *1) CONFIGURE DYNAMIC DNS* Register to a dynamic dns and get an hostname, I'll use a dummy " dsl.homeip.net" for this guide. I'm using DynDns, if you choose another provider be sure it supports "Wildcards". Configure the ddns name in Services->Dynamic DNS and be sure to enable "Wildcards". *2) CONFIGURE DNS FORWARDER* I will use as example for this guide the pfSense LAN IP 192.168.0.1. Check "Enable DNS forwarder". Your phones must use ONLY your pfSense LAN IP address as DNS server, check it! Below "You may enter records that override the results from the forwarders below." add an hostname this way, BE SURE to replace with your correct ddns and LAN IP: Host: dsl Domain: homeip.net IP Address: 192.168.0.1 Save configuration and check with your PC (with pfSense configured as primary DNS), ping to dsl.homeip.net must resolve to LAN IP, ping to fs.dsl.homeip.net must resolve to your WAN IP. *3) CONFIGURE INTERNAL PROFILE* Extensions from outside that register to WAN IP must use TCP protocol. Always use your ddns, "dsl.homeip.net", for registration domain/SIP host. Go to Services->Freeswitch->Profiles and click edit for internal.xml. Uncomment and/or set with this values: *4) CONFIGURE LAN PROFILE* Extensions from inside that register to LAN IP must use UDP protocol. Always use your ddns, "dsl.homeip.net", for registration domain/SIP host. Go to Services->Freeswitch->Profiles and click edit for lan.xml. Be sure to change 192.168.0.1 with you LAN IP! Uncomment and/or set with this values: *5) CONFIGURE VARS* Go to Services->Freeswitch->Vars. Be sure to prepend "fs." or anything else you like to ddns name in external_rtp_* set values! It's needed because pfsense locally will resolve you registered ddns to LAN IP address. Uncomment and/or set with this values: *6) TLS ENCRYPTION* ONLY IF you want TLS/SSLv23 encryption also set in vars.xml: Then go to SSH console and input the commands (respond Y to questions and change to your ddns name "dsl.homeip.net"): cd /usr/local/freeswitch/bin/ ./gentls_cert setup ./gentls_cert create -cn dsl.homeip.net -alt DNS:dsl.homeip.net Your master certificate is in /usr/local/freeswitch/conf/ssl/CA/ with name cafile.pem Install in your SIP phones or if you use a Windows softphone, download it and rename to "cafile.crt". Double click to add it to certificate store, default options when asked. I verified it works with Windows Vista and Eyebeam Softphone. (If you use Eyebeam, it can't receive encrypted calls, under Security tab check only preference for encryption calls) *7) RESTART FS* Don't only issue a reloadxml, restart the FS serice. Maybe something is missing (my setup is working and I'm not 100% sure that's all what you need from default config), let me know and I'll update the thread. Cheers, Mannix 2009/9/11 jun yang > when i set local_ip_v4 to 0.0.0.0 i see the info below: > 2009-09-11 20:22:27.15625 [WARNING] sofia.c:2291 Invalid IP 0.0.0.0 > replaced with 218.21.105.133 > 2009-09-11 20:22:27.15625 [WARNING] sofia.c:2300 Invalid IP 0.0.0.0 > replaced with 218.21.105.133 > 2009-09-11 20:22:27.15625 [NOTICE] sofia.c:1509 Adding Alias [0.0.0.0] for > profile [internal] > > 2009/9/11 jun yang > > i add >> >> before >> >> and it has no effect all the same. >> >> is that something wrong. >> >> 2009/9/11 Jason White >> >> jun yang wrote: >>> > when freeswitch start ,it auto bind to the pubic ip, so the lan user >>> cann't >>> > connect to freeswitch use lan ip. >>> > i have setting >>> > >>> > but have no effect, freeswitch also auto bind to the public ip. >>> > any help is thanks. >>> >>> Set local_ip_v4 in vars.xml to your desired IP address. >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/cf172932/attachment-0001.html From yj13535428332 at gmail.com Fri Sep 11 05:55:47 2009 From: yj13535428332 at gmail.com (jun yang) Date: Fri, 11 Sep 2009 20:55:47 +0800 Subject: [Freeswitch-users] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip In-Reply-To: <6536698d0909110421l347575ccp64bd5615854f473c@mail.gmail.com> References: <6536698d0909110421l347575ccp64bd5615854f473c@mail.gmail.com> Message-ID: <6536698d0909110555j5c498030o2589f3b36a867e60@mail.gmail.com> i also found that: 2009/7/17 Raul Fragoso >: >* You can not do that with a single profile. Each profile is bound to only *>* one local IP, so if you need to bind to more than one you will have to *>* create a new profile and set the specific sip-ip/rtp-ip params for them. *>* but cann't understand how to do..* 2009/9/11 jun yang > the os have three ip, one public ipv4 with adsl which is dynamic assigned > every time, two lan ip in diffrent scope, 192.169.0.2 ,192.168.5.2. > when freeswitch start ,it auto bind to the pubic ip, so the lan user cann't > connect to freeswitch use lan ip. > i have setting > > but have no effect, freeswitch also auto bind to the public ip. > any help is thanks. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/c2299e6e/attachment.html From woodydickson at gmail.com Fri Sep 11 06:06:40 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Fri, 11 Sep 2009 21:06:40 +0800 Subject: [Freeswitch-users] possible sofia_contact bug Message-ID: Hi, I am having a strange problem here. sofia status shows that the user is registered, but sofia_contact says the user is not registered. Does anyone know why this is happening? freeswitch at localhost.localdomain> sofia status profile internal reg 180004 API CALL [sofia(status profile internal reg 180004)] output: Registrations: ================================================================================================= Call-ID: 530339592782-1484696326482 at 192.168.1.163 User: 180004 at 192.168.1.102 Contact: 180004 Agent: Voip Phone 1.0 Status: Registered(UDP)(unknown) EXP(2009-09-12 04:59:36) Host: localhost.localdomain IP: 192.168.1.163 Port: 9000 Auth-User: 180004 Auth-Realm: 192.168.1.102 ================================================================================================= freeswitch at localhost.localdomain> sofia_contact 180004 at 192.168.1.102 API CALL [sofia_contact(180004 at 192.168.1.102)] output: error/user_not_registered freeswitch at localhost.localdomain> freeswitch at localhost.localdomain> sofia_contact user/180004 API CALL [sofia_contact(user/180004)] output: error/facility_not_subscribed -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/131a6754/attachment.html From brian at freeswitch.org Fri Sep 11 06:19:55 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Sep 2009 08:19:55 -0500 Subject: [Freeswitch-users] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip In-Reply-To: <6536698d0909110555j5c498030o2589f3b36a867e60@mail.gmail.com> References: <6536698d0909110421l347575ccp64bd5615854f473c@mail.gmail.com> <6536698d0909110555j5c498030o2589f3b36a867e60@mail.gmail.com> Message-ID: <8549660C-1DEF-4017-90CE-43FA51E6C929@freeswitch.org> You can NOT bind to 0.0.0.0 you can however use ${local_ip_v4} and if the IP changes sofia will bounce the profile and update the IP. /b On Sep 11, 2009, at 7:55 AM, jun yang wrote: > i also found that: > 2009/7/17 Raul Fragoso : > > You can not do that with a single profile. Each profile is bound > to only > > > one local IP, so if you need to bind to more than one you will > have to > > create a new profile and set the specific sip-ip/rtp-ip params for > them. > > > but cann't understand how to do.. > > > 2009/9/11 jun yang > the os have three ip, one public ipv4 with adsl which is dynamic > assigned every time, two lan ip in diffrent scope, 192.169.0.2 , > 192.168.5.2. > when freeswitch start ,it auto bind to the pubic ip, so the lan user > cann't connect to freeswitch use lan ip. > i have setting > > but have no effect, freeswitch also auto bind to the public ip. > any help is thanks. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/59f4b217/attachment.html From brian at freeswitch.org Fri Sep 11 06:20:19 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Sep 2009 08:20:19 -0500 Subject: [Freeswitch-users] memory leak In-Reply-To: References: <191c3a030909020842j61369a45va923080ea0a88b40@mail.gmail.com> <191c3a030909040757y93c105bs8939559f90142b9e@mail.gmail.com> Message-ID: <51E4B595-4D57-46B3-ABB8-E9EC0A439BF2@freeswitch.org> Next Bug? Huh? :P /b On Sep 11, 2009, at 2:32 AM, Benedikt Fraunhofer wrote: > > Thx again and looking forward to the next bug :) From brian at freeswitch.org Fri Sep 11 06:21:29 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Sep 2009 08:21:29 -0500 Subject: [Freeswitch-users] How to subscribe to custom event in cli? In-Reply-To: <6536698d0909110148r215b92d7k621764cdd55d2a26@mail.gmail.com> References: <6536698d0909110148r215b92d7k621764cdd55d2a26@mail.gmail.com> Message-ID: You need to telnet to the socket or use fs_cli... example... telnet 0 8021 auth ClueCon events all plain (or what ever commands you wish to run) /b On Sep 11, 2009, at 3:48 AM, jun yang wrote: > how can i subscribe to custom event in cli. > cli: load mod_event_socket > say Module mod_event_socket Already Loaded! > but i use > cli: event plain CHANNEL_CREATE > return event: Command not found! > cli: api event plain CHANNEL_CREATE > return api: Command not found! > > then what is the correct command? > thanks for some hint! From brian at freeswitch.org Fri Sep 11 06:41:00 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Sep 2009 08:41:00 -0500 Subject: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? In-Reply-To: <4AA9FBA5.5090403@kounitskiy.com> References: <4aa965af.161bf30a.61f8.7c7d@mx.google.com> <4AA9FBA5.5090403@kounitskiy.com> Message-ID: <959258A1-5948-4F8F-900C-E2BA3FC68977@freeswitch.org> Thats normal too. /b On Sep 11, 2009, at 2:26 AM, Anatoliy Kounitskiy wrote: > It's normal to have to two records for a call - Start and Stop > message. > > From what i see - you have one start and stop for each leg of the > call. > > Regards, > AK From brian at freeswitch.org Fri Sep 11 06:42:59 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Sep 2009 08:42:59 -0500 Subject: [Freeswitch-users] Implementing h extension in FS In-Reply-To: References: Message-ID: No you should never be doing your billing inline like this. You should be doing this externally of your application not inside your dialplan. /b On Sep 10, 2009, at 11:40 PM, Ahmed Munir wrote: > Thanks for reply, well actually I'm doing billing after call hangup. > If h extension is interupts I'm sending to it to addcdr context > which interupts perl script for billing purpose. As I'm listing down > below asterisk configuration; From brian at freeswitch.org Fri Sep 11 07:01:03 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Sep 2009 09:01:03 -0500 Subject: [Freeswitch-users] Friday Meeting at 11AM CST Message-ID: <1BBA7816-D38F-4F6D-A0DD-DCA68069B419@freeswitch.org> http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11 Here is the agenda please review and add to it anything you think we should cover. Thanks, Brian From anthony.minessale at gmail.com Fri Sep 11 08:07:18 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Sep 2009 10:07:18 -0500 Subject: [Freeswitch-users] Implementing h extension in FS In-Reply-To: References: Message-ID: <191c3a030909110807x1130cc4ak63bfd2da98bb2b5a@mail.gmail.com> FreeSWITCH is driven by a state machine and execute and hangup are opposing states so once you change to hangup state that is the end of executing extensions. asterisk has 4 special extensions s h i and t we don't support any of them because our dialplan concept and paradigm is completely different. There is a feature in FS called api_hangup_hook which is a variable you can set to a desired script to execute when the call hangs up. you should be able to find it on the wiki On Thu, Sep 10, 2009 at 11:40 PM, Ahmed Munir wrote: > Thanks for reply, well actually I'm doing billing after call hangup. If h > extension is interupts I'm sending to it to addcdr context which interupts > perl script for billing purpose. As I'm listing down below asterisk > configuration; > > h => > { > NOOP("Call Completed with Carrier ${CARRIER}"); > goto add_cdr|h|1; > }; > > context add_cdr > { > _X. => > { > Hangup(); > }; > h => > { > Set(CALL_END_TIME=${EPOCH}); > //&print_variables(); > NOOP("Call Ended: Card:${CARDNUM} Destination:${CALLEDNUM} > Caller-ID:${CALLERID(num)}"); > if (${DIALEXECUTED}=YES) > { > NOOP("Dial-Status:${DIALSTATUS}"); > }else > { > NOOP("Dial was not Executed"); > }; > DeadAGI(/vopium/agi/billing.pl); > NOOP(); > }; > > }; > > Kindly advice me how I pass/translate h extension in FS in this situation > i.e. or there is > other way around??? > ------------------------------ > *From: *Michael Collins > *Reply-To: * > *Date: *Thu, 10 Sep 2009 00:55:02 -0700 > *To: * > *Subject: *Re: [Freeswitch-users] Implementing h extension in FS > > > > On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir > wrote: > > HI, > > I'm newbie in FS, I want to know how to implement h extension of asterisk > to FS. As I listed down below; > > h => > { > NOOP("Call Completed with Carrier ${CARRIER}"); > goto add_cdr|h|1; > }; > > My other question is, which application/function/class is use in mod_perl > to check the channel status? i.e. busy, answer,hangup,ringing,etc. > > > Kindly advice me soon. > > -- > Regards, > > Ahmed Munir > > > It depends on what you are trying to accomplish, but the closest thing > you'll find in FS to the 'h' extension is the channel variable > api_hangup_hook which lets you launch an API at the end of the call. It > sounds like you are working on CDR processing. Please tell us more about > your application and we'll do our best to offer advice. > -MC > > -- > Regards, > > Ahmed Munir > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/1f784610/attachment-0001.html From anthony.minessale at gmail.com Fri Sep 11 08:10:49 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Sep 2009 10:10:49 -0500 Subject: [Freeswitch-users] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip In-Reply-To: <8549660C-1DEF-4017-90CE-43FA51E6C929@freeswitch.org> References: <6536698d0909110421l347575ccp64bd5615854f473c@mail.gmail.com> <6536698d0909110555j5c498030o2589f3b36a867e60@mail.gmail.com> <8549660C-1DEF-4017-90CE-43FA51E6C929@freeswitch.org> Message-ID: <191c3a030909110810j49023634ta7dbf4e619e3c240@mail.gmail.com> sip in general cannot properly support binding to 0.0.0.0 for a UAS, there is no easy way for the sip stack to know which traffic is for which host and all of the outbound traffic will appear to go out a single interface when no specific binding is made. running each ip on it's own profile is the correct way to do multi ip configurations. On Fri, Sep 11, 2009 at 8:19 AM, Brian West wrote: > You can NOT bind to 0.0.0.0 you can however use ${local_ip_v4} and if the > IP changes sofia will bounce the profile and update the IP. > /b > > On Sep 11, 2009, at 7:55 AM, jun yang wrote: > > i also found that: > > 2009/7/17 Raul Fragoso >: > >* You can not do that with a single profile. Each profile is bound to only > *>* one local IP, so if you need to bind to more than one you will have to > *>* create a new profile and set the specific sip-ip/rtp-ip params for them. > *>* > but cann't understand how to do..* > > > > 2009/9/11 jun yang > >> the os have three ip, one public ipv4 with adsl which is dynamic assigned >> every time, two lan ip in diffrent scope, 192.169.0.2 ,192.168.5.2. >> when freeswitch start ,it auto bind to the pubic ip, so the lan user >> cann't connect to freeswitch use lan ip. >> i have setting >> >> but have no effect, freeswitch also auto bind to the public ip. >> any help is thanks. >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/d13fbbbc/attachment.html From anthony.minessale at gmail.com Fri Sep 11 08:11:57 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Sep 2009 10:11:57 -0500 Subject: [Freeswitch-users] How to subscribe to custom event in cli? In-Reply-To: References: <6536698d0909110148r215b92d7k621764cdd55d2a26@mail.gmail.com> Message-ID: <191c3a030909110811n7904ca7fvec8c55ec062b5d7@mail.gmail.com> or from fs_cli /events plain all On Fri, Sep 11, 2009 at 8:21 AM, Brian West wrote: > You need to telnet to the socket or use fs_cli... example... > > telnet 0 8021 > auth ClueCon > events all plain (or what ever commands you wish to run) > > /b > > > On Sep 11, 2009, at 3:48 AM, jun yang wrote: > > > how can i subscribe to custom event in cli. > > cli: load mod_event_socket > > say Module mod_event_socket Already Loaded! > > but i use > > cli: event plain CHANNEL_CREATE > > return event: Command not found! > > cli: api event plain CHANNEL_CREATE > > return api: Command not found! > > > > then what is the correct command? > > thanks for some hint! > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/954dc8a1/attachment.html From anthony.minessale at gmail.com Fri Sep 11 08:31:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 11 Sep 2009 10:31:17 -0500 Subject: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? In-Reply-To: <4AA9FBA5.5090403@kounitskiy.com> References: <4aa965af.161bf30a.61f8.7c7d@mx.google.com> <4AA9FBA5.5090403@kounitskiy.com> Message-ID: <191c3a030909110831o4e0a4844obf3c339d58f5358a@mail.gmail.com> set the variable process_cdr=false on that a_leg first thing in your dialplan On Fri, Sep 11, 2009 at 2:26 AM, Anatoliy Kounitskiy < anatoliy at kounitskiy.com> wrote: > It's normal to have to two records for a call - Start and Stop message. > > From what i see - you have one start and stop for each leg of the call. > > Regards, > AK > > email lists wrote: > > > > Hello, > > > > > > > > Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate > > RADIUS messages being generated for individual calls (sample messages > > for one call below). Looking at the "Acct-Unique-Session-Id" and > > "Acct-Session-Id" fields, it would appear that perhaps each call leg > > results in a pair of start/stop RADIUS messages; is this the expected > > behavior? If so, is there a way to disable RADIUS messaging for what > > I presume is the "ingress" or A leg of the call? > > > > > > > > Any leads would be appreciated. > > > > > > > > Thanks in advance. > > > > > > > > Vladimir > > > > > > > > Thu Sep 10 10:37:25 2009 > > > > Acct-Status-Type = Start > > > > Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" > > > > User-Name = "8135793256" > > > > Freeswitch-Src = "8135793256" > > > > Freeswitch-CLID = "sipp" > > > > Freeswitch-Dst = "14043297226" > > > > Freeswitch-Dialplan = "XML" > > > > Framed-IP-Address = 50.46.50.55 > > > > Freeswitch-Context = "public" > > > > Freeswitch-Source = "mod_sofia" > > > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" > > > > NAS-Port = 0 > > > > Acct-Delay-Time = 0 > > > > NAS-IP-Address = 1.1.1.1 > > > > Acct-Unique-Session-Id = "097c8472ff7bcec7" > > > > Timestamp = 1252604245 > > > > Request-Authenticator = Verified > > > > > > > > Thu Sep 10 10:37:25 2009 > > > > Acct-Status-Type = Start > > > > Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" > > > > User-Name = "8135793256" > > > > Freeswitch-Src = "8135793256" > > > > Freeswitch-CLID = "sipp" > > > > Freeswitch-Dst = "14043297226 at x.x.x.x" > > > > Freeswitch-Dialplan = "XML" > > > > Framed-IP-Address = 50.46.50.55 > > > > Freeswitch-Context = "public" > > > > Freeswitch-Source = "mod_sofia" > > > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" > > > > NAS-Port = 0 > > > > Acct-Delay-Time = 0 > > > > NAS-IP-Address = 1.1.1.1 > > > > Acct-Unique-Session-Id = "53f729e173e8c0a9" > > > > Timestamp = 1252604245 > > > > Request-Authenticator = Verified > > > > > > > > Thu Sep 10 10:37:57 2009 > > > > Acct-Status-Type = Stop > > > > Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" > > > > Freeswitch-Hangupcause = Normal-Unspecified > > > > User-Name = "8135793256" > > > > Freeswitch-Src = "8135793256" > > > > Freeswitch-CLID = "sipp" > > > > Freeswitch-Dst = "14043297226" > > > > Freeswitch-Dialplan = "XML" > > > > Framed-IP-Address = 50.46.50.55 > > > > Freeswitch-Context = "public" > > > > Freeswitch-Source = "mod_sofia" > > > > Freeswitch-Lastapp = "bridge" > > > > Freeswitch-Billusec = 32029926 > > > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" > > > > Freeswitch-Callanswerdate = "2009-09-10T10:22:00.319197-0700" > > > > Freeswitch-Callenddate = "2009-09-10T10:22:32.349123-0700" > > > > Acct-Session-Time = 32 > > > > NAS-Port = 0 > > > > Acct-Delay-Time = 0 > > > > NAS-IP-Address = 1.1.1.1 > > > > Acct-Unique-Session-Id = "097c8472ff7bcec7" > > > > Timestamp = 1252604277 > > > > Request-Authenticator = Verified > > > > > > > > Thu Sep 10 10:38:02 2009 > > > > Acct-Status-Type = Stop > > > > Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" > > > > Freeswitch-Hangupcause = Normal-Clearing > > > > User-Name = "8135793256" > > > > Freeswitch-Src = "8135793256" > > > > Freeswitch-CLID = "sipp" > > > > Freeswitch-Dst = "14043297226 at x.x.x.x" > > > > Freeswitch-Dialplan = "XML" > > > > Framed-IP-Address = 50.46.50.55 > > > > Freeswitch-Context = "public" > > > > Freeswitch-Source = "mod_sofia" > > > > Freeswitch-Billusec = 32049973 > > > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" > > > > Freeswitch-Callanswerdate = "2009-09-10T10:22:00.289136-0700" > > > > Freeswitch-Callenddate = "2009-09-10T10:22:32.339109-0700" > > > > Acct-Session-Time = 32 > > > > NAS-Port = 0 > > > > Acct-Delay-Time = 0 > > > > NAS-IP-Address = 1.1.1.1 > > > > Acct-Unique-Session-Id = "53f729e173e8c0a9" > > > > Timestamp = 1252604282 > > > > Request-Authenticator = Verified** > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/0e796a5f/attachment-0001.html From gmaruzz at celliax.org Fri Sep 11 08:47:48 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 11 Sep 2009 17:47:48 +0200 Subject: [Freeswitch-users] Friday Meeting at 11AM CST In-Reply-To: <1BBA7816-D38F-4F6D-A0DD-DCA68069B419@freeswitch.org> References: <1BBA7816-D38F-4F6D-A0DD-DCA68069B419@freeswitch.org> Message-ID: <7b197bef0909110847l3e5006f9ye41811066fc6be17@mail.gmail.com> On Fri, Sep 11, 2009 at 4:01 PM, Brian West wrote: > http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_11 > > Here is the agenda please review and add to it anything you think we > should cover. This time too, you all can follow the conference calling Skype the skypeuser "skypiax5", then press "1" on the Skype dialpad (max 20 concurrent users). -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From diego.viola at gmail.com Fri Sep 11 08:50:34 2009 From: diego.viola at gmail.com (Diego Viola) Date: Fri, 11 Sep 2009 15:50:34 +0000 Subject: [Freeswitch-users] Implementing h extension in FS In-Reply-To: References: Message-ID: <86a32abc0909110850n751ed639s22aeed8d2b516a67@mail.gmail.com> You could create a daemon like this that listens for the CHANNEL_HANGUP_COMPLETE event and send your CDR to the db. http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ruby/callcard/cdr.rb Then do the billing stuff outside FreeSWITCH or use mod_nibblebill. I suggest also that you enable mod_xml_cdr or mod_cdr_csv so you always have a copy of the CDR on disk in case if something fails (like the db). Diego On Fri, Sep 11, 2009 at 4:40 AM, Ahmed Munir wrote: > Thanks for reply, well actually I'm doing billing after call hangup. If h > extension is interupts I'm sending to it to addcdr context which interupts > perl script for billing purpose. As I'm listing down below asterisk > configuration; > > h => > { > NOOP("Call Completed with Carrier ${CARRIER}"); > goto add_cdr|h|1; > }; > > context add_cdr > { > _X. => > { > Hangup(); > }; > h => > { > Set(CALL_END_TIME=${EPOCH}); > //&print_variables(); > NOOP("Call Ended: Card:${CARDNUM} Destination:${CALLEDNUM} > Caller-ID:${CALLERID(num)}"); > if (${DIALEXECUTED}=YES) > { > NOOP("Dial-Status:${DIALSTATUS}"); > }else > { > NOOP("Dial was not Executed"); > }; > DeadAGI(/vopium/agi/billing.pl); > NOOP(); > }; > > }; > > Kindly advice me how I pass/translate h extension in FS in this situation > i.e. or there is > other way around??? > ------------------------------ > *From: *Michael Collins > *Reply-To: * > *Date: *Thu, 10 Sep 2009 00:55:02 -0700 > *To: * > *Subject: *Re: [Freeswitch-users] Implementing h extension in FS > > > > On Wed, Sep 9, 2009 at 10:16 PM, Ahmed Munir > wrote: > > HI, > > I'm newbie in FS, I want to know how to implement h extension of asterisk > to FS. As I listed down below; > > h => > { > NOOP("Call Completed with Carrier ${CARRIER}"); > goto add_cdr|h|1; > }; > > My other question is, which application/function/class is use in mod_perl > to check the channel status? i.e. busy, answer,hangup,ringing,etc. > > > Kindly advice me soon. > > -- > Regards, > > Ahmed Munir > > > It depends on what you are trying to accomplish, but the closest thing > you'll find in FS to the 'h' extension is the channel variable > api_hangup_hook which lets you launch an API at the end of the call. It > sounds like you are working on CDR processing. Please tell us more about > your application and we'll do our best to offer advice. > -MC > > -- > Regards, > > Ahmed Munir > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/e14e3628/attachment.html From msc at freeswitch.org Fri Sep 11 09:01:46 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 Sep 2009 09:01:46 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Starting, Please Call In! Message-ID: <87f2f3b90909110901n166e8d9aq9639dbba20085926@mail.gmail.com> FYI, the conference is starting. Please join us! sip:888 at conference.freeswitch.org 213-799-1400 -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/d325921f/attachment.html From pjintheusa at gmail.com Fri Sep 11 09:02:36 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Fri, 11 Sep 2009 12:02:36 -0400 Subject: [Freeswitch-users] uuid_exists - does it still exist? In-Reply-To: <87f2f3b90909101501k1d459ddcl15b29018f2d46521@mail.gmail.com> References: <367751820909031255l1c6947c6t1bf577c77551f9b3@mail.gmail.com> <87f2f3b90909031456s76a12874xe0a6ebfa26babb10@mail.gmail.com> <367751820909101424h787e7757vd67649562c8d07f8@mail.gmail.com> <87f2f3b90909101501k1d459ddcl15b29018f2d46521@mail.gmail.com> Message-ID: <367751820909110902i2d943f0bi9e4b57ffc6bc4f4e@mail.gmail.com> Yep - I built out a very simple test: string uuid = "12345678-1234-4321-123456789012"; apiResult = fsApi.Execute("originate", string.Format("{{ignore_early_media=false,absolute_codec_string='PCMU'}}[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2} &park", uuid, "2129996599", "6093693828")); Log.WriteLine(LogLevel.Alert, "Originate returns: {0}", apiResult); Session.Execute("park", ""); relevant log details: 2009-09-11 11:28:22.750000 [ALERT] switch_cpp.cpp:1130 Originate returns: +OK 12345678-1234-4321-123456789012 uuid_getvar 12345678-1234-4321-123456789012 uuid RETURNS: 3e1ef4ee-8e39-b841-a9a0-6ac997b56275 uuid _kill 3e1ef4ee-8e39-b841-a9a0-6ac997b56275 -ERR No Such Channel However: uuid _kill 12345678-1234-4321-123456789012 RETURNS +OK and drops the call. I repeated the test and attached the screenshot. Be interested to know where that 3e1ef4ee-8e39-b841-a9a0-6ac997b56275 comes from. All academic - just for interest and better understanding. On Thu, Sep 10, 2009 at 6:01 PM, Michael Collins wrote: > As a sanity test, can you bring up a bridged call and run this test > directly from the command line and compare the results to what's in your > script? > -MC > > > On Thu, Sep 10, 2009 at 2:24 PM, Phillip Jones wrote: > >> Strangely - the uuid_getvar uuid workaround >> does not work for me. >> >> This is the result of: >> >> apiResult = fsApi.ExecuteString(string.Format("uuid_getvar {0} uuid", >> call.Uuid)); >> Log.WriteLine(LogLevel.Alert, "RESULT: uuid_getvar {0} is: {1}", >> call.Uuid, apiResult); >> >> returns a different uuid????? >> >> 2009-09-10 17:12:26.953125 [ALERT] switch_cpp.cpp:1130 RESULT: uuid_getvar >> 54dd24be-b0da-684f-acee-38c7530b4c2b is: >> 1a0e83db-240c-ac4e-ae45-bf5d5b46f5c3 >> >> the passed uuid is vaild however: >> >> 2009-09-10 17:12:26.953125 [ALERT] switch_cpp.cpp:1130 RESULT: uuid_kill >> 54dd24be-b0da-684f-acee-38c7530b4c2b is: +OK >> >> Can a call leg have two uuids?? >> >> >> On Thu, Sep 3, 2009 at 5:56 PM, Michael Collins wrote: >> >>> >>> >>> On Thu, Sep 3, 2009 at 2:23 PM, Benedikt Fraunhofer < >>> fraunhofer.lists.freeswitch-001 at traced.net> wrote: >>> >>>> Hi, >>>> >>>> >>>> > Usage: uuid_exists >>>> > However when I call via an API call I get: >>>> > INVALID COMMAND! >>>> > I also don't see it in MOD_COMMAND.C >>>> >>>> As a workaround or if your are unable to upgrade, you can use >>>> "uuid_getvar [some_uuid] thisVariableDoesNotExist" >>>> ("thisVariableDoesNotExist" is any variable you can think of, a valid >>>> one or literally 'thisVariableDoesNotExist' :) >>>> You'll either get an error that this channel does not exist any longer >>>> or "undef" for the channel variable. >>>> The api will return "+OK" in case the channel still exists, and >>>> "ERROR" in case it does not. >>>> >>>> Beni. >>>> >>>> >>> You could even do this: >>> uuid_getvar uuid >>> >>> If it exists then the return will be the uuid. :) >>> >>> Although I must say I recommend this instead: >>> make current >>> >>> :) >>> -MC >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/7bb0123a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: fs-test.jpg Type: image/jpeg Size: 104449 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/7bb0123a/attachment-0001.jpg From gmaruzz at celliax.org Fri Sep 11 09:12:34 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 11 Sep 2009 18:12:34 +0200 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Starting, Please Call In! In-Reply-To: <87f2f3b90909110901n166e8d9aq9639dbba20085926@mail.gmail.com> References: <87f2f3b90909110901n166e8d9aq9639dbba20085926@mail.gmail.com> Message-ID: <7b197bef0909110912w10a65be3vcb9aa7bc6023c2ad@mail.gmail.com> On Fri, Sep 11, 2009 at 6:01 PM, Michael Collins wrote: > FYI, the conference is starting. Please join us! > sip:888 at conference.freeswitch.org > 213-799-1400 This time too, you all can follow the conference calling Skype the skypeuser "skypiax5", then press "1" on the Skype dialpad (max 20 concurrent users). -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From dome at tel.co.th Fri Sep 11 09:22:20 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Fri, 11 Sep 2009 23:22:20 +0700 Subject: [Freeswitch-users] Music Background Message-ID: <8ccbff060909110922p41d9cb91raa39605e080ea9b7@mail.gmail.com> Dear Sir, Is posible to play music for background when call connect ? Example when i call my wife some time i need romantic song :) BG Dome C. From mike at jerris.com Fri Sep 11 09:23:45 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Sep 2009 12:23:45 -0400 Subject: [Freeswitch-users] mod_xml_curl.c Oversized file detected [1056100 bytes] In-Reply-To: <4AA0E320.8000708@gmx.net> References: <4A9FFD64.2000701@gmx.net> <191c3a030909031130n453786ccyd71ee0b9f92a3955@mail.gmail.com> <4AA0E320.8000708@gmx.net> Message-ID: <373E5B74-E62C-46A4-B2B9-414B43742CCF@jerris.com> if you want you could contribute a patch to make that a config option (of course defaulting to the current value). Mike On Sep 4, 2009, at 5:51 AM, Peter P GMX wrote: > Thanks Anthony, > > that did the trick. > > Best regards > Peter > > Anthony Minessale schrieb: >> you can edit mod_xml_curl.c line 64 >> and increase XML_CURL_MAX_BYTES >> >> >> On Thu, Sep 3, 2009 at 12:31 PM, Peter P GMX > > wrote: >> >> Hello, >> >> in a B2BUA scenario we have 2000 defined gateways (defined but not >> registered yet). >> When reloading mod_sofia Freeswitch complains about the XML-Curl >> File >> size > 1MB and deactivates all gateways: >> mod_xml_curl.c:121 Oversized file detected [1056100 bytes] >> >> Is there any way to overcome this? Currently we have 2000 gateways >> defined. Finally we will have about 10.000. And we will not be >> able to >> reduce the file size below 1 MB. It will become ~ 2-3 MB maybe. >> >> Best Regards >> Peter From mike at jerris.com Fri Sep 11 09:27:25 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Sep 2009 12:27:25 -0400 Subject: [Freeswitch-users] mod_dingaling: dl_login command syntax In-Reply-To: References: Message-ID: What errors do you get? Mike On Sep 4, 2009, at 8:35 AM, Muhammad Shahzad wrote: > Hi, > > i am have FS SVN revision 14760, i am trying to use mod_xml_curl > against mod_dingaling. When i call xml_curl url in browser i get > mod_dingaling configuration correctly, also when i do reload > mod_dingaling it fetches its configuration from xml_curl correctly. > BUT when i try to use dl_login command to login a jingle profile it > does not work. I have tried both syntax, > > Syntax 1: > ======= > dl_login profile=abcd > > Where abcd is a valid jingle profile fetch-able from xml_curl. > > Syntax 2: > ======= > dl_login name=abcd;login=XXX at gmail.com/ > talk;pass=YYY;dialplan=XML;context=public;rtp- > ip=auto;sasl=plain;tls=true;exten=1001 > > All these values are correct and work if i reload mod_dingaling but > they don't work with dl_login, and give following output. > > USAGE: Existing Profile: > dl_login profile= > Dynamic Profile: > dl_login var1=val1;var2=val2;varN=valN > > I don't think xml_curl has any role in this syntax. > > Can you please correct me if i am doing something wrong in here or > is it a bug in mod_dingaling. > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/3501cc2d/attachment.html From mike at jerris.com Fri Sep 11 09:30:46 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Sep 2009 12:30:46 -0400 Subject: [Freeswitch-users] Run a command on event In-Reply-To: <960738410909040732o3f08ec28x6d1869aa6f4649b9@mail.gmail.com> References: <960738410909020727q3286bce9q2ff7d4edfcf6e025@mail.gmail.com> <86a32abc0909020746n1250604ds3ebaf3a3d7abdd21@mail.gmail.com> <960738410909020847u2562a1bei6cedba26e5b06cb6@mail.gmail.com> <88D1D05C-B36D-4E7B-9DFC-58688DFD3467@avgs.ca> <960738410909040732o3f08ec28x6d1869aa6f4649b9@mail.gmail.com> Message-ID: <9AC22BBB-F25D-4A43-8D0C-B6203980B006@jerris.com> You can do it in perl or lua using a startup script that creates an event listener. Mike On Sep 4, 2009, at 10:32 AM, Mathieu Parent wrote: > Hi > > On Thu, Sep 3, 2009 at 6:52 AM, Mathieu Rene > wrote: >> See http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules#Subscribing_to_events >> > > Thanks. > > I have tried this method without success and finally replaced the > voicemail section in dialplan by a spidermonkey script with > session.setHangupHook(). Test passed! > > Mathieu Parent From mike at jerris.com Fri Sep 11 09:35:37 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Sep 2009 12:35:37 -0400 Subject: [Freeswitch-users] XML Dial Plan vs Language Modules In-Reply-To: <5070fcbd0909040737v6d39a234j62d6e1610a001b47@mail.gmail.com> References: <87f2f3b90909040006s15c1f733lff93b4549dee6da7@mail.gmail.com> <5070fcbd0909040737v6d39a234j62d6e1610a001b47@mail.gmail.com> Message-ID: <96A8B959-AD3E-42ED-BCAB-ADECE249B3CD@jerris.com> generally it keeps the overhead of running the script around during the whole call. Mike On Sep 4, 2009, at 10:37 AM, Shameem Shiek wrote: > Hi Michael, > > Why is it not recommended to do the brdge app right in the script? > The reason I ask this, I did have lot of trouble using Park/Fifo app > in the script and the whole thing started working after I did the > UUID transfer and have the things I wanted executed as part of the > Dial plan. > > Also, How many concurrent sessions can one support in ESL using > Python/Ruby compared to using Lua? > > Thanks. > > On Fri, Sep 4, 2009 at 3:06 AM, Michael Collins > wrote: > > > On Thu, Sep 3, 2009 at 10:59 PM, Muhammad Shahzad > wrote: > Hi, > > I couple of my team members are working on translating a very long > Asterisk Dial Plan to FreeSWITCH XML Dial Plan. Now reference to > wiki link below, > > Before you go through all the trouble of translating the dialplan be > sure to review the application itself. In many cases just doing a > dialplan translation results in less efficient use of FreeSWITCH's > powerful features. Be sure that you are looking at the way > FreeSWITCH handles various situations and take advantage of its > power and ease of use. > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#About_Dialplan_Variables > > The dial plan variables are not getting initialized as expected. I > was just wondering if we move this variable get and set stuff to any > language module say mod_perl, will that make any difference > performance wise? I mean we will be invoking a Perl interpreter for > each incoming call, won't that be expensive in terms of RAM and CPU > usage and thus reducing number of calls this FS deployment can handle? > > I have guys with programming skills in Perl, PHP, Python, Java and > LUA languages. Which language do you recommend for this, again in > terms of speed and performance? > > > Lua is very portable and we've done tests with hundreds of > concurrent Lua scripts running. The other languages are heavier but > they'll still handle quite a few concurrent sessions. Just be sure > that you don't do the bridge app right in the script, use transfer > instead and have the dialplan process any bridging that you need to > do. > > -MC > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/38c7f804/attachment.html From mike at jerris.com Fri Sep 11 09:42:22 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Sep 2009 12:42:22 -0400 Subject: [Freeswitch-users] Call Transfer Problem In-Reply-To: <504742.69041.qm@web37507.mail.mud.yahoo.com> References: <504742.69041.qm@web37507.mail.mud.yahoo.com> Message-ID: Please open a bug on http://jira.freeswitch.org for this issue. Please test it on current svn trunk first as well. Mike On Sep 4, 2009, at 7:54 PM, DJB wrote: > I have a call transfer problem with Freeswitch > > Here is the call flow: > > I call from the PSTN (A party) into my Polycom phone (B-party) > which is registered to FreeSwtich. The Freeswtich is setup not to > route media as I have an SBC acting as a mirror proxy that will do > all the NAT and media routing. > > The inbound call is setup fine and there is two way voice. I then > blind transfer from the Polycom to my Cell phone. I see the polycom > send a SIP refer to Freeswitch and it sends a 202 accepted fine and > that leg between the Polycom (B party) and the A party is torn down > fine like its supposed to be. The Freeswitch places the outbound > call (the number the call is transferring to C-party) and that call > completes. However now there is one way audio between the A party > and C party . I see RTP streaming back from the egress carrier where > the call was transfered to so the A party can hear the C party but > the C party cannot hear the A party . When I look at the SIP traces > of the original inbound call from the A-party I see a SIP re-invite > from free switch to place the call on hold (contains Freeswitch RTP > address to I can hear hold music) while it is transferring the call > and the A-party does hear on hold music from Freeswitch while the > call is being transferred. However I do not see a second re-invite > from freeswitch to pass the media IP it got from the egress leg back > to the original inbound leg. If my inbound gateway does not get a re- > invite from Freeswitch to redirect its media to the new RTP address > of of the egress carrier it will not do so hence the one way voice. > > How do I get the Freeswitch to re-invite the original ingress leg > once it gets the SIP 183 from the egress with the new RTP info ? > Free switch is sending the first SIP re-invite to my inbound gateway > with new media IP (IP of itself) so the A-party can hear on hold > music , but does not send a second re-invite to my inbound gateway > after it receives the new RTP address from the egress carrier for > the call that was transferred back out. > > Thank you. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/aa9345f3/attachment-0001.html From mike at jerris.com Fri Sep 11 09:45:45 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Sep 2009 12:45:45 -0400 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> Message-ID: <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> Following up, did a bug get created for this issue? Mike On Sep 5, 2009, at 2:40 AM, mayamatakeshi wrote: > > On Fri, Sep 4, 2009 at 7:28 PM, mayamatakeshi > wrote: > > On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi > wrote: > Hello, > I'm testing FS support for the header Path (FS is behind opensips). > It pretty much works: I tested calling from one user to the other > and calls work perfectly. > However, I've noticed that when I register my terminal directly with > FS without going thru the proxy, I receive an unsolicited NOTIFY > containing Message-Waiting information. But when I register via > proxy, FS doesn't send this NOTIFY. > What could be causing this difference of behavior? (enabling debug > (F8) doesn't show anything for registration handling). > > I have enabled Sofia debug and I can see NTA is complaining about > invalid URI when building the NOTIFY: > > nua: nua_notify: entering > nua(0x9b3c1e8): sent signal r_notify > nua(0x9b3c1e8): recv signal r_notify > nua: nua_stack_set_params: entering > nua(0x9b3c1e8): adding notify usage with event message-summary > nta_leg_tcreate(0x9b74c68) > nta outgoing create: invalid URI > nta: outgoing_free(0x9b74928) > nua(0x9b3c1e8): event r_notify 900 Internal error at nua_client.c:711 > nua(0x9b3c1e8): removing notify usage with event message-summary > > My REGISTER relayed by opensips is this: > > REGISTER sip:test.com SIP/2.0 > Record-Route: > > Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKe62c.49e9f6c1.0 > Via: SIP/2.0/UDP > 192.168.2.121 > : > 5060 > ;received > = > 192.168.2.121 > ;rport=5060;branch=z9hG4bKPj4uAYgDuRbilYy4lCWcjlDKIDAtf-9RdS > Max-Forwards: 69 > From: ;tag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5 > To: > Call-ID: JvQ.apMLiJtfHa7z4ShIfgBPi5jIbtBv > CSeq: 14872 REGISTER > Contact: > Expires: 60 > Authorization: Digest username="user1", realm="test.com", > nonce="7d911eef-2c16-4deb-99f6-afcff9968a19", uri="sip: > 192.168.2.100", response="df29caeb78790b4527f1176622cbf192", > algorithm=MD5, cnonce="5.EXCbM3RZTx6iOh1cvUzUvEZTs2eheG", qop=auth, > nc=00000001 > Content-Length: 0 > Path: > > I hope someone can point out a problem. > I'm looking at NTA with gdb but I'm slow on this. > > The invalid URI nta is complaining about is the route_uri extracted > from the Contact stored upon registration. > The difference of behavior between INVITE (works) and NOTIFY > (doesn't work) via proxy, seems to be because for INVITE, mod_sofia > code (function sofia_glue_do_invite in sofia_glue.c) calls > sofia_overcome_sip_uri_weakness to adjust the route_uri. > But for a NOTIFY, this function is not called (and it cannot be > called, as there's no session which is required as a parameter). > > In my case I can see that basically what > sofia_overcome_sip_uri_weakness does is to remove the "<" , ">" > around the route_uri. > I messed with the code in sofia_glue_send_notify to just remove "<" > and ">" and after that I was able to receive the NOTIFY. > So I believe there is some code lacking in FS to properly permit > UAs registering via proxy to receive NOTIFY. > I might be wrong: if there is anyone using this scenario > successfully, please let me know. Otherwise, I'll open a ticket on > JIRA. > > regards, > takeshi > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/db56b7ec/attachment.html From shaheryarkh at googlemail.com Fri Sep 11 10:18:53 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 11 Sep 2009 23:18:53 +0600 Subject: [Freeswitch-users] mod_dingaling: dl_login command syntax In-Reply-To: References: Message-ID: actually, mod_dingaling is not reading configuration from xml_curl unless we reload mod_dingaling, which obviously fails if dingaling profile is in call etc. So, i am writing a patch right now to enable this functionality, almost finished just to perfect some memory management things. Thank you. On Fri, Sep 11, 2009 at 10:27 PM, Michael Jerris wrote: > What errors do you get? > Mike > > On Sep 4, 2009, at 8:35 AM, Muhammad Shahzad wrote: > > Hi, > > i am have FS SVN revision 14760, i am trying to use mod_xml_curl against > mod_dingaling. When i call xml_curl url in browser i get mod_dingaling > configuration correctly, also when i do reload mod_dingaling it fetches its > configuration from xml_curl correctly. BUT when i try to use dl_login > command to login a jingle profile it does not work. I have tried both > syntax, > > Syntax 1: > ======= > dl_login profile=abcd > > Where abcd is a valid jingle profile fetch-able from xml_curl. > > Syntax 2: > ======= > dl_login name=abcd;login= > XXX at gmail.com/talk;pass=YYY;dialplan=XML;context=public;rtp-ip=auto;sasl=plain;tls=true;exten=1001 > > All these values are correct and work if i reload mod_dingaling but they > don't work with dl_login, and give following output. > > USAGE: Existing Profile: > dl_login profile= > Dynamic Profile: > dl_login var1=val1;var2=val2;varN=valN > > I don't think xml_curl has any role in this syntax. > > Can you please correct me if i am doing something wrong in here or is it a > bug in mod_dingaling. > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/b135249c/attachment.html From jerry.richards at teotech.com Fri Sep 11 10:25:10 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Fri, 11 Sep 2009 10:25:10 -0700 Subject: [Freeswitch-users] Inbound Gateway Call Not Working Message-ID: I am trying to configure a Grandstream gateway to work with FS. I can make outbound calls without a problem. However, inbound calls are getting a 403 Forbidden from FS in response to the INVITE from the gateway. Now, the INVITE's from address is the caller's number (e.g. 1112223333), which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy Authentication Required and the gateway uses username "Anonymous" and the uri "sip:4000 at 192.168.72.38" (4000 is the destination for all calls from the gateway). Is there an example configuration for this scenario? Thanks and Best Regards, Jerry From chris at cloudtel.com Fri Sep 11 10:37:55 2009 From: chris at cloudtel.com (Chris Burns) Date: Fri, 11 Sep 2009 13:37:55 -0400 Subject: [Freeswitch-users] Music Background In-Reply-To: <8ccbff060909110922p41d9cb91raa39605e080ea9b7@mail.gmail.com> References: <8ccbff060909110922p41d9cb91raa39605e080ea9b7@mail.gmail.com> Message-ID: <200909111337.55373.chris@cloudtel.com> Check out the variables ringback and transfer_ringback. The local extension in the default dialplan is a good example. For romance, I recommend 80s rock ballads. YMMV. On September 11, 2009 12:22:20 pm Dome Charoenyost wrote: > Dear Sir, > Is posible to play music for background when call connect ? > Example when i call my wife some time i need romantic song :) > > BG > > Dome C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dome at tel.co.th Fri Sep 11 10:47:43 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sat, 12 Sep 2009 00:47:43 +0700 Subject: [Freeswitch-users] Music Background In-Reply-To: <200909111337.55373.chris@cloudtel.com> References: <8ccbff060909110922p41d9cb91raa39605e080ea9b7@mail.gmail.com> <200909111337.55373.chris@cloudtel.com> Message-ID: <8ccbff060909111047h2640ea9aj6eff6ff8bec45d49@mail.gmail.com> 2009/9/12 Chris Burns : > Check out the variables ringback and transfer_ringback. The local extension in > the default dialplan is a good example. Music rinback is Ok now. but I'm looking for solution for stream sound to channel both leg when call is answer. > > For romance, I recommend 80s rock ballads. YMMV. I'll try :) > > On September 11, 2009 12:22:20 pm Dome Charoenyost wrote: >> Dear Sir, >> ? ? ? ? ? ? Is posible to play music for background when call connect ? >> ? ? ?Example when i call my wife some time i need romantic song :) >> >> BG >> >> Dome C. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From brian at freeswitch.org Fri Sep 11 10:51:31 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Sep 2009 12:51:31 -0500 Subject: [Freeswitch-users] mod_dingaling: dl_login command syntax In-Reply-To: References: Message-ID: Also for tests make sure you fuzz test it also .. giving it invalid data shouldn't crash ... so try that when you're done too. /b On Sep 11, 2009, at 12:18 PM, Muhammad Shahzad wrote: > actually, mod_dingaling is not reading configuration from xml_curl > unless we reload mod_dingaling, which obviously fails if dingaling > profile is in call etc. > > So, i am writing a patch right now to enable this functionality, > almost finished just to perfect some memory management things. > > Thank you. From msc at freeswitch.org Fri Sep 11 10:54:01 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 Sep 2009 10:54:01 -0700 Subject: [Freeswitch-users] Inbound Gateway Call Not Working In-Reply-To: References: Message-ID: <87f2f3b90909111054s24db624dhf668517d610a9328@mail.gmail.com> On Fri, Sep 11, 2009 at 10:25 AM, Jerry Richards wrote: > I am trying to configure a Grandstream gateway to work with FS. I can make > outbound calls without a problem. However, inbound calls are getting a 403 > Forbidden from FS in response to the INVITE from the gateway. > > Now, the INVITE's from address is the caller's number (e.g. 1112223333), > which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy > Authentication Required and the gateway uses username "Anonymous" and the > uri "sip:4000 at 192.168.72.38 " (4000 is the > destination for all calls from the > gateway). > > Is there an example configuration for this scenario? > > Thanks and Best Regards, > Jerry > > Do you need authentication in this scenario? If not then you can add the gateway's IP address in the ACL "domains" in acl.conf.xml. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/a78f99d9/attachment.html From jerry.richards at teotech.com Fri Sep 11 10:56:52 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Fri, 11 Sep 2009 10:56:52 -0700 Subject: [Freeswitch-users] Inbound Gateway Call Not Working Message-ID: <78C2CE3D6E5946868352349681EF5034@greyhawk.tonecommander.com> By the way, the FS DEBUG console is saying the following when an inbound call is made: Rejected by acl "domains". Falling back to Digest auth. Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Friday, September 11, 2009 10:25 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: Inbound Gateway Call Not Working I am trying to configure a Grandstream gateway to work with FS. I can make outbound calls without a problem. However, inbound calls are getting a 403 Forbidden from FS in response to the INVITE from the gateway. Now, the INVITE's from address is the caller's number (e.g. 1112223333), which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy Authentication Required and the gateway uses username "Anonymous" and the uri "sip:4000 at 192.168.72.38" (4000 is the destination for all calls from the gateway). Is there an example configuration for this scenario? Thanks and Best Regards, Jerry From msc at freeswitch.org Fri Sep 11 11:03:09 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 11 Sep 2009 11:03:09 -0700 Subject: [Freeswitch-users] Inbound Gateway Call Not Working In-Reply-To: <78C2CE3D6E5946868352349681EF5034@greyhawk.tonecommander.com> References: <78C2CE3D6E5946868352349681EF5034@greyhawk.tonecommander.com> Message-ID: <87f2f3b90909111103o4627af88vf7cb465b4ca38ab8@mail.gmail.com> On Fri, Sep 11, 2009 at 10:56 AM, Jerry Richards wrote: > By the way, the FS DEBUG console is saying the following when an inbound > call is made: > > Rejected by acl "domains". Falling back to Digest auth. > > Yes, then it's a matter of whether you want to use digest auth or not. If not then just add a line like this in the domains section in acl.conf.xml: where x.x.x.x is the IP addr of the gateway from which the call is coming. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/6102eed1/attachment.html From m.sobkow at marketelsystems.com Fri Sep 11 11:03:58 2009 From: m.sobkow at marketelsystems.com (Mark Sobkow) Date: Fri, 11 Sep 2009 12:03:58 -0600 Subject: [Freeswitch-users] Missing sofia.conf? Message-ID: <4AAA910E.9020307@marketelsystems.com> When I try to d a load mod_sofia, I get an error message indicating that Freeswitch can't find sofia.conf. There _is_ a sofia.conf.xml in the autoload directory, which I _thought_ was the main sofia configuration file. Do I need to copy it to sofia.conf? If so, where do I copy it to? From chris at cloudtel.com Fri Sep 11 11:23:04 2009 From: chris at cloudtel.com (Chris Burns) Date: Fri, 11 Sep 2009 14:23:04 -0400 Subject: [Freeswitch-users] Music Background In-Reply-To: <8ccbff060909111047h2640ea9aj6eff6ff8bec45d49@mail.gmail.com> References: <8ccbff060909110922p41d9cb91raa39605e080ea9b7@mail.gmail.com> <200909111337.55373.chris@cloudtel.com> <8ccbff060909111047h2640ea9aj6eff6ff8bec45d49@mail.gmail.com> Message-ID: <200909111423.04248.chris@cloudtel.com> There are a few ways you could go about dropping into a conference and playing the song in from a separate channel. On September 11, 2009 01:47:43 pm Dome Charoenyost wrote: > 2009/9/12 Chris Burns : > > Check out the variables ringback and transfer_ringback. The local > > extension in the default dialplan is a good example. > > Music rinback is Ok now. but I'm looking for solution for stream sound > to channel both leg when call is answer. > > > For romance, I recommend 80s rock ballads. YMMV. > > I'll try :) > > > On September 11, 2009 12:22:20 pm Dome Charoenyost wrote: > >> Dear Sir, > >> ? ? ? ? ? ? Is posible to play music for background when call connect ? > >> ? ? ?Example when i call my wife some time i need romantic song :) > >> > >> BG > >> > >> Dome C. > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From csa at nowthor.com Fri Sep 11 11:31:34 2009 From: csa at nowthor.com (Carlos S. Antunes) Date: Fri, 11 Sep 2009 14:31:34 -0400 Subject: [Freeswitch-users] Late codec negotiation: any drawbacks? Message-ID: <4AAA9786.5090506@nowthor.com> Hello! As I have a fax machine connected to an adapter that does T.38 (Grandstream HandyTone 502), I am playing with late codec negotiation and proxy media. However, because late codec negotiation is a profile-wide affair, I would like to know if there are any potential drawbacks I should be aware of. Thanks! Carlos From brian at freeswitch.org Fri Sep 11 11:50:44 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Sep 2009 13:50:44 -0500 Subject: [Freeswitch-users] Missing sofia.conf? In-Reply-To: <4AAA910E.9020307@marketelsystems.com> References: <4AAA910E.9020307@marketelsystems.com> Message-ID: <6EAC63D4-C661-4C90-B37F-0DEB74AAE750@freeswitch.org> make samples /b On Sep 11, 2009, at 1:03 PM, Mark Sobkow wrote: > When I try to d a load mod_sofia, I get an error message indicating > that > Freeswitch can't find sofia.conf. There _is_ a sofia.conf.xml in the > autoload directory, which I _thought_ was the main sofia configuration > file. Do I need to copy it to sofia.conf? If so, where do I copy > it to? From janvb at live.com Fri Sep 11 12:17:29 2009 From: janvb at live.com (Jan Berger) Date: Fri, 11 Sep 2009 21:17:29 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Weekly Conference Starting, Please Call In! In-Reply-To: <7b197bef0909110912w10a65be3vcb9aa7bc6023c2ad@mail.gmail.com> References: <87f2f3b90909110901n166e8d9aq9639dbba20085926@mail.gmail.com> <7b197bef0909110912w10a65be3vcb9aa7bc6023c2ad@mail.gmail.com> Message-ID: hi, Was using this to listen in and most of the time it worked ok, but I had to close and call in loads of times because sound went crap - but that's probably skype - don't know. Jan > From: gmaruzz at celliax.org > Date: Fri, 11 Sep 2009 18:12:34 +0200 > To: freeswitch-users at lists.freeswitch.org > CC: freeswitch-dev at lists.freeswitch.org > Subject: Re: [Freeswitch-dev] [Freeswitch-users] FreeSWITCH Weekly Conference Starting, Please Call In! > > On Fri, Sep 11, 2009 at 6:01 PM, Michael Collins wrote: > > FYI, the conference is starting. Please join us! > > sip:888 at conference.freeswitch.org > > 213-799-1400 > > This time too, you all can follow the conference calling Skype the > skypeuser "skypiax5", then press "1" on the Skype dialpad (max 20 > concurrent users). > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org _________________________________________________________________ With Windows Live, you can organize, edit, and share your photos. http://www.microsoft.com/middleeast/windows/windowslive/products/photo-gallery-edit.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/9be16491/attachment.html From shaheryarkh at googlemail.com Fri Sep 11 12:52:46 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Sat, 12 Sep 2009 01:52:46 +0600 Subject: [Freeswitch-users] mod_dingaling: dl_login command syntax In-Reply-To: References: Message-ID: sure, i have a full QA department who will take case of all possible cases. Then it can be tested by our community. Thank you. On Fri, Sep 11, 2009 at 11:51 PM, Brian West wrote: > Also for tests make sure you fuzz test it also .. giving it invalid > data shouldn't crash ... so try that when you're done too. > > /b > > On Sep 11, 2009, at 12:18 PM, Muhammad Shahzad wrote: > > > actually, mod_dingaling is not reading configuration from xml_curl > > unless we reload mod_dingaling, which obviously fails if dingaling > > profile is in call etc. > > > > So, i am writing a patch right now to enable this functionality, > > almost finished just to perfect some memory management things. > > > > Thank you. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090912/381bde71/attachment-0001.html From brian at freeswitch.org Fri Sep 11 13:16:11 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Sep 2009 15:16:11 -0500 Subject: [Freeswitch-users] mod_dingaling: dl_login command syntax In-Reply-To: References: Message-ID: <1152D8AC-266A-44FA-9F4A-17E4B6D654DA@freeswitch.org> Kewl I have a fuzz test I do also thats automated that throws all kinds of crazy stuff at all the api's. /b On Sep 11, 2009, at 2:52 PM, Muhammad Shahzad wrote: > sure, i have a full QA department who will take case of all possible > cases. Then it can be tested by our community. > > Thank you. From jerry.richards at teotech.com Fri Sep 11 13:27:18 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Fri, 11 Sep 2009 13:27:18 -0700 Subject: [Freeswitch-users] Inbound Gateway Call Not Working Message-ID: <4C556285D65B41AAACFD70A662969D2C@greyhawk.tonecommander.com> Thanks. I added the to both the "lan" list and "domain" list in the acl.conf.xml file and it does not try to authenticate anymore. However, now it replies to the INVITE with a 480 TEMPORARILY UNAVAILABLE. Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Friday, September 11, 2009 10:57 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: RE: Inbound Gateway Call Not Working By the way, the FS DEBUG console is saying the following when an inbound call is made: Rejected by acl "domains". Falling back to Digest auth. Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Friday, September 11, 2009 10:25 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: Inbound Gateway Call Not Working I am trying to configure a Grandstream gateway to work with FS. I can make outbound calls without a problem. However, inbound calls are getting a 403 Forbidden from FS in response to the INVITE from the gateway. Now, the INVITE's from address is the caller's number (e.g. 1112223333), which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy Authentication Required and the gateway uses username "Anonymous" and the uri "sip:4000 at 192.168.72.38" (4000 is the destination for all calls from the gateway). Is there an example configuration for this scenario? Thanks and Best Regards, Jerry From shaheryarkh at googlemail.com Fri Sep 11 13:42:33 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Sat, 12 Sep 2009 02:42:33 +0600 Subject: [Freeswitch-users] mod_dingaling: dl_login command syntax In-Reply-To: <1152D8AC-266A-44FA-9F4A-17E4B6D654DA@freeswitch.org> References: <1152D8AC-266A-44FA-9F4A-17E4B6D654DA@freeswitch.org> Message-ID: great, can you share it with me? Thank you. On Sat, Sep 12, 2009 at 2:16 AM, Brian West wrote: > Kewl I have a fuzz test I do also thats automated that throws all > kinds of crazy stuff at all the api's. > > /b > > On Sep 11, 2009, at 2:52 PM, Muhammad Shahzad wrote: > > > sure, i have a full QA department who will take case of all possible > > cases. Then it can be tested by our community. > > > > Thank you. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090912/ebfe0f7b/attachment.html From brian at freeswitch.org Fri Sep 11 14:14:38 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 11 Sep 2009 16:14:38 -0500 Subject: [Freeswitch-users] mod_dingaling: dl_login command syntax In-Reply-To: References: <1152D8AC-266A-44FA-9F4A-17E4B6D654DA@freeswitch.org> Message-ID: I'll dig it up this weekend and get you a copy of it.. its a perl script that writes out some js that I run via jsrun /b On Sep 11, 2009, at 3:42 PM, Muhammad Shahzad wrote: > great, can you share it with me? > > Thank you. From jmesquita at freeswitch.org Fri Sep 11 16:57:41 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 11 Sep 2009 20:57:41 -0300 Subject: [Freeswitch-users] possible sofia_contact bug In-Reply-To: References: Message-ID: Just thinking out loud. Wouldn't be sofia_contact 180004 at 192.168.1.163 ? jmesquita On Fri, Sep 11, 2009 at 10:06 AM, Woody Dickson wrote: > Hi, > > I am having a strange problem here. sofia status shows that the user is > registered, but sofia_contact says the user is not registered. > Does anyone know why this is happening? > > > freeswitch at localhost.localdomain> sofia status profile internal reg 180004 > API CALL [sofia(status profile internal reg 180004)] output: > > Registrations: > > ================================================================================================= > Call-ID: 530339592782-1484696326482 at 192.168.1.163 > User: 180004 at 192.168.1.102 > Contact: 180004 > Agent: Voip Phone 1.0 > Status: Registered(UDP)(unknown) EXP(2009-09-12 04:59:36) > Host: localhost.localdomain > IP: 192.168.1.163 > Port: 9000 > Auth-User: 180004 > Auth-Realm: 192.168.1.102 > > > ================================================================================================= > > > freeswitch at localhost.localdomain> sofia_contact 180004 at 192.168.1.102 > API CALL [sofia_contact(180004 at 192.168.1.102)] output: > error/user_not_registered > > freeswitch at localhost.localdomain> > > freeswitch at localhost.localdomain> sofia_contact user/180004 > API CALL [sofia_contact(user/180004)] output: > error/facility_not_subscribed > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/f3d6dbee/attachment.html From lautram.mathieu at gmail.com Fri Sep 11 17:04:28 2009 From: lautram.mathieu at gmail.com (Mathieu Lautram) Date: Sat, 12 Sep 2009 02:04:28 +0200 Subject: [Freeswitch-users] Event-socket python uuid Message-ID: <4f1bf75e0909111704w1ad3602ej37bc3802320d78c7@mail.gmail.com> Hi there, Here is my problem: I'd like to set up a switchboard. I'm using python scripts called when a call is coming. Here is my public.xml: * * Next, in my test2.py, I put the uuid of the session in a database: *import os, cgi, MySQLdb, time from freeswitch import * def handler(session, args): uuid = session.getVariable("uuid") myconnection = MySQLdb.connect(host = "localhost", user = "root", passwd = "root", db = "testfreeswitch") mycursor = myconnection.cursor() mycursor.execute("INSERT INTO fileAttente VALUES (NULL, '0123456789', 'LIBRE', '" + uuid + "')") session.execute("park")* Everything runs fine from here. After this, I use telnet to have a connection to freeswitch. And... I'm stuck. I would like this: a file corresponding to the caller (test2) a file corresponding to the callee; This file will be pretty much the same that test2. a file which will be run in the same time that Freeswitch; This file will dialog with Freeswitch and the database. The real problem that I have is that I don't know how to create a new session corresponding to the callee (via a python script). If somebody could help me, it could be really great =) Thanks a lot Mathieu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090912/2ef787e1/attachment.html From mike at jerris.com Fri Sep 11 17:16:20 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 11 Sep 2009 20:16:20 -0400 Subject: [Freeswitch-users] Chat redirect In-Reply-To: <27c25bc40909060332i29a8139eh63adbbe98845a8c1@mail.gmail.com> References: <27c25bc40909051126t619e34e0q85fa8b097a61240f@mail.gmail.com> <27c25bc40909060332i29a8139eh63adbbe98845a8c1@mail.gmail.com> Message-ID: <95FAEA88-6C90-4C60-B13A-1E8612EA461F@jerris.com> This would require changes to the c code in mod_sofia. If you have a patch to change this behavior (probably should address configuration and authentication as well as this could be a denial of service path) you can post it to http://jira.freeswitch.org. Mike On Sep 6, 2009, at 6:32 AM, Juan Backson wrote: > Hi Brian, > > From the event socket, there is no message received when a MESSAGE > is sent from one sip user to another. If both users are registered, > I can send message between them. But if the receiving party is not > registered, I want to be able to store it. > > However, there is no way to intercept this MESSAGE. > > Is there anyway to solve this problem. > > thx, > jb > > On Sun, Sep 6, 2009 at 2:36 AM, Brian West > wrote: > Not automatically. But you could use the event socket to get the > message and forward it via ESL. > /b > > On Sep 5, 2009, at 1:26 PM, Juan Backson wrote: > > > > > If so, how can it be done? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/849a6e28/attachment-0001.html From yj13535428332 at gmail.com Fri Sep 11 18:39:20 2009 From: yj13535428332 at gmail.com (jun yang) Date: Sat, 12 Sep 2009 09:39:20 +0800 Subject: [Freeswitch-users] how can i tell freeswitch listen on 0.0.0.0 and a fixed ip In-Reply-To: <191c3a030909110810j49023634ta7dbf4e619e3c240@mail.gmail.com> References: <6536698d0909110421l347575ccp64bd5615854f473c@mail.gmail.com> <6536698d0909110555j5c498030o2589f3b36a867e60@mail.gmail.com> <8549660C-1DEF-4017-90CE-43FA51E6C929@freeswitch.org> <191c3a030909110810j49023634ta7dbf4e619e3c240@mail.gmail.com> Message-ID: <6536698d0909111839n1008791fq8bec549b531d1a20@mail.gmail.com> thanks for the info that it is a sip problem. seems it should be doc in wiki to explain that how to configure freeswitch so that client can connect from any interface, cause not everyone play with freeswitch is a sip guru. so thanks any way, i should learn more with sip and freeswitch. 2009/9/11 Anthony Minessale > sip in general cannot properly support binding to 0.0.0.0 for a UAS, there > is no easy way for the sip stack to know which traffic is for which host and > all of the outbound traffic will appear to go out a single interface when no > specific binding is made. > > running each ip on it's own profile is the correct way to do multi ip > configurations. > > > > On Fri, Sep 11, 2009 at 8:19 AM, Brian West wrote: > >> You can NOT bind to 0.0.0.0 you can however use ${local_ip_v4} and if the >> IP changes sofia will bounce the profile and update the IP. >> /b >> >> On Sep 11, 2009, at 7:55 AM, jun yang wrote: >> >> i also found that: >> >> 2009/7/17 Raul Fragoso >: >> >> >* You can not do that with a single profile. Each profile is bound to only >> >> *>* one local IP, so if you need to bind to more than one you will have to >> *>* create a new profile and set the specific sip-ip/rtp-ip params for them. >> *>* >> but cann't understand how to do..* >> >> >> >> 2009/9/11 jun yang >> >>> the os have three ip, one public ipv4 with adsl which is dynamic assigned >>> every time, two lan ip in diffrent scope, 192.169.0.2 ,192.168.5.2. >>> when freeswitch start ,it auto bind to the pubic ip, so the lan user >>> cann't connect to freeswitch use lan ip. >>> i have setting >>> >>> but have no effect, freeswitch also auto bind to the public ip. >>> any help is thanks. >>> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090912/695da7b1/attachment.html From mayamatakeshi at gmail.com Fri Sep 11 19:03:06 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Sat, 12 Sep 2009 11:03:06 +0900 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> Message-ID: <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris wrote: > Following up, did a bug get created for this issue? > Hello, yes. http://jira.freeswitch.org/browse/MODSOFIA-26 > > On Sep 5, 2009, at 2:40 AM, mayamatakeshi wrote: > > > On Fri, Sep 4, 2009 at 7:28 PM, mayamatakeshi wrote: > >> >> On Wed, Sep 2, 2009 at 7:59 PM, mayamatakeshi wrote: >> >>> Hello, >>> I'm testing FS support for the header Path (FS is behind opensips). >>> It pretty much works: I tested calling from one user to the other and >>> calls work perfectly. >>> However, I've noticed that when I register my terminal directly with FS >>> without going thru the proxy, I receive an unsolicited NOTIFY containing >>> Message-Waiting information. But when I register via proxy, FS doesn't send >>> this NOTIFY. >>> What could be causing this difference of behavior? (enabling debug (F8) >>> doesn't show anything for registration handling). >>> >> >> I have enabled Sofia debug and I can see NTA is complaining about invalid >> URI when building the NOTIFY: >> >> nua: nua_notify: entering >> nua(0x9b3c1e8): sent signal r_notify >> nua(0x9b3c1e8): recv signal r_notify >> nua: nua_stack_set_params: entering >> nua(0x9b3c1e8): adding notify usage with event message-summary >> nta_leg_tcreate(0x9b74c68) >> nta outgoing create: invalid URI >> nta: outgoing_free(0x9b74928) >> nua(0x9b3c1e8): event r_notify 900 Internal error at nua_client.c:711 >> nua(0x9b3c1e8): removing notify usage with event message-summary >> >> My REGISTER relayed by opensips is this: >> >> REGISTER sip:test.com SIP/2.0 >> Record-Route: < >> sip:192.168.2.100;lr=on;ftag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5> >> Via: SIP/2.0/UDP 192.168.2.100;branch=z9hG4bKe62c.49e9f6c1.0 >> Via: SIP/2.0/UDP 192.168.2.121:5060 >> ;received=192.168.2.121;rport=5060;branch=z9hG4bKPj4uAYgDuRbilYy4lCWcjlDKIDAtf-9RdS >> Max-Forwards: 69 >> From: >> >;tag=AhFSdiltk3H4mrmGXICgRHFiU59ZuCk5 >> To: > >> Call-ID: JvQ.apMLiJtfHa7z4ShIfgBPi5jIbtBv >> CSeq: 14872 REGISTER >> Contact: >> Expires: 60 >> Authorization: Digest username="user1", realm="test.com", >> nonce="7d911eef-2c16-4deb-99f6-afcff9968a19", uri="sip:192.168.2.100", >> response="df29caeb78790b4527f1176622cbf192", algorithm=MD5, >> cnonce="5.EXCbM3RZTx6iOh1cvUzUvEZTs2eheG", qop=auth, nc=00000001 >> Content-Length: 0 >> Path: >> ;lr;received=sip:192.168.2.121:5060> >> >> I hope someone can point out a problem. >> I'm looking at NTA with gdb but I'm slow on this. > > > The invalid URI nta is complaining about is the route_uri extracted from > the Contact stored upon registration. > The difference of behavior between INVITE (works) and NOTIFY (doesn't work) > via proxy, seems to be because for INVITE, mod_sofia code (function > sofia_glue_do_invite in sofia_glue.c) calls sofia_overcome_sip_uri_weakness > to adjust the route_uri. > But for a NOTIFY, this function is not called (and it cannot be called, as > there's no session which is required as a parameter). > > In my case I can see that basically what sofia_overcome_sip_uri_weakness > does is to remove the "<" , ">" around the route_uri. > I messed with the code in sofia_glue_send_notify to just remove "<" and ">" > and after that I was able to receive the NOTIFY. > So I believe there is some code lacking in FS to properly permit UAs > registering via proxy to receive NOTIFY. > I might be wrong: if there is anyone using this scenario successfully, > please let me know. Otherwise, I'll open a ticket on JIRA. > > regards, > takeshi > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090912/32f9ff37/attachment.html From yj13535428332 at gmail.com Fri Sep 11 19:09:59 2009 From: yj13535428332 at gmail.com (jun yang) Date: Sat, 12 Sep 2009 10:09:59 +0800 Subject: [Freeswitch-users] How to subscribe to custom event in cli? In-Reply-To: <191c3a030909110811n7904ca7fvec8c55ec062b5d7@mail.gmail.com> References: <6536698d0909110148r215b92d7k621764cdd55d2a26@mail.gmail.com> <191c3a030909110811n7904ca7fvec8c55ec062b5d7@mail.gmail.com> Message-ID: <6536698d0909111909u5cb2092fk4c92d250f81db30e@mail.gmail.com> found it. the correct typing in fs_cli is : /event plain CHANNELL_CREATE freeswitch at internal> /event plain CHANNEL_CREATE +OK event listener enabled plain 2009/9/11 Anthony Minessale > or from fs_cli > > /events plain all > > > > On Fri, Sep 11, 2009 at 8:21 AM, Brian West wrote: > >> You need to telnet to the socket or use fs_cli... example... >> >> telnet 0 8021 >> auth ClueCon >> events all plain (or what ever commands you wish to run) >> >> /b >> >> >> On Sep 11, 2009, at 3:48 AM, jun yang wrote: >> >> > how can i subscribe to custom event in cli. >> > cli: load mod_event_socket >> > say Module mod_event_socket Already Loaded! >> > but i use >> > cli: event plain CHANNEL_CREATE >> > return event: Command not found! >> > cli: api event plain CHANNEL_CREATE >> > return api: Command not found! >> > >> > then what is the correct command? >> > thanks for some hint! >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090912/9236838b/attachment-0001.html From marc at kasteris.com Fri Sep 11 20:22:20 2009 From: marc at kasteris.com (Marc Orenberg) Date: Fri, 11 Sep 2009 20:22:20 -0700 (PDT) Subject: [Freeswitch-users] Need help setting-up a Sangoma A101DE card. Message-ID: <477723.443.qm@web50811.mail.re2.yahoo.com> Hello, I'm having trouble setting-up a T1 in Alaska to work with a Sangoma A101DE card. I've followed the instructions on the Sangoma and FreeSWITCH websites, and a support guy from Sangoma has dialed-in twice. It could be an issue with the T1 itself, but I'm not sure how to rule that out. I'm happy to pay somebody for their time. My email is marc at kasteris.com Thanks, Marc -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090911/6f6c18c7/attachment.html From moises.silva at gmail.com Fri Sep 11 21:12:37 2009 From: moises.silva at gmail.com (Moises Silva) Date: Sat, 12 Sep 2009 00:12:37 -0400 Subject: [Freeswitch-users] Need help setting-up a Sangoma A101DE card. In-Reply-To: <477723.443.qm@web50811.mail.re2.yahoo.com> References: <477723.443.qm@web50811.mail.re2.yahoo.com> Message-ID: On Fri, Sep 11, 2009 at 11:22 PM, Marc Orenberg wrote: > I'm having trouble setting-up a T1 in Alaska to work with a Sangoma A101DE > card. > I've followed the instructions on the Sangoma and FreeSWITCH websites, and > a support guy from Sangoma has dialed-in twice. > Is this a PRI link? what does the Sangoma support guy said? I can take a look if you contact me in irc.freenode.org at #openzap -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy at sangoma.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090912/2955de9f/attachment.html From jmesquita at freeswitch.org Fri Sep 11 22:21:20 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 12 Sep 2009 02:21:20 -0300 Subject: [Freeswitch-users] Chat redirect In-Reply-To: <95FAEA88-6C90-4C60-B13A-1E8612EA461F@jerris.com> References: <27c25bc40909051126t619e34e0q85fa8b097a61240f@mail.gmail.com> <27c25bc40909060332i29a8139eh63adbbe98845a8c1@mail.gmail.com> <95FAEA88-6C90-4C60-B13A-1E8612EA461F@jerris.com> Message-ID: I am anxious to provide my first real patch into FreeSWITCH and since this looked like a good candidate, I looked at the code for a little while and I have a few thoughts about the subject. FreeSWITCH (mod_sofia) does not route chat messages to endpoints who are not reachable (obviously). If you look at the API, the mod_sofia won't even take the message if endpoint is not registered and will respond with "Cannot find user". So, basically, to implement what you are looking for, you need to have hooks set upon message receival (from mod_sofia point of view). mod_sofia only sends events on ESL when message has been sent to the destination endpoint. The way I see, there are 2 options here. The quick way and the hard (not so hard) way. The quick way is to just fire an event when registered user is not found and it will depende on something external to replay the message when user is offline. The longer way is to make the core queue offline messages and deliver them when user register. What I would like to hear from the core dudes is, which one is wanted? None is a good answer too. Regards, jmesquita On Fri, Sep 11, 2009 at 9:16 PM, Michael Jerris wrote: > This would require changes to the c code in mod_sofia. If you have a patch > to change this behavior (probably should address configuration and > authentication as well as this could be a denial of service path) you can > post it to http://jira.freeswitch.org. > Mike > > On Sep 6, 2009, at 6:32 AM, Juan Backson wrote: > > Hi Brian, > > From the event socket, there is no message received when a MESSAGE is sent > from one sip user to another. If both users are registered, I can send > message between them. But if the receiving party is not registered, I want > to be able to store it. > > However, there is no way to intercept this MESSAGE. > > Is there anyway to solve this problem. > > thx, > jb > > On Sun, Sep 6, 2009 at 2:36 AM, Brian West wrote: > >> Not automatically. But you could use the event socket to get the >> message and forward it via ESL. >> /b >> >> On Sep 5, 2009, at 1:26 PM, Juan Backson wrote: >> >> > >> > If so, how can it be done? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090912/d636597f/attachment.html From dome at tel.co.th Fri Sep 11 22:50:59 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Sat, 12 Sep 2009 12:50:59 +0700 Subject: [Freeswitch-users] Music Background In-Reply-To: <200909111423.04248.chris@cloudtel.com> References: <8ccbff060909110922p41d9cb91raa39605e080ea9b7@mail.gmail.com> <200909111337.55373.chris@cloudtel.com> <8ccbff060909111047h2640ea9aj6eff6ff8bec45d49@mail.gmail.com> <200909111423.04248.chris@cloudtel.com> Message-ID: <8ccbff060909112250w612121dbjf1e04e47aead5c36@mail.gmail.com> 2009/9/12 Chris Burns : > There are a few ways you could go about dropping into a conference and playing > the song in from a separate channel. Good idea :) Thank. > > On September 11, 2009 01:47:43 pm Dome Charoenyost wrote: >> 2009/9/12 Chris Burns : >> > Check out the variables ringback and transfer_ringback. The local >> > extension in the default dialplan is a good example. >> >> Music rinback is Ok now. but I'm looking for solution for stream sound >> to channel both leg when call is answer. >> >> > For romance, I recommend 80s rock ballads. YMMV. >> >> I'll try :) >> >> > On September 11, 2009 12:22:20 pm Dome Charoenyost wrote: >> >> Dear Sir, >> >> ? ? ? ? ? ? Is posible to play music for background when call connect ? >> >> ? ? ?Example when i call my wife some time i need romantic song :) >> >> >> >> BG >> >> >> >> Dome C. >> >> >> >> _______________________________________________ >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From woodydickson at gmail.com Sat Sep 12 00:10:14 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sat, 12 Sep 2009 15:10:14 +0800 Subject: [Freeswitch-users] possible sofia_contact bug In-Reply-To: References: Message-ID: Hi, I am pretty sure it should be user at domain as I have used it before. Does anyone know why sofia_contact does not return correct result? thx, woody 2009/9/12 Jo?o Mesquita > Just thinking out loud. Wouldn't be > > sofia_contact 180004 at 192.168.1.163 ? > > jmesquita > > On Fri, Sep 11, 2009 at 10:06 AM, Woody Dickson wrote: > >> Hi, >> >> I am having a strange problem here. sofia status shows that the user is >> registered, but sofia_contact says the user is not registered. >> Does anyone know why this is happening? >> >> >> freeswitch at localhost.localdomain> sofia status profile internal reg >> 180004 >> API CALL [sofia(status profile internal reg 180004)] output: >> >> Registrations: >> >> ================================================================================================= >> Call-ID: 530339592782-1484696326482 at 192.168.1.163 >> User: 180004 at 192.168.1.102 >> Contact: 180004 >> Agent: Voip Phone 1.0 >> Status: Registered(UDP)(unknown) EXP(2009-09-12 04:59:36) >> Host: localhost.localdomain >> IP: 192.168.1.163 >> Port: 9000 >> Auth-User: 180004 >> Auth-Realm: 192.168.1.102 >> >> >> ================================================================================================= >> >> >> freeswitch at localhost.localdomain> sofia_contact 180004 at 192.168.1.102 >> API CALL [sofia_contact(180004 at 192.168.1.102)] output: >> error/user_not_registered >> >> freeswitch at localhost.localdomain> >> >> freeswitch at localhost.localdomain> sofia_contact user/180004 >> API CALL [sofia_contact(user/180004)] output: >> error/facility_not_subscribed >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090912/8d493e08/attachment-0001.html From anthony.minessale at gmail.com Sat Sep 12 08:25:35 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 12 Sep 2009 10:25:35 -0500 Subject: [Freeswitch-users] possible sofia_contact bug In-Reply-To: References: Message-ID: <191c3a030909120825u4c5a7763ra4feda8c17107e22@mail.gmail.com> connect to sqlite directly with the sqlite3 binary and dump the record for that registration. On Sat, Sep 12, 2009 at 2:10 AM, Woody Dickson wrote: > Hi, > I am pretty sure it should be user at domain as I have used it before. > > Does anyone know why sofia_contact does not return correct result? > > thx, > woody > > 2009/9/12 Jo?o Mesquita > > Just thinking out loud. Wouldn't be >> >> sofia_contact 180004 at 192.168.1.163 ? >> >> jmesquita >> >> On Fri, Sep 11, 2009 at 10:06 AM, Woody Dickson wrote: >> >>> Hi, >>> >>> I am having a strange problem here. sofia status shows that the user is >>> registered, but sofia_contact says the user is not registered. >>> Does anyone know why this is happening? >>> >>> >>> freeswitch at localhost.localdomain> sofia status profile internal reg >>> 180004 >>> API CALL [sofia(status profile internal reg 180004)] output: >>> >>> Registrations: >>> >>> ================================================================================================= >>> Call-ID: 530339592782-1484696326482 at 192.168.1.163 >>> User: 180004 at 192.168.1.102 >>> Contact: 180004 >>> Agent: Voip Phone 1.0 >>> Status: Registered(UDP)(unknown) EXP(2009-09-12 04:59:36) >>> Host: localhost.localdomain >>> IP: 192.168.1.163 >>> Port: 9000 >>> Auth-User: 180004 >>> Auth-Realm: 192.168.1.102 >>> >>> >>> ================================================================================================= >>> >>> >>> freeswitch at localhost.localdomain> sofia_contact 180004 at 192.168.1.102 >>> API CALL [sofia_contact(180004 at 192.168.1.102)] output: >>> error/user_not_registered >>> >>> freeswitch at localhost.localdomain> >>> >>> freeswitch at localhost.localdomain> sofia_contact user/180004 >>> API CALL [sofia_contact(user/180004)] output: >>> error/facility_not_subscribed >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090912/8651dff1/attachment.html From kjv at ken-ton.com Sat Sep 12 16:24:52 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Sat, 12 Sep 2009 19:24:52 -0400 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) Message-ID: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> Seems normal, right??? Keep scrolling, or search for "JUST WRONG!" and you'll see it below... = = = = = = = = = = = = = = = = = = = = = = = = = ======================================================================== Name internal Domain Name pbx.ken-ton.com DBName sofia_reg_internal Pres Hosts Dialplan XML Context default Challenge Realm auto_to RTP-IP 96.243.27.2 SIP-IP 96.243.27.2 URL sip:mod_sofia at 96.243.27.2:5060 BIND-URL sip:mod_sofia at 96.243.27.2:5060 TLS-URL sip:mod_sofia at 96.243.27.2:5061 TLS-BIND-URL sips:mod_sofia at 96.243.27.2:5061;transport=tls HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS PCMU at 20i,GSM,H264,H263-2000,H263-1998,H263,H261 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 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: = = = = = = = = = = = = = = = = = = = = = = = = = ======================================================================== Call-ID: MTkxNDliNGM3MWY5M2Q4NjdlZDA0MDAwZjM0ZjdjMDY. User: 2200 at pbx.ken-ton.com Contact: "Karl Vesterling" Agent: eyeBeam release 1100v stamp 47069 Status: Registered(UDP)(unknown) EXP(2009-09-12 18:17:26) Host: router IP: 173.66.67.136 Port: 40025 Auth-User: 2200 Auth-Realm: pbx.ken-ton.com Call-ID: 0002fd3b-b4470013-5ae02248-67310c4d at 192.168.75.130 User: 2234 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:29:51) Host: router IP: 192.168.75.130 Port: 5060 Auth-User: 2234 Auth-Realm: pbx.ken-ton.com Call-ID: 001380c2-b3940011-47ebe4a8-3328920d at 192.168.75.137 User: 2293 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:29:51) Host: router IP: 192.168.75.137 Port: 5060 Auth-User: 2293 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-769b0013-1743ed8a-0f617acf at 192.168.75.135 User: 2225 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:29:51) Host: router IP: 192.168.75.135 Port: 5060 Auth-User: 2225 Auth-Realm: pbx.ken-ton.com Call-ID: 000532ff-77cc0012-4a8f333b-04872d27 at 192.168.75.136 User: 2284 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:29:52) Host: router IP: 192.168.75.136 Port: 5060 Auth-User: 2284 Auth-Realm: pbx.ken-ton.com Call-ID: 00053281-ce020011-198a5344-64b5b721 at 192.168.75.134 User: 2243 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:29:54) Host: router IP: 192.168.75.134 Port: 5060 Auth-User: 2243 Auth-Realm: pbx.ken-ton.com Call-ID: 000532d2-fc6c0014-2aaff36c-6ce913bf at 192.168.75.129 User: 2270 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:29:55) Host: router IP: 192.168.75.129 Port: 5060 Auth-User: 2270 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-79280017-1af73774-17068556 at 192.168.75.131 User: 2253 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:29:56) Host: router IP: 192.168.75.131 Port: 5060 Auth-User: 2253 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-78b00011-53989e6e-0f2f8cb0 at 192.168.75.133 User: 2263 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:29:57) Host: router IP: 192.168.75.133 Port: 5060 Auth-User: 2263 Auth-Realm: pbx.ken-ton.com Call-ID: 0002fd3b-b447000f-06dc2f61-1702f900 at 192.168.75.130 User: 2230 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:29:57) Host: router IP: 192.168.75.130 Port: 5060 Auth-User: 2230 Auth-Realm: pbx.ken-ton.com Call-ID: 001380c2-b3940010-23396419-6472d390 at 192.168.75.137 User: 2292 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:29:58) Host: router IP: 192.168.75.137 Port: 5060 Auth-User: 2292 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-769b000e-1c52af93-4acaad40 at 192.168.75.135 User: 2220 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:29:58) Host: router IP: 192.168.75.135 Port: 5060 Auth-User: 2220 Auth-Realm: pbx.ken-ton.com Call-ID: 000532ff-77cc000f-3086f2c1-3cf7bd42 at 192.168.75.136 User: 2281 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:29:59) Host: router IP: 192.168.75.136 Port: 5060 Auth-User: 2281 Auth-Realm: pbx.ken-ton.com Call-ID: 00053281-ce02000f-2ee652e6-4cec5d3d at 192.168.75.134 User: 2241 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:00) Host: router IP: 192.168.75.134 Port: 5060 Auth-User: 2241 Auth-Realm: pbx.ken-ton.com Call-ID: 000532d2-fc6c0015-7030077d-3cacc806 at 192.168.75.129 User: 2271 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:01) Host: router IP: 192.168.75.129 Port: 5060 Auth-User: 2271 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-79280018-703d39d8-019e6a52 at 192.168.75.131 User: 2254 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:02) Host: router IP: 192.168.75.131 Port: 5060 Auth-User: 2254 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-78b00010-7467a06c-2f59ed26 at 192.168.75.133 User: 2262 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:03) Host: router IP: 192.168.75.133 Port: 5060 Auth-User: 2262 Auth-Realm: pbx.ken-ton.com Call-ID: 0002fd3b-b4470011-7b62517a-2c9ce7a2 at 192.168.75.130 User: 2232 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:05) Host: router IP: 192.168.75.130 Port: 5060 Auth-User: 2232 Auth-Realm: pbx.ken-ton.com Call-ID: 000532ff-77cc0010-06bec2f0-4eea8b28 at 192.168.75.136 User: 2282 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:05) Host: router IP: 192.168.75.136 Port: 5060 Auth-User: 2282 Auth-Realm: pbx.ken-ton.com Call-ID: 001380c2-b394000f-75d10e7c-50cd182c at 192.168.75.137 User: 2291 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:07) Host: router IP: 192.168.75.137 Port: 5060 Auth-User: 2291 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-769b0012-4cdc3e28-2d2948f4 at 192.168.75.135 User: 2224 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:09) Host: router IP: 192.168.75.135 Port: 5060 Auth-User: 2224 Auth-Realm: pbx.ken-ton.com Call-ID: 00053281-ce02000e-390ffad7-0b3f733e at 192.168.75.134 User: 2240 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:09) Host: router IP: 192.168.75.134 Port: 5060 Auth-User: 2240 Auth-Realm: pbx.ken-ton.com Call-ID: 000532d2-fc6c0016-6916982d-1cb035ff at 192.168.75.129 User: 2272 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:09) Host: router IP: 192.168.75.129 Port: 5060 Auth-User: 2272 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-78b00013-586999c0-1420d05a at 192.168.75.133 User: 2265 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:09) Host: router IP: 192.168.75.133 Port: 5060 Auth-User: 2265 Auth-Realm: pbx.ken-ton.com Call-ID: 001380c2-b394000e-3a5432a8-3b9284d0 at 192.168.75.137 User: 2290 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:09) Host: router IP: 192.168.75.137 Port: 5060 Auth-User: 2290 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-79280016-1bfd52a8-1416e4cc at 192.168.75.131 User: 2252 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:09) Host: router IP: 192.168.75.131 Port: 5060 Auth-User: 2252 Auth-Realm: pbx.ken-ton.com Call-ID: 000532d2-fc6c0018-0b8e00af-409b8552 at 192.168.75.129 User: 2274 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:14) Host: router IP: 192.168.75.129 Port: 5060 Auth-User: 2274 Auth-Realm: pbx.ken-ton.com Call-ID: 00053281-ce020013-23362487-2d5c3318 at 192.168.75.134 User: 2245 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:15) Host: router IP: 192.168.75.134 Port: 5060 Auth-User: 2245 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-78b0000e-0af25ad7-140fed4b at 192.168.75.133 User: 2260 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:15) Host: router IP: 192.168.75.133 Port: 5060 Auth-User: 2260 Auth-Realm: pbx.ken-ton.com Call-ID: 0002fd3b-b4470012-2b9308f0-23b01ac6 at 192.168.75.130 User: 2233 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:15) Host: router IP: 192.168.75.130 Port: 5060 Auth-User: 2233 Auth-Realm: pbx.ken-ton.com Call-ID: 001380c2-b3940013-46bbb056-7fa7466c at 192.168.75.137 User: 2295 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:15) Host: router IP: 192.168.75.137 Port: 5060 Auth-User: 2295 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-79280014-0c429fb5-3cabf05c at 192.168.75.131 User: 2250 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:17) Host: router IP: 192.168.75.131 Port: 5060 Auth-User: 2250 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-769b0010-739085ed-079d79d6 at 192.168.75.135 User: 2222 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:18) Host: router IP: 192.168.75.135 Port: 5060 Auth-User: 2222 Auth-Realm: pbx.ken-ton.com Call-ID: 00053281-ce020012-2e061072-25b5755b at 192.168.75.134 User: 2244 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:18) Host: router IP: 192.168.75.134 Port: 5060 Auth-User: 2244 Auth-Realm: pbx.ken-ton.com Call-ID: 000532d2-fc6c0019-10263d50-0da47df1 at 192.168.75.129 User: 2275 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:18) Host: router IP: 192.168.75.129 Port: 5060 Auth-User: 2275 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-79280019-2358a693-53ad96cc at 192.168.75.131 User: 2255 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:18) Host: router IP: 192.168.75.131 Port: 5060 Auth-User: 2255 Auth-Realm: pbx.ken-ton.com Call-ID: 00055e37-78b0000f-141776f7-257f325a at 192.168.75.133 User: 2261 at pbx.ken-ton.com Contact: "user" Agent: Cisco-CP7960G/8.0 Status: Registered(UDP-NAT)(unknown) EXP(2009-09-12 16:30:19) Host: router IP: 192.168.75.133 Port: 5060 Auth-User: 2261 Auth-Realm: pbx.ken-ton.com = = = = = = = = = = = = = = = = = = = = = = = = = ======================================================================== JUST WRONG: perpetual and relentless, so much so that the switch is so pre- occupied it takes 25 seconds for it to service an incoming call, which fails miserably: recv 566 bytes from udp/[192.168.75.129]:5060 at 20:28:46.731704: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.129:5060;branch=z9hG4bK432d80f6 From: ;tag=000532d2fc6c518650ed3468-5830a71b To: Call-ID: 000532d2-fc6c0018-0b8e00af-409b8552 at 192.168.75.129 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4676 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.134]:5060 at 20:28:46.734944: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.134:5060;branch=z9hG4bK2d6de983 From: ;tag=00053281ce0251783088a827-43177dd4 To: Call-ID: 00053281-ce020013-23362487-2d5c3318 at 192.168.75.134 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4779 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.131]:5060 at 20:28:46.740820: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.131:5060;branch=z9hG4bK2290d738 From: ;tag=00055e377928516c1d1f7883-7f5b8ee0 To: Call-ID: 00055e37-79280016-1bfd52a8-1416e4cc at 192.168.75.131 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4604 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.130]:5060 at 20:28:46.775514: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.130:5060;branch=z9hG4bK2d8442ff From: ;tag=0002fd3bb447515e14255edd-1f577b05 To: Call-ID: 0002fd3b-b4470012-2b9308f0-23b01ac6 at 192.168.75.130 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4868 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.133]:5060 at 20:28:46.775709: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.133:5060;branch=z9hG4bK25179f6c From: ;tag=00055e3778b0516a435d5398-403f7e8c To: Call-ID: 00055e37-78b0000e-0af25ad7-140fed4b at 192.168.75.133 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4619 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.137]:5060 at 20:28:46.779659: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.137:5060;branch=z9hG4bK18abc4ba From: ;tag=001380c2b394516e127ce6c3-78ba082d To: Call-ID: 001380c2-b3940013-46bbb056-7fa7466c at 192.168.75.137 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:47 GMT CSeq: 4664 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.131]:5060 at 20:28:46.780593: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.131:5060;branch=z9hG4bK1771c72f From: ;tag=00055e377928516d0d6ae1fc-4149e0d9 To: Call-ID: 00055e37-79280014-0c429fb5-3cabf05c at 192.168.75.131 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4535 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.135]:5060 at 20:28:46.791205: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.135:5060;branch=z9hG4bK0cca70fd From: ;tag=00055e37769b517512958175-54b83912 To: Call-ID: 00055e37-769b0010-739085ed-079d79d6 at 192.168.75.135 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 3796 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.134]:5060 at 20:28:46.804976: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.134:5060;branch=z9hG4bK1190a04d From: ;tag=00053281ce02517951945329-58804fac To: Call-ID: 00053281-ce020012-2e061072-25b5755b at 192.168.75.134 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4615 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.129]:5060 at 20:28:46.821644: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.129:5060;branch=z9hG4bK031368e7 From: ;tag=000532d2fc6c518743228066-3d01e1a3 To: Call-ID: 000532d2-fc6c0019-10263d50-0da47df1 at 192.168.75.129 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4722 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.131]:5060 at 20:28:46.830668: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.131:5060;branch=z9hG4bK5a5f2250 From: ;tag=00055e377928516e4cda2b8a-3198b9dd To: Call-ID: 00055e37-79280019-2358a693-53ad96cc at 192.168.75.131 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4384 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.133]:5060 at 20:28:46.835486: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.133:5060;branch=z9hG4bK5e06486c From: ;tag=00055e3778b0516b0b12e866-7701faaa To: Call-ID: 00055e37-78b0000f-141776f7-257f325a at 192.168.75.133 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4112 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.130]:5060 at 20:28:46.855548: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.130:5060;branch=z9hG4bK0179f02a From: ;tag=0002fd3bb447515f7f6818f5-59484d1c To: Call-ID: 0002fd3b-b4470010-70cf642a-0981c906 at 192.168.75.130 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4184 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.137]:5060 at 20:28:46.859760: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.137:5060;branch=z9hG4bK4f4749b4 From: ;tag=001380c2b394516f4fd348fa-73cab379 To: Call-ID: 001380c2-b3940012-4b4ac5e6-5f1b0675 at 192.168.75.137 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:47 GMT CSeq: 4267 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.134]:5060 at 20:28:46.874887: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.134:5060;branch=z9hG4bK7ef0b21b From: ;tag=00053281ce02517a77ceba60-156b135f To: Call-ID: 00053281-ce020010-00a0cc83-3cecff1c at 192.168.75.134 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 5000 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.129]:5060 at 20:28:46.891647: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.129:5060;branch=z9hG4bK50f01a46 From: ;tag=000532d2fc6c518876b6bf1d-0ff1dace To: Call-ID: 000532d2-fc6c0017-3ec99613-1547698e at 192.168.75.129 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4066 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.130]:5060 at 20:28:46.905515: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.130:5060;branch=z9hG4bK7bc4f7eb From: ;tag=0002fd3bb4475160677c5e11-0dedbbe6 To: Call-ID: 0002fd3b-b4470014-60355a0f-1c9c5a2f at 192.168.75.130 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4619 REGISTERDisabled sip debugging on internal frees User-Agent: Cisco-CP7960G/8.0witch at internal> Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.131]:5060 at 20:28:46.910852: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.131:5060;branch=z9hG4bK27fda1d7 From: ;tag=00055e377928516f2a05a886-3429f16c To: Call-ID: 00055e37-79280015-5a76991f-540ee3ec at 192.168.75.131 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4129 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.133]:5060 at 20:28:46.915498: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.133:5060;branch=z9hG4bK70baf697 From: ;tag=00055e3778b0516c2541f92e-32c1caf4 To: Call-ID: 00055e37-78b00012-4843519d-016120a1 at 192.168.75.133 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4470 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.135]:5060 at 20:28:46.951252: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.135:5060;branch=z9hG4bK34d0a0f3 From: ;tag=00055e37769b51762563e1cb-31e49786 To: Call-ID: 00055e37-769b0011-5085c472-415fd0b0 at 192.168.75.135 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 3778 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.135]:5060 at 20:28:46.991182: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.135:5060;branch=z9hG4bK4a25b945 From: ;tag=00055e37769b51772f46b608-2badfd60 To: Call-ID: 00055e37-769b000f-7a8d37a7-5bf8bbb2 at 192.168.75.135 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4406 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.130]:5060 at 20:28:47.095360: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.130:5060;branch=z9hG4bK132f58f7 From: ;tag=0002fd3bb447515b5a6e3c5a-501a30cf To: Call-ID: 0002fd3b-b4470013-5ae02248-67310c4d at 192.168.75.130 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:27 GMT CSeq: 4472 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ send 643 bytes to udp/[192.168.75.137]:5060 at 20:28:47.132384: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.75.137:5060;branch=z9hG4bK3e373264 From: ;tag=001380c2b39451691725cba3-7e577a94 To: ;tag=Kg7yeKNjKmgQS Call-ID: 001380c2-b3940012-4b4ac5e6-5f1b0675 at 192.168.75.137 CSeq: 4266 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14819 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="pbx.ken-ton.com", nonce="187f112b-9dcb-40f6-9723-fdcd42cb0921", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.129]:5060 at 20:28:47.141573: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.129:5060;branch=z9hG4bK56f41d1e From: ;tag=000532d2fc6c51832cccf910-53b4c896 To: Call-ID: 000532d2-fc6c0014-2aaff36c-6ce913bf at 192.168.75.129 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:31 GMT CSeq: 4758 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.131]:5060 at 20:28:47.150656: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.131:5060;branch=z9hG4bK10385346 From: ;tag=00055e377928516a0e23bfd9-6b8b31c2 To: Call-ID: 00055e37-79280017-1af73774-17068556 at 192.168.75.131 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:31 GMT CSeq: 4693 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.131]:5060 at 20:28:47.430803: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.131:5060;branch=z9hG4bK66d20b08 From: ;tag=00055e377928516b633cb802-4673544a To: Call-ID: 00055e37-79280018-703d39d8-019e6a52 at 192.168.75.131 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:35 GMT CSeq: 4632 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ send 643 bytes to udp/[192.168.75.134]:5060 at 20:28:47.499644: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.75.134:5060;branch=z9hG4bK017abc7f From: ;tag=00053281ce025174466e4c3d-69bb0761 To: ;tag=mS0Qge6NgX69m Call-ID: 00053281-ce020010-00a0cc83-3cecff1c at 192.168.75.134 CSeq: 4999 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14819 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="pbx.ken-ton.com", nonce="cd905876-7efc-4483-926c-24676145bed3", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.137]:5060 at 20:28:47.509771: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.137:5060;branch=z9hG4bK133fe10f From: ;tag=001380c2b394516a2363dc42-1fe5e2f0 To: Call-ID: 001380c2-b3940011-47ebe4a8-3328920d at 192.168.75.137 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:28 GMT CSeq: 4469 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.137]:5060 at 20:28:47.639745: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.137:5060;branch=z9hG4bK30d1f36a From: ;tag=001380c2b394516c708e9a33-589b375a To: Call-ID: 001380c2-b394000f-75d10e7c-50cd182c at 192.168.75.137 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4142 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.135]:5060 at 20:28:47.651193: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.135:5060;branch=z9hG4bK0ca45296 From: ;tag=00055e37769b517471758c32-79d90303 To: Call-ID: 00055e37-769b0012-4cdc3e28-2d2948f4 at 192.168.75.135 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4183 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.134]:5060 at 20:28:47.654868: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.134:5060;branch=z9hG4bK08b6c271 From: ;tag=00053281ce025177177b47cb-042394a6 To: Call-ID: 00053281-ce02000e-390ffad7-0b3f733e at 192.168.75.134 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4916 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.129]:5060 at 20:28:47.671713: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.129:5060;branch=z9hG4bK62fbd82b From: ;tag=000532d2fc6c5185114c226a-27eb914b To: Call-ID: 000532d2-fc6c0016-6916982d-1cb035ff at 192.168.75.129 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4873 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.133]:5060 at 20:28:47.695442: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.133:5060;branch=z9hG4bK719f8650 From: ;tag=00055e3778b0516959175750-349569d2 To: Call-ID: 00055e37-78b00013-586999c0-1420d05a at 192.168.75.133 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4490 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.137]:5060 at 20:28:47.709748: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.137:5060;branch=z9hG4bK4b9a1d91 From: ;tag=001380c2b394516d6b76b31c-1b2c2fa2 To: Call-ID: 001380c2-b394000e-3a5432a8-3b9284d0 at 192.168.75.137 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 3951 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.129]:5060 at 20:28:47.731628: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.129:5060;branch=z9hG4bK432d80f6 From: ;tag=000532d2fc6c518650ed3468-5830a71b To: Call-ID: 000532d2-fc6c0018-0b8e00af-409b8552 at 192.168.75.129 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4676 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.134]:5060 at 20:28:47.734847: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.134:5060;branch=z9hG4bK2d6de983 From: ;tag=00053281ce0251783088a827-43177dd4 To: Call-ID: 00053281-ce020013-23362487-2d5c3318 at 192.168.75.134 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4779 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.131]:5060 at 20:28:47.740764: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.131:5060;branch=z9hG4bK2290d738 From: ;tag=00055e377928516c1d1f7883-7f5b8ee0 To: Call-ID: 00055e37-79280016-1bfd52a8-1416e4cc at 192.168.75.131 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4604 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.130]:5060 at 20:28:47.775293: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.130:5060;branch=z9hG4bK2d8442ff From: ;tag=0002fd3bb447515e14255edd-1f577b05 To: Call-ID: 0002fd3b-b4470012-2b9308f0-23b01ac6 at 192.168.75.130 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4868 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.133]:5060 at 20:28:47.775532: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.133:5060;branch=z9hG4bK25179f6c From: ;tag=00055e3778b0516a435d5398-403f7e8c To: Call-ID: 00055e37-78b0000e-0af25ad7-140fed4b at 192.168.75.133 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4619 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.137]:5060 at 20:28:47.779674: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.137:5060;branch=z9hG4bK18abc4ba From: ;tag=001380c2b394516e127ce6c3-78ba082d To: Call-ID: 001380c2-b3940013-46bbb056-7fa7466c at 192.168.75.137 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:47 GMT CSeq: 4664 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.131]:5060 at 20:28:47.780498: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.131:5060;branch=z9hG4bK1771c72f From: ;tag=00055e377928516d0d6ae1fc-4149e0d9 To: Call-ID: 00055e37-79280014-0c429fb5-3cabf05c at 192.168.75.131 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4535 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.135]:5060 at 20:28:47.791134: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.135:5060;branch=z9hG4bK0cca70fd From: ;tag=00055e37769b517512958175-54b83912 To: Call-ID: 00055e37-769b0010-739085ed-079d79d6 at 192.168.75.135 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 3796 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.134]:5060 at 20:28:47.804825: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.134:5060;branch=z9hG4bK1190a04d From: ;tag=00053281ce02517951945329-58804fac To: Call-ID: 00053281-ce020012-2e061072-25b5755b at 192.168.75.134 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4615 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.129]:5060 at 20:28:47.821617: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.129:5060;branch=z9hG4bK031368e7 From: ;tag=000532d2fc6c518743228066-3d01e1a3 To: Call-ID: 000532d2-fc6c0019-10263d50-0da47df1 at 192.168.75.129 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4722 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.131]:5060 at 20:28:47.830585: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.131:5060;branch=z9hG4bK5a5f2250 From: ;tag=00055e377928516e4cda2b8a-3198b9dd To: Call-ID: 00055e37-79280019-2358a693-53ad96cc at 192.168.75.131 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4384 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.133]:5060 at 20:28:47.835593: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.133:5060;branch=z9hG4bK5e06486c From: ;tag=00055e3778b0516b0b12e866-7701faaa To: Call-ID: 00055e37-78b0000f-141776f7-257f325a at 192.168.75.133 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4112 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.130]:5060 at 20:28:47.855368: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.130:5060;branch=z9hG4bK0179f02a From: ;tag=0002fd3bb447515f7f6818f5-59484d1c To: Call-ID: 0002fd3b-b4470010-70cf642a-0981c906 at 192.168.75.130 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4184 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.137]:5060 at 20:28:47.859657: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.137:5060;branch=z9hG4bK4f4749b4 From: ;tag=001380c2b394516f4fd348fa-73cab379 To: Call-ID: 001380c2-b3940012-4b4ac5e6-5f1b0675 at 192.168.75.137 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:47 GMT CSeq: 4267 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.134]:5060 at 20:28:47.874796: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.134:5060;branch=z9hG4bK7ef0b21b From: ;tag=00053281ce02517a77ceba60-156b135f To: Call-ID: 00053281-ce020010-00a0cc83-3cecff1c at 192.168.75.134 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 5000 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.129]:5060 at 20:28:47.891590: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.129:5060;branch=z9hG4bK50f01a46 From: ;tag=000532d2fc6c518876b6bf1d-0ff1dace To: Call-ID: 000532d2-fc6c0017-3ec99613-1547698e at 192.168.75.129 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4066 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.130]:5060 at 20:28:47.905385: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.130:5060;branch=z9hG4bK7bc4f7eb From: ;tag=0002fd3bb4475160677c5e11-0dedbbe6 To: Call-ID: 0002fd3b-b4470014-60355a0f-1c9c5a2f at 192.168.75.130 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:45 GMT CSeq: 4619 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.131]:5060 at 20:28:47.910652: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.131:5060;branch=z9hG4bK27fda1d7 From: ;tag=00055e377928516f2a05a886-3429f16c To: Call-ID: 00055e37-79280015-5a76991f-540ee3ec at 192.168.75.131 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4129 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.133]:5060 at 20:28:47.915589: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.133:5060;branch=z9hG4bK70baf697 From: ;tag=00055e3778b0516c2541f92e-32c1caf4 To: Call-ID: 00055e37-78b00012-4843519d-016120a1 at 192.168.75.133 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4470 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.135]:5060 at 20:28:47.951193: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.135:5060;branch=z9hG4bK34d0a0f3 From: ;tag=00055e37769b51762563e1cb-31e49786 To: Call-ID: 00055e37-769b0011-5085c472-415fd0b0 at 192.168.75.135 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 3778 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.135]:5060 at 20:28:47.991099: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.135:5060;branch=z9hG4bK4a25b945 From: ;tag=00055e37769b51772f46b608-2badfd60 To: Call-ID: 00055e37-769b000f-7a8d37a7-5bf8bbb2 at 192.168.75.135 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:46 GMT CSeq: 4406 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ send 643 bytes to udp/[192.168.75.135]:5060 at 20:28:47.997528: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.75.135:5060;branch=z9hG4bK09847b07 From: ;tag=00055e37769b5171654c0618-3c6e065f To: ;tag=N2Sgj9pSD6vvg Call-ID: 00055e37-769b000f-7a8d37a7-5bf8bbb2 at 192.168.75.135 CSeq: 4405 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14819 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="pbx.ken-ton.com", nonce="43013c55- e58e-4c39-b6fa-ff0a5317ad15", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.133]:5060 at 20:28:48.025467: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.133:5060;branch=z9hG4bK5b7bce81 From: ;tag=00055e3778b0516705cd3b3a-656a7e1d To: Call-ID: 00055e37-78b00011-53989e6e-0f2f8cb0 at 192.168.75.133 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:32 GMT CSeq: 4579 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ recv 566 bytes from udp/[192.168.75.135]:5060 at 20:28:48.029825: ------------------------------------------------------------------------ REGISTER sip:pbx.ken-ton.com SIP/2.0 Via: SIP/2.0/UDP 192.168.75.135:5060;branch=z9hG4bK1324745c From: ;tag=00055e37769b51721c4497a3-2b7e38b3 To: Call-ID: 00055e37-769b0013-1743ed8a-0f617acf at 192.168.75.135 Max-Forwards: 70 Date: Sat, 12 Sep 2009 20:28:28 GMT CSeq: 3998 REGISTER User-Agent: Cisco-CP7960G/8.0 Contact: ; +sip.instance="";+u.sip! model.ccm.cisco.com="7" Content-Length: 0 Expires: 30 ------------------------------------------------------------------------ Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 From brian at freeswitch.org Sat Sep 12 18:19:02 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 12 Sep 2009 20:19:02 -0500 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> Message-ID: <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> Sounds like you have Force-RPORT on which you can't do with a 7960. /b On Sep 12, 2009, at 6:24 PM, Karl Vesterling wrote: > > Seems normal, right??? Keep scrolling, or search for "JUST WRONG!" and > you'll see it below... From brian at freeswitch.org Sat Sep 12 18:20:55 2009 From: brian at freeswitch.org (Brian West) Date: Sat, 12 Sep 2009 20:20:55 -0500 Subject: [Freeswitch-users] possible sofia_contact bug In-Reply-To: <191c3a030909120825u4c5a7763ra4feda8c17107e22@mail.gmail.com> References: <191c3a030909120825u4c5a7763ra4feda8c17107e22@mail.gmail.com> Message-ID: <64B3FED1-AC06-42D1-AEE0-38797A0D3CD6@freeswitch.org> Also I'm going to suspect you have removed the domain aliases from the profile. If you have then you can't just do sofia_contact user at domain... You must do sofia_contact profile/user at domain since your hint for the domain is no longer on the profile. /b On Sep 12, 2009, at 10:25 AM, Anthony Minessale wrote: > connect to sqlite directly with the sqlite3 binary and dump the > record for that registration. From msc at freeswitch.org Sat Sep 12 19:06:05 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 12 Sep 2009 19:06:05 -0700 Subject: [Freeswitch-users] Zaptel Analog FXO Outbound Dialing Message-ID: <4252B0B6-8C0F-4BDB-93DB-5CF5CC66FA14@freeswitch.org> Gang, Does anyone have a working analog setup? I've reproduced symptoms on two different systems, one with the Rhino 8 port card and one with the Nxtvox TDM400 clone. Symptoms are identical when dialing out on the FXO port. The outbound dialing established a connection with the called party but audio does not flow to the endpoint until approx 30 seconds after the call is answered. Digilord is having the same issue. Anyway, before I open a bug report I wanted some feedback from those who are using FXO analog ports. Please let me know of your experiences. Thanks! -MC Sent from my iPhone From sprice at gmail.com Sat Sep 12 20:58:22 2009 From: sprice at gmail.com (SP) Date: Sat, 12 Sep 2009 22:58:22 -0500 Subject: [Freeswitch-users] Zaptel Analog FXO Outbound Dialing In-Reply-To: <4252B0B6-8C0F-4BDB-93DB-5CF5CC66FA14@freeswitch.org> References: <4252B0B6-8C0F-4BDB-93DB-5CF5CC66FA14@freeswitch.org> Message-ID: <7e2ac3270909122058x4b25bbf0h8f98a469621cab6f@mail.gmail.com> I have a Sangoma U100 usbfxo setup with wanpipe and zaptel. Audio is immediate. Obviously your milage may vary. :) On Sat, Sep 12, 2009 at 21:06, Michael S Collins wrote: > Gang, > > Does anyone have a working analog setup? I've reproduced symptoms on > two different systems, one with the Rhino 8 port card and one with the > Nxtvox TDM400 clone. Symptoms are identical when dialing out on the > FXO port. The outbound dialing established a connection with the > called party but audio does not flow to the endpoint until approx 30 > seconds after the call is answered. Digilord is having the same issue. > Anyway, before I open a bug report I wanted some feedback from those > who are using FXO analog ports. > > Please let me know of your experiences. Thanks! > -MC > > Sent from my iPhone > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Shannon From msc at freeswitch.org Sat Sep 12 22:15:46 2009 From: msc at freeswitch.org (Michael S Collins) Date: Sat, 12 Sep 2009 22:15:46 -0700 Subject: [Freeswitch-users] Zaptel Analog FXO Outbound Dialing In-Reply-To: <7e2ac3270909122058x4b25bbf0h8f98a469621cab6f@mail.gmail.com> References: <4252B0B6-8C0F-4BDB-93DB-5CF5CC66FA14@freeswitch.org> <7e2ac3270909122058x4b25bbf0h8f98a469621cab6f@mail.gmail.com> Message-ID: <91CDA5F8-D8DF-4864-A915-07A146599331@freeswitch.org> Sent from my iPhone On Sep 12, 2009, at 8:58 PM, SP wrote: > I have a Sangoma U100 usbfxo setup with wanpipe and zaptel. Audio > is immediate. > > Obviously your milage may vary. :) > My suspicion is that this is only for zaptel type cards. Our tests with Sangoma analog cards have all been pretty successful. But thanks for info! Anyone else using Rhino, Digium, or compatible analog cards? Thanks! -MC > On Sat, Sep 12, 2009 at 21:06, Michael S Collins > wrote: >> Gang, >> >> Does anyone have a working analog setup? I've reproduced symptoms on >> two different systems, one with the Rhino 8 port card and one with >> the >> Nxtvox TDM400 clone. Symptoms are identical when dialing out on the >> FXO port. The outbound dialing established a connection with the >> called party but audio does not flow to the endpoint until approx 30 >> seconds after the call is answered. Digilord is having the same >> issue. >> Anyway, before I open a bug report I wanted some feedback from those >> who are using FXO analog ports. >> >> Please let me know of your experiences. Thanks! >> -MC >> >> Sent from my iPhone >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Shannon > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mcampbellsmith at gmail.com Sun Sep 13 02:34:23 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 13 Sep 2009 19:34:23 +1000 Subject: [Freeswitch-users] Unknown call drops.. INFO DTMF(3) Message-ID: <33c87fa30909130234u21541ae7o61c522d014e3dacd@mail.gmail.com> Hi! I have just experienced some call drops and each time the sequence is the same in the freeswitch.log file. Both parties are sure that they did not accidentally hit the 3 button to send the DTMF tone (and the same thing has happened four times already after ~5 minutes). 2009-09-13 19:28:23.835216 [DEBUG] sofia.c:4329 INFO DTMF(3) 2009-09-13 19:28:23.835216 [DEBUG] sofia.c:4450 dispatched freeswitch event for INFO 2009-09-13 19:28:23.859408 [DEBUG] switch_rtp.c:1624 Send start packet for [3] ts=2591120 dur=160/160/13120 seq=64923 2009-09-13 19:28:23.879439 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=320/320/13120 seq=64924 2009-09-13 19:28:23.899455 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=480/480/13120 seq=64925 : : 2009-09-13 19:28:25.439404 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=12800/12800/13120 seq=65002 2009-09-13 19:28:25.459312 [DEBUG] switch_rtp.c:1560 Send middle packet for [3] ts=2591120 dur=12960/12960/13120 seq=65003 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65004 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65005 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65006 2009-09-13 19:28:33.879341 [NOTICE] sofia.c:322 Hangup sofia/external/ [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2009-09-13 19:28:33.879341 [DEBUG] switch_channel.c:1683 Send signal sofia/external/ [KILL] 2009-09-13 19:28:33.879341 [DEBUG] switch_core_session.c:932 Send signal sofia/external/ [BREAK] 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:371 sofia/external/ ending bridge by request from write function 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:426 sofia/internal_nat/1000 at 192.168.1.120 receive message [UNBRIDGE] 2009-09-13 19:28:33.900940 [DEBUG] switch_core_session.c:630 Send signal sofia/internal_nat/1000 at 192.168.1.120 [BREAK] 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:452 BRIDGE THREAD DONE [sofia/internal_nat/1000 at 192.168.1.120] 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:454 Send signal sofia/external/ [BREAK] 2009-09-13 19:28:33.912049 [NOTICE] switch_core_state_machine.c:179 Hangup sofia/internal_nat/1000 at 192.168.1.120 [CS_EXECUTE] [NORMAL_CLEARING] Anyone have any idea what this sequence means and why I am getting this? Is it my sip provider or something in FreeSwitch? What does the 'Send end packet for [3] ts=2591120 dur=13120/13120/13120 seq=65006' mean? Notice that dur (duration?) is increasing a lot until the call drops. Thanks! From woodydickson at gmail.com Sun Sep 13 05:20:42 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sun, 13 Sep 2009 20:20:42 +0800 Subject: [Freeswitch-users] possible sofia_contact bug In-Reply-To: <64B3FED1-AC06-42D1-AEE0-38797A0D3CD6@freeswitch.org> References: <191c3a030909120825u4c5a7763ra4feda8c17107e22@mail.gmail.com> <64B3FED1-AC06-42D1-AEE0-38797A0D3CD6@freeswitch.org> Message-ID: Thanks Brian, you are correct. My problem is solved. Thank you so much. On Sun, Sep 13, 2009 at 9:20 AM, Brian West wrote: > Also I'm going to suspect you have removed the domain aliases from the > profile. If you have then you can't just do sofia_contact > user at domain... You must do sofia_contact profile/user at domain since > your hint for the domain is no longer on the profile. > > /b > > On Sep 12, 2009, at 10:25 AM, Anthony Minessale wrote: > > > connect to sqlite directly with the sqlite3 binary and dump the > > record for that registration. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090913/4ce8a17e/attachment.html From woodydickson at gmail.com Sun Sep 13 05:22:22 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Sun, 13 Sep 2009 20:22:22 +0800 Subject: [Freeswitch-users] voicemail problem Message-ID: Hi, While trying to record some sounds with the voicemail app, I keep getting message saying my record is below the minimal length even I was actually still speaking. Is it not detecting my voice? How can I configure it so that freeswitch's vm app can detect my speech? Thanks, woody -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090913/9d7a9db9/attachment.html From kjv at ken-ton.com Sun Sep 13 06:10:32 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Sun, 13 Sep 2009 09:10:32 -0400 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> Message-ID: That's a negative Brian. There is so much registration traffic, that (theory) any incoming calls take 25 seconds before they're even shown in the CLI. And of course, they fail. Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Sep 12, 2009, at 9:19 PM, Brian West wrote: > Sounds like you have Force-RPORT on which you can't do with a 7960. > > /b > > On Sep 12, 2009, at 6:24 PM, Karl Vesterling wrote: > >> >> Seems normal, right??? Keep scrolling, or search for "JUST WRONG!" >> and >> you'll see it below... > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > From markmorreny at gmail.com Sun Sep 13 07:17:44 2009 From: markmorreny at gmail.com (mark morreny) Date: Sun, 13 Sep 2009 22:17:44 +0800 Subject: [Freeswitch-users] fifo strategy and fifo outbound Message-ID: <20ad6b920909130717w4ed524f7i70a047981cf82097@mail.gmail.com> Hi, I want to know how fifo_strategy is to be used. When specifying MOST_PPL or WAIT_LONGEST, how will freeswitch select the caller for consumer? Does consumers wait on more than one fifo and freeswitch would select the fifo based on either MOST_PPL or WAIT_LONGEST? Then, freeswitch will just select the caller that has waited longest? Is that how it works? Does fifo has the ability to make outbound call to callers? If so, how to set it up? thx, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090913/9bbc853f/attachment-0001.html From gcd at i.ph Sun Sep 13 07:27:51 2009 From: gcd at i.ph (Nandy Dagondon) Date: Sun, 13 Sep 2009 22:27:51 +0800 Subject: [Freeswitch-users] FS 1.0.4 erl configure error Message-ID: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> hi, i want to enable odbc support which is required in mod_lcr feature. however, i encounter ./configure problem after installing Erlang R13B01. this is the portion of the error messages: ....... checking for erl... /usr/local/bin/erl checking erlang version... 5.7.2 checking erlang libdir... /usr/local/lib/erlang/lib/erl_interface-3.6.2/lib checking erlang incdir... /usr/local/lib/erlang/lib/erl_interface-3.6.2/include checking ei.h usability... yes checking ei.h presence... no configure: WARNING: ei.h: accepted by the compiler, rejected by the preprocessor! configure: WARNING: ei.h: proceeding with the compiler's result checking for ei.h... yes checking for ei_encode_version in -lei... yes checking for ei_link_unlink in -lei... no configure: Your erlang seems OK, do not forget to enable mod_erlang_event in modules.conf configure: creating ./config.status config.status: creating src/include/switch_version.h.in .infig.status: error: cannot find input file: Makefile -------- END -------- i set ERL_TOP environment variable to the source directory. has anyone encountered this problem? can anyone give me a hint what's wrong. i'm compiling FS 1.0.4. thank you, /nandy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090913/6be3d063/attachment.html From tayeb.meftah at gmail.com Sun Sep 13 08:56:01 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Sun, 13 Sep 2009 15:56:01 +0000 Subject: [Freeswitch-users] FS 1.0.4 erl configure error In-Reply-To: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> References: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> Message-ID: <4AAD1611.2030803@gmail.com> hello, i think you enabled mod_erlang_event in the modules.conf install unixodbc if is not installed thanks Nandy Dagondon a ?crit : > hi, > > i want to enable odbc support which is required in mod_lcr feature. > however, i encounter ./configure problem after installing Erlang > R13B01. this is the portion of the error messages: > > ....... > checking for erl... /usr/local/bin/erl > checking erlang version... 5.7.2 > checking erlang libdir... > /usr/local/lib/erlang/lib/erl_interface-3.6.2/lib > checking erlang incdir... > /usr/local/lib/erlang/lib/erl_interface-3.6.2/include > checking ei.h usability... yes > checking ei.h presence... no > configure: WARNING: ei.h: accepted by the compiler, rejected by the > preprocessor! > configure: WARNING: ei.h: proceeding with the compiler's result > checking for ei.h... yes > checking for ei_encode_version in -lei... yes > checking for ei_link_unlink in -lei... no > configure: Your erlang seems OK, do not forget to enable > mod_erlang_event in modules.conf > configure: creating ./config.status > config.status: creating src/include/switch_version.h.in > > .infig.status: error: cannot find input file: Makefile > -------- END -------- > > i set ERL_TOP environment variable to the source directory. has anyone > encountered this problem? can anyone give me a hint what's wrong. i'm > compiling FS 1.0.4. > > thank you, > /nandy > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4421 (20090913) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4421 (20090913) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090913/96580c19/attachment.html From markmorreny at gmail.com Sun Sep 13 08:01:10 2009 From: markmorreny at gmail.com (mark morreny) Date: Sun, 13 Sep 2009 23:01:10 +0800 Subject: [Freeswitch-users] skill-based ACD Message-ID: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> Hello Has any tried setting up an ACD based on skillset? The current out-of-box version of fifo does not seem to support acd based on agent skillset. Does anyone have any experience in doing it with some external scripting using lua or javascript? I am interested in hearing how others may have done it as I am trying to implement one myself. thx, mark -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090913/bd471840/attachment.html From brian at freeswitch.org Sun Sep 13 08:15:59 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 13 Sep 2009 10:15:59 -0500 Subject: [Freeswitch-users] voicemail problem In-Reply-To: References: Message-ID: that means the media isn't actually making it to the FS server I suspect. /b On Sep 13, 2009, at 7:22 AM, Woody Dickson wrote: > Hi, > > While trying to record some sounds with the voicemail app, I keep > getting message saying my record is below the minimal length even I > was actually still speaking. > > Is it not detecting my voice? How can I configure it so that > freeswitch's vm app can detect my speech? > > Thanks, > woody From kjv at ken-ton.com Sun Sep 13 12:59:25 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Sun, 13 Sep 2009 15:59:25 -0400 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> Message-ID: <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> RESOLVED!!! Folks, evidently this is a problem with Cisco Firmware P0S3-08-11-00 I forgot that (a long long time ago) I had dropped that firmware into that site. Phones hadn't been rebooted in (a while)... Oddly enough, once you get past (X) number of phones, the registration chatter created by the bug was too much for FS to keep up with. P0S3-08-8-00 works perfectly fine. Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Sep 13, 2009, at 9:10 AM, Karl Vesterling wrote: > That's a negative Brian. > > There is so much registration traffic, that (theory) any incoming > calls take 25 seconds before they're even shown in the CLI. And of > course, they fail. > > > Best Regards, > Karl J. Vesterling > kjv at ken-ton.com > 202-461-3231 x0 > > On Sep 12, 2009, at 9:19 PM, Brian West wrote: > >> Sounds like you have Force-RPORT on which you can't do with a 7960. >> >> /b >> >> On Sep 12, 2009, at 6:24 PM, Karl Vesterling wrote: >> >>> >>> Seems normal, right??? Keep scrolling, or search for "JUST WRONG!" >>> and >>> you'll see it below... >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > From brian at freeswitch.org Sun Sep 13 13:23:11 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 13 Sep 2009 15:23:11 -0500 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> Message-ID: I haven't seen this issue in 8.12 either... Maybe thats why 8.11 isn't on the website last I checked? /b On Sep 13, 2009, at 2:59 PM, Karl Vesterling wrote: > RESOLVED!!! > > Folks, evidently this is a problem with Cisco Firmware P0S3-08-11-00 > > I forgot that (a long long time ago) I had dropped that firmware into > that site. > Phones hadn't been rebooted in (a while)... > > Oddly enough, once you get past (X) number of phones, the registration > chatter created by the bug was too much for FS to keep up with. > > P0S3-08-8-00 works perfectly fine. > > > Best Regards, > Karl J. Vesterling > kjv at ken-ton.com > 202-461-3231 x0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090913/8ac1a3a8/attachment.html From kjv at ken-ton.com Sun Sep 13 15:30:27 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Sun, 13 Sep 2009 18:30:27 -0400 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> Message-ID: <8A58CD3E-F0C3-4BAD-8DB9-3C9D3A22A70F@ken-ton.com> New development. Even though the initial registration succeeds, the subsequent registrations fail... ??Search me?? But that's just too weird for me... Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Sep 13, 2009, at 4:23 PM, Brian West wrote: > I haven't seen this issue in 8.12 either... Maybe thats why 8.11 > isn't on the website last I checked? > > /b > > On Sep 13, 2009, at 2:59 PM, Karl Vesterling wrote: > >> RESOLVED!!! >> >> Folks, evidently this is a problem with Cisco Firmware P0S3-08-11-00 >> >> I forgot that (a long long time ago) I had dropped that firmware into >> that site. >> Phones hadn't been rebooted in (a while)... >> >> Oddly enough, once you get past (X) number of phones, the >> registration >> chatter created by the bug was too much for FS to keep up with. >> >> P0S3-08-8-00 works perfectly fine. >> >> >> Best Regards, >> Karl J. Vesterling >> kjv at ken-ton.com >> 202-461-3231 x0 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090913/25521192/attachment-0001.html From panayotov.vd at gmail.com Sun Sep 13 23:02:49 2009 From: panayotov.vd at gmail.com (Vassil Panayotov) Date: Mon, 14 Sep 2009 09:02:49 +0300 Subject: [Freeswitch-users] Sangoma A500 - dial out from specific port group? In-Reply-To: <8a9b664c0909092354v62cc84b8mcf525d46e91c3e72@mail.gmail.com> References: <8a9b664c0909082320l3be47aedk8d67c2c83d5b411d@mail.gmail.com> <8a9b664c0909092354v62cc84b8mcf525d46e91c3e72@mail.gmail.com> Message-ID: <8a9b664c0909132302m6cf3ff91i4592cf7ad971a5d0@mail.gmail.com> Just want to say that you were right. Updating to trunk solved the problem. It seems that I updated/rebuilt my copy just before the patch was applied on 4 September. Thank you again! On Thu, Sep 10, 2009 at 9:54 AM, Vassil Panayotov wrote: > Michael, Moises and Octavio thank you for your replies! > > The server will be shipped to another site today and I can't test > thoroughly now. > When it is installed I will update this thread. > > Best regards, > Vassil > > On Thu, Sep 10, 2009 at 1:58 AM, Octavio Ruiz wrote: >> On Wed, Sep 9, 2009 at 01:20, Vassil Panayotov wrote: >>> Hi, >>> >>> Is it possible to originate calls from specific A500 ports with FreeSWITCH? >>> I am using a A504 (8 BRI interfaces), and I want some outbound calls to be >>> made from specific BRI interfaces. >> >> You can't define several spans in openzap.conf for boost, the >> sangoma_brid config file is where you define groups, so your config >> should look like this: >> >> /// smg_bri.conf >> ...... >> >> group=1 >> spans=1 >> >> group=2 >> spans=2 >> >> group=3 >> spans=3 >> >> ...... >> >> /// openzap.conf >> >> ?[span wanpipe BoostBRI] >> ?trunk_type => bri >> ?b-channel => 1:1-2 >> ?b-channel => 2:1-2 >> ?b-channel => 3:1-2 >> ?b-channel => 4:1-2 >> ?b-channel => 5:1-2 >> ?b-channel => 6:1-2 >> ?b-channel => 7:1-2 >> ?b-channel => 8:1-2 >> >> /// openzap.conf.xml >> >> ? >> ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? ? >> ? ? >> ? >> >> >> Then, you can Dial to your span/group number 3 with: >> >> freeswitch> ? ?originate openzap/1/a/12345 at g3 >> |&() >> freeswitch> ? ?originate openzap/1/a/12345 at G3 >> |&() >> freeswitch> ? ?originate openzap/1/a/12345 at r3 >> |&() >> freeswitch> ? ?originate openzap/1/a/12345 at R3 >> |&() >> >> >> If you are using FS 1.0.4, there is a bug, you can fix it with this >> -already in trunk- patch. >> >> Index: src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c >> =================================================================== >> --- libs/openzap/src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c.orig >> +++ libs/openzap/src/ozmod/ozmod_ss7_boost/ozmod_ss7_boost.c >> @@ -282,6 +282,8 @@ >> ? ? ? ?} >> >> ? ? ? ?ss7bc_call_init(&event, caller_data->cid_num.digits, ani, r); >> + ? ? ? //ss7_bc_call_init will clear the trunk_group val so we need to set it again >> + ? ? ? event.trunk_group=tg; >> >> ? ? ? ?if (gr && *(gr+1)) { >> >> Best regards, >> >> -- >> Octavio H. Ruiz Cervera >> Tel.: (+52 55) 8590-9000 Ext. 7016 >> Mobile: (+52 1 55) 4358-4565 >> Sent from Mexico City, DF, Mexico >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > From nandy1925 at gmail.com Sun Sep 13 23:20:14 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Mon, 14 Sep 2009 14:20:14 +0800 Subject: [Freeswitch-users] FS 1.0.4 erl configure error In-Reply-To: <4AAD1611.2030803@gmail.com> References: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> <4AAD1611.2030803@gmail.com> Message-ID: <7d0bfd8c0909132320j1b2d1113j78da26b07a37e526@mail.gmail.com> meftah, i disabled mod_erlang_event in modules.conf. unixodbc is installed already. still ... the same error message. tks for your input. /nandy On Sun, Sep 13, 2009 at 11:56 PM, Meftah Tayeb wrote: > hello, > i think you enabled mod_erlang_event in the modules.conf > install unixodbc if is not installed > thanks > > Nandy Dagondon a ?crit : > > hi, > > i want to enable odbc support which is required in mod_lcr feature. > however, i encounter ./configure problem after installing Erlang R13B01. > this is the portion of the error messages: > > ....... > checking for erl... /usr/local/bin/erl > checking erlang version... 5.7.2 > checking erlang libdir... /usr/local/lib/erlang/lib/erl_interface-3.6.2/lib > checking erlang incdir... > /usr/local/lib/erlang/lib/erl_interface-3.6.2/include > checking ei.h usability... yes > checking ei.h presence... no > configure: WARNING: ei.h: accepted by the compiler, rejected by the > preprocessor! > configure: WARNING: ei.h: proceeding with the compiler's result > checking for ei.h... yes > checking for ei_encode_version in -lei... yes > checking for ei_link_unlink in -lei... no > configure: Your erlang seems OK, do not forget to enable mod_erlang_event > in modules.conf > configure: creating ./config.status > config.status: creating src/include/switch_version.h.in > .infig.status: error: cannot find input file: Makefile > -------- END -------- > > i set ERL_TOP environment variable to the source directory. has anyone > encountered this problem? can anyone give me a hint what's wrong. i'm > compiling FS 1.0.4. > > thank you, > /nandy > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > __________ Information from ESET NOD32 Antivirus, version of virus signature database 4421 (20090913) __________ > > The message was checked by ESET NOD32 Antivirus. > http://www.eset.com > > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4421 (20090913) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/64fa95bd/attachment.html From ahmedmunir007 at gmail.com Mon Sep 14 00:18:14 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Mon, 14 Sep 2009 13:18:14 +0600 Subject: [Freeswitch-users] How to filter the allowed string Message-ID: Hi, I'm newbie in FS. I want to know how to Filter the string to include only the allowed characters in FS? Kindly advice me. -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/6571e2ca/attachment.html From juanbackson at gmail.com Mon Sep 14 01:24:19 2009 From: juanbackson at gmail.com (Juan Backson) Date: Mon, 14 Sep 2009 16:24:19 +0800 Subject: [Freeswitch-users] freeswitch on blackfin + uclinux Message-ID: <27c25bc40909140124o505c6c42r59b7ce1627b26540@mail.gmail.com> Hi, Does anyone have any luck on porting freeswitch to blackfin + uclinux? Is this a feasible option? jb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/a46677e7/attachment.html From hads at nice.net.nz Mon Sep 14 01:37:17 2009 From: hads at nice.net.nz (Hadley Rich) Date: Mon, 14 Sep 2009 20:37:17 +1200 Subject: [Freeswitch-users] freeswitch on blackfin + uclinux In-Reply-To: <27c25bc40909140124o505c6c42r59b7ce1627b26540@mail.gmail.com> References: <27c25bc40909140124o505c6c42r59b7ce1627b26540@mail.gmail.com> Message-ID: <200909142037.17261.hads@nice.net.nz> On Mon, 14 Sep 2009 20:24:19 Juan Backson wrote: > Does anyone have any luck on porting freeswitch to blackfin + uclinux? From memory there is an issue with APR so no. It would be sweet though :) -- https://nicegear.co.nz VoIP and Open Source Hardware From jingwei.yang at gmail.com Mon Sep 14 01:46:49 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Mon, 14 Sep 2009 16:46:49 +0800 Subject: [Freeswitch-users] compilation error with the latest codes Message-ID: <13529f9d0909140146r115db921l5a2f9bb92a582412@mail.gmail.com> Hi Folks, I've got a compilation error with the latest codes (r14842) Making all in packages Creating mod_sofia_la-mod_sofia.lo Compiling mod_sofia.c ... Creating mod_sofia_la-sofia.lo Compiling sofia.c ... sofia.c: In function ???sofia_handle_sip_r_invite???: sofia.c:3221: error: expected expression before ???< References: <13529f9d0909140146r115db921l5a2f9bb92a582412@mail.gmail.com> Message-ID: <4D169DC3-E8A4-44A4-816A-52857772FBFB@freeswitch.org> You have a merge conflict please svn revert sofia.c /b On Sep 14, 2009, at 3:46 AM, Jingwei Yang wrote: > Hi Folks, > > I've got a compilation error with the latest codes (r14842) > > Making all in packages > Creating mod_sofia_la-mod_sofia.lo > Compiling mod_sofia.c ... > Creating mod_sofia_la-sofia.lo > Compiling sofia.c ... > sofia.c: In function ???sofia_handle_sip_r_invite???: > sofia.c:3221: error: expected expression before ???< make[5]: *** [mod_sofia_la-sofia.lo] Error 1 > make[4]: *** [all] Error 2 > make[3]: *** [mod_sofia-all] Error 1 > make[2]: *** [all-recursive] Error 1 > > Does anyone have ideas about this? From morten.henckel at gmail.com Sun Sep 13 07:27:23 2009 From: morten.henckel at gmail.com (Morten Henckel) Date: Sun, 13 Sep 2009 16:27:23 +0200 Subject: [Freeswitch-users] Recording inbound call including DTMF - possible ? In-Reply-To: <25753ef00909130142w68b12c5pf44f4bc40f2376a@mail.gmail.com> References: <25753ef00909130142w68b12c5pf44f4bc40f2376a@mail.gmail.com> Message-ID: <25753ef00909130727geaa6002k80fbca40611d603@mail.gmail.com> Hi I need to measure DTM digits duration and interdigit delay for various phones in a two stage dialing scenario. I.e Phone dials DID and after answer then the second number My set-up is: Phone->PSTN network->DID(inband DTMF) ->FS I ha ve FS to answer the call and record the call - all this is fine. However when i analyse the rdecording the Digits are being cut off down to 10 msec "bursts" - I trust its FS that cust the DTMF in order to avoid further propogation inband to second leg of the call. Is theer a way to avoid this ? I.e record the inbound call without DTMF processing ? Thx Morten -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090913/c214a27d/attachment-0001.html From paul.degt at gmail.com Sun Sep 13 16:47:21 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Sun, 13 Sep 2009 19:47:21 -0400 Subject: [Freeswitch-users] Asterisk 1.6 connecting to FS 1.4 Message-ID: <4AAD8489.6000806@gmail.com> Hi, A client of ours is trying to connect his * to our FS, outgoing calls work fine, unfortunately when we try to forward an incoming call to his * it's not going through. I see his registration in our internal profile which looks just fine. We try to forward incoming calls using this in FS dialplan: Only abnormal things I can see in FS logs are: 2009-09-13 19:17:31.869158 [ERR] sofia_reg.c:1570 No Matching gateway found 2009-09-13 19:17:31.869158 [NOTICE] sofia_reg.c:1590 Hangup sofia/internal/4000000[CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] Why would FS look for a gateway in this case? And what MANDATORY_IE_MISSING would mean here? Call gets forwarded to VM as if user was unavailable.Hangup is initiated by us in this case. Client uses this configuration in *: /etc/asterisk/sip.conf: >>>>>> /etc/asterisk/sip.conf: >>>>>> register=>4000000:mysippassword at versafon.com/4000000 >>>>>> >>>>>> [4000000] >>>>>> type=friend >>>>>> username=4000000 >>>>>> secret=mysippassword >>>>>> host=versafon.com >>>>>> canreinvite=no >>>>>> fromuser=4000000 >>>>>> dtmfmode=rfc2833 >>>>>> context=versafon-incoming From freeswitch-users at dwayne-hubbard.com Sun Sep 13 17:35:29 2009 From: freeswitch-users at dwayne-hubbard.com (freeswitch-users at dwayne-hubbard.com) Date: Sun, 13 Sep 2009 21:35:29 -0300 Subject: [Freeswitch-users] Zaptel Analog FXO Outbound Dialing In-Reply-To: 91CDA5F8-D8DF-4864-A915-07A146599331@freeswitch.org Message-ID: <20090914003529.784c4fc4@neutrino.joshua-colp.com> _____ My suspicion is that this is only for zaptel type cards. Our tests with Sangoma analog cards have all been pretty successful. But thanks for info! Anyone else using Rhino, Digium, or compatible analog cards?I am not experiencing an audio delay. My configuration is exactly as documented on the Zaptel Tutorial wiki page (http://wiki.freeswitch.org/wiki/Zaptel_Tutorial). I'm using a Digium TDM400P, Zaptel 1.4 revision 4630, and FreeSWITCH trunk revision 14842. If you want me to try anything for you, I'm 'Deeewayne' on IRC. -Dwayne. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090913/ee1a02a0/attachment.html From woodydickson at gmail.com Mon Sep 14 06:08:56 2009 From: woodydickson at gmail.com (Woody Dickson) Date: Mon, 14 Sep 2009 21:08:56 +0800 Subject: [Freeswitch-users] problem with performance testing Message-ID: Hi, I tried to performance test freeswitch with media proxy thur fs. With 400 cps, I start to see 2000 channels remaining in Freeswitch, and then "no read codec" error starts to pop up. With only 1875 channels, how come freeswitch is complaining about no read codec? Also, I am using media_proxy = true, whey should it need a codec anyway? freeswitch at mycom.com> 2009-09-14 20:59:16.777675 [ERR] switch_core_io.c:118 sofia/external/12323232 at 192.168.1.116:5911 has no read codec. 2009-09-14 20:59:30.815547 [ERR] switch_core_io.c:118 sofia/external/ 12323232 at 192.168.1.116:5911 has no read codec. 2009-09-14 20:59:30.815547 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: [] 2009-09-14 20:59:45.349181 [ERR] switch_core_io.c:118 sofia/external/ 12323232 at 192.168.1.116:5911 has no read codec. 2009-09-14 20:59:45.350179 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: [] 2009-09-14 21:00:00.104559 [ERR] switch_core_io.c:118 sofia/external/ 12323232 at 192.168.1.116:5911 has no read codec. 2009-09-14 21:00:04.495545 [ERR] switch_core_io.c:118 sofia/external/ 12323232 at 192.168.1.116:5911 has no read codec. 2009-09-14 21:00:16.996438 [ERR] switch_core_io.c:118 sofia/external/ 12323232 at 192.168.1.116:5911 has no read codec. 2009-09-14 21:00:25.905617 [ERR] switch_core_io.c:118 sofia/external/ 12323232 at 192.168.1.116:5911 has no read codec. freeswitch at mycom.com> show channels count API CALL [show(channels count)] output: 1875 total. freeswitch at mycom.com> 2009-09-14 21:00:36.212767 [ERR] switch_core_io.c:118 sofia/external/12323232 at 192.168.1.116:5911 has no read codec. After I paused the traffic from sipp and when sipp finished, I still got a bunch of zombie channels that are in CONSUME_MEDIA stage. What is the cause of these zombies and how can I fix it? uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure 5f013ece-d8a0-4ee0-bce0-3a56c05dc225,outbound,2009-09-14 20:53:55,1252932835,sofia/external/12323232 at 192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 9347ea12-7ffb-4e96-8ee9-b2ac1ee57752,outbound,2009-09-14 20:53:57,1252932837,sofia/external/12323232 at 192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 5dc2f6f2-7d58-4582-b680-40bb3ed330ef,outbound,2009-09-14 20:54:02,1252932842,sofia/external/12323232 at 192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 66361571-b022-42b9-9507-87d1bfc01b03,outbound,2009-09-14 20:54:03,1252932843,sofia/external/12323232 at 192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 7a075ea5-7c01-4951-b826-cf982df03501,outbound,2009-09-14 20:54:03,1252932843,sofia/external/12323232 at 192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, bd8c9905-dbab-4ab7-a9af-b7e2b87cc0b0,outbound,2009-09-14 20:54:05,1252932845,sofia/external/12323232 at 192.168.1.116:5911 ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 ,,,XML,default,PROXY,8000,PROXY,8000, 3af84a6c-02f2-44ae-8a4b-1c7940522005,outbound,2009-09-14 20:54:06,1252932846,sofia/external -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/c12d722a/attachment.html From anthony.minessale at gmail.com Mon Sep 14 06:14:49 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Sep 2009 08:14:49 -0500 Subject: [Freeswitch-users] problem with performance testing In-Reply-To: References: Message-ID: <191c3a030909140614q7263fc0ck3d972ea91b530cf4@mail.gmail.com> >> After I paused the traffic from sipp and when sipp finished, I still got a bunch of zombie channels that are in CONSUME_MEDIA stage. What is the cause of >> these zombies and how can I fix it? One way might be to not DDoS your box at 400cps? (You are out of rtp ports *and* you are pushing your machine too hard.) Only 1875 channels? hmm..... *shrug* On Mon, Sep 14, 2009 at 8:08 AM, Woody Dickson wrote: > Hi, > > I tried to performance test freeswitch with media proxy thur fs. With 400 > cps, I start to see 2000 channels remaining in Freeswitch, and then "no read > codec" error starts to pop up. With only 1875 channels, how come freeswitch > is complaining about no read codec? Also, I am using media_proxy = true, > whey should it need a codec anyway? > > > freeswitch at mycom.com> 2009-09-14 20:59:16.777675 [ERR] > switch_core_io.c:118 sofia/external/12323232 at 192.168.1.116:5911 has no > read codec. > 2009-09-14 20:59:30.815547 [ERR] switch_core_io.c:118 sofia/external/ > 12323232 at 192.168.1.116:5911 has no read codec. > 2009-09-14 20:59:30.815547 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: > [] > 2009-09-14 20:59:45.349181 [ERR] switch_core_io.c:118 sofia/external/ > 12323232 at 192.168.1.116:5911 has no read codec. > 2009-09-14 20:59:45.350179 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: > [] > 2009-09-14 21:00:00.104559 [ERR] switch_core_io.c:118 sofia/external/ > 12323232 at 192.168.1.116:5911 has no read codec. > 2009-09-14 21:00:04.495545 [ERR] switch_core_io.c:118 sofia/external/ > 12323232 at 192.168.1.116:5911 has no read codec. > 2009-09-14 21:00:16.996438 [ERR] switch_core_io.c:118 sofia/external/ > 12323232 at 192.168.1.116:5911 has no read codec. > 2009-09-14 21:00:25.905617 [ERR] switch_core_io.c:118 sofia/external/ > 12323232 at 192.168.1.116:5911 has no read codec. > > freeswitch at mycom.com> show channels count > API CALL [show(channels count)] output: > > 1875 total. > > freeswitch at mycom.com> 2009-09-14 21:00:36.212767 [ERR] > switch_core_io.c:118 sofia/external/12323232 at 192.168.1.116:5911 has no > read codec. > > > After I paused the traffic from sipp and when sipp finished, I still got a > bunch of zombie channels that are in CONSUME_MEDIA stage. What is the cause > of these zombies and how can I fix it? > > > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure > 5f013ece-d8a0-4ee0-bce0-3a56c05dc225,outbound,2009-09-14 > 20:53:55,1252932835,sofia/external/12323232 at 192.168.1.116:5911 > ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 > ,,,XML,default,PROXY,8000,PROXY,8000, > 9347ea12-7ffb-4e96-8ee9-b2ac1ee57752,outbound,2009-09-14 > 20:53:57,1252932837,sofia/external/12323232 at 192.168.1.116:5911 > ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 > ,,,XML,default,PROXY,8000,PROXY,8000, > 5dc2f6f2-7d58-4582-b680-40bb3ed330ef,outbound,2009-09-14 > 20:54:02,1252932842,sofia/external/12323232 at 192.168.1.116:5911 > ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 > ,,,XML,default,PROXY,8000,PROXY,8000, > 66361571-b022-42b9-9507-87d1bfc01b03,outbound,2009-09-14 > 20:54:03,1252932843,sofia/external/12323232 at 192.168.1.116:5911 > ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 > ,,,XML,default,PROXY,8000,PROXY,8000, > 7a075ea5-7c01-4951-b826-cf982df03501,outbound,2009-09-14 > 20:54:03,1252932843,sofia/external/12323232 at 192.168.1.116:5911 > ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 > ,,,XML,default,PROXY,8000,PROXY,8000, > bd8c9905-dbab-4ab7-a9af-b7e2b87cc0b0,outbound,2009-09-14 > 20:54:05,1252932845,sofia/external/12323232 at 192.168.1.116:5911 > ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 > ,,,XML,default,PROXY,8000,PROXY,8000, > 3af84a6c-02f2-44ae-8a4b-1c7940522005,outbound,2009-09-14 > 20:54:06,1252932846,sofia/external > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/854f96be/attachment.html From kjv at ken-ton.com Mon Sep 14 06:15:26 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Mon, 14 Sep 2009 09:15:26 -0400 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: <8A58CD3E-F0C3-4BAD-8DB9-3C9D3A22A70F@ken-ton.com> References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> <8A58CD3E-F0C3-4BAD-8DB9-3C9D3A22A70F@ken-ton.com> Message-ID: Swapping hardware... I've noticed other "odd" things... Things that shouldn't happen, do...... But not consistently.... The phrase, "It's computing Jim, but not as we know it..." pretty much describes the situation. Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Sep 13, 2009, at 6:30 PM, Karl Vesterling wrote: > New development. > > Even though the initial registration succeeds, the subsequent > registrations fail... > > ??Search me?? But that's just too weird for me... > > > > Best Regards, > Karl J. Vesterling > kjv at ken-ton.com > 202-461-3231 x0 > > On Sep 13, 2009, at 4:23 PM, Brian West wrote: > >> I haven't seen this issue in 8.12 either... Maybe thats why 8.11 >> isn't on the website last I checked? >> >> /b >> >> On Sep 13, 2009, at 2:59 PM, Karl Vesterling wrote: >> >>> RESOLVED!!! >>> >>> Folks, evidently this is a problem with Cisco Firmware P0S3-08-11-00 >>> >>> I forgot that (a long long time ago) I had dropped that firmware >>> into >>> that site. >>> Phones hadn't been rebooted in (a while)... >>> >>> Oddly enough, once you get past (X) number of phones, the >>> registration >>> chatter created by the bug was too much for FS to keep up with. >>> >>> P0S3-08-8-00 works perfectly fine. >>> >>> >>> Best Regards, >>> Karl J. Vesterling >>> kjv at ken-ton.com >>> 202-461-3231 x0 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/ab7eb04f/attachment-0001.html From anthony.minessale at gmail.com Mon Sep 14 06:17:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 14 Sep 2009 08:17:58 -0500 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> <8A58CD3E-F0C3-4BAD-8DB9-3C9D3A22A70F@ken-ton.com> Message-ID: <191c3a030909140617v396063e7n86cd8fdb39d75fa9@mail.gmail.com> The first hint was when the firmware rev began with the letters POS On Mon, Sep 14, 2009 at 8:15 AM, Karl Vesterling wrote: > Swapping hardware... I've noticed other "odd" things... Things that > shouldn't happen, do...... But not consistently.... The phrase, "It's > computing Jim, but not as we know it..." pretty much describes the > situation. > > Best Regards, > Karl J. Vesterling > kjv at ken-ton.com > 202-461-3231 x0 > > On Sep 13, 2009, at 6:30 PM, Karl Vesterling wrote: > > New development. > Even though the initial registration succeeds, the subsequent registrations > fail... > > ??Search me?? But that's just too weird for me... > > > > Best Regards, > Karl J. Vesterling > kjv at ken-ton.com > 202-461-3231 x0 > > On Sep 13, 2009, at 4:23 PM, Brian West wrote: > > I haven't seen this issue in 8.12 either... Maybe thats why 8.11 isn't on > the website last I checked? > /b > > On Sep 13, 2009, at 2:59 PM, Karl Vesterling wrote: > > RESOLVED!!! > > Folks, evidently this is a problem with Cisco Firmware P0S3-08-11-00 > > I forgot that (a long long time ago) I had dropped that firmware into > that site. > Phones hadn't been rebooted in (a while)... > > Oddly enough, once you get past (X) number of phones, the registration > chatter created by the bug was too much for FS to keep up with. > > P0S3-08-8-00 works perfectly fine. > > > Best Regards, > Karl J. Vesterling > kjv at ken-ton.com > 202-461-3231 x0 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/4cc3594d/attachment.html From brian at freeswitch.org Mon Sep 14 06:20:07 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Sep 2009 08:20:07 -0500 Subject: [Freeswitch-users] Asterisk 1.6 connecting to FS 1.4 In-Reply-To: <4AAD8489.6000806@gmail.com> References: <4AAD8489.6000806@gmail.com> Message-ID: <47524061-43E2-4699-AE97-4EED7346573C@freeswitch.org> This means the far end is sending you a challenge and we do not know how to answer it... please review how to setup a gateway on the Wiki so you can authenticate. /b On Sep 13, 2009, at 6:47 PM, paul.degt at gmail.com wrote: > Only abnormal things I can see in FS logs are: > 2009-09-13 19:17:31.869158 [ERR] sofia_reg.c:1570 No Matching > gateway found > 2009-09-13 19:17:31.869158 [NOTICE] sofia_reg.c:1590 Hangup > sofia/internal/4000000[CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] From brian at freeswitch.org Mon Sep 14 06:22:18 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Sep 2009 08:22:18 -0500 Subject: [Freeswitch-users] Recording inbound call including DTMF - possible ? In-Reply-To: <25753ef00909130727geaa6002k80fbca40611d603@mail.gmail.com> References: <25753ef00909130142w68b12c5pf44f4bc40f2376a@mail.gmail.com> <25753ef00909130727geaa6002k80fbca40611d603@mail.gmail.com> Message-ID: <35F845AB-B132-40D3-8B48-0A04C73C93AC@freeswitch.org> On Sep 13, 2009, at 9:27 AM, Morten Henckel wrote: > However when i analyse the rdecording the Digits are being cut off > down to 10 msec "bursts" - I trust its FS that cust the DTMF in > order to avoid further propogation inband to second leg of the call. Nope if its rfc2833 its not us cutting the dtmf if you have a TDM gateway in the mix that is prob. what is doing it. > > Is theer a way to avoid this ? I.e record the inbound call without > DTMF processing ? No. The blib of DTMF you hear is not ours to remove its the remote gateways job... FreeSWTICH usually only does 2833 so it could be hearing the little bit of DTMF coming in or out from the endpoints. If you check the cdr hangup you'll have digit_log which will be a log of all digits dialed during the call. > > Thx > > Morten > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/5c770945/attachment.html From brian at freeswitch.org Mon Sep 14 06:22:41 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Sep 2009 08:22:41 -0500 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: <191c3a030909140617v396063e7n86cd8fdb39d75fa9@mail.gmail.com> References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> <8A58CD3E-F0C3-4BAD-8DB9-3C9D3A22A70F@ken-ton.com> <191c3a030909140617v396063e7n86cd8fdb39d75fa9@mail.gmail.com> Message-ID: HAHA I couldn't have said this better! /b On Sep 14, 2009, at 8:17 AM, Anthony Minessale wrote: > The first hint was when the firmware rev began with the letters POS From jmesquita at freeswitch.org Mon Sep 14 06:37:27 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 14 Sep 2009 10:37:27 -0300 Subject: [Freeswitch-users] How to filter the allowed string In-Reply-To: References: Message-ID: Not sure I understand what you mean. Can you explain what you are trying to achieve a little bit better? jmesquita On Mon, Sep 14, 2009 at 4:18 AM, Ahmed Munir wrote: > Hi, > I'm newbie in FS. I want to know how to Filter the string to include only > the allowed characters in FS? > > Kindly advice me. > -- > Regards, > > Ahmed Munir > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/f5a788f9/attachment.html From tculjaga at gmail.com Mon Sep 14 06:47:15 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 14 Sep 2009 15:47:15 +0200 Subject: [Freeswitch-users] problem with performance testing In-Reply-To: References: Message-ID: <65d96fc80909140647m5bc88901wde3e32acb1309caf@mail.gmail.com> Hi Woody, well, it is quite hard to answer you back with this logs... you didn't tell us: 1. what machine are you running (CPU/RAM) 2. what distro are you running - 32 or 64 bit (i had some lets say "experience" with a wrong selection :P) 3. what is your configuration (dialplan/sip_profiles) 4. did you disable all logging? 5. what modules are loaded (you should load minimal modules - at least disable conference) 6. if you moved the db files to a ram disk 7. how do you start the calls (slowly with 10 - 20 CPS or you are DDoS-ing it with 400 right away). 8. how long do you keep the calls going? 9. what is the the current CPU usage when stresstesting 10. what is the amount of read/writes to from/to your HDD ... i could go on and on with the list. Here Anthony has nothing to tell you except that you reached the maximum and you are currently killing the machine/application Also, what are you looking for? ... A machine that can do a lot of simultaneous calls or a machine that can do a lot of CPS? you should decide at some point. I have a lot of experience with commercial SoftSwitches and i can tell you that FreeSWITCH performance is something outstanding ... i'm getting a reliable 500 CPS on just one FS machine (dualcore xeon 2.33 GHz - bogomips 4670). just to compare: NetCentrex as a comercial SoftSwitch can do only 480 CPS (distribuited on 10 nodes)... and of course signaling only. And this is enough to run my 12.000 calls. Tihomir. On Mon, Sep 14, 2009 at 3:08 PM, Woody Dickson wrote: > Hi, > > I tried to performance test freeswitch with media proxy thur fs. With 400 > cps, I start to see 2000 channels remaining in Freeswitch, and then "no read > codec" error starts to pop up. With only 1875 channels, how come freeswitch > is complaining about no read codec? Also, I am using media_proxy = true, > whey should it need a codec anyway? > > > freeswitch at mycom.com> 2009-09-14 20:59:16.777675 [ERR] > switch_core_io.c:118 sofia/external/12323232 at 192.168.1.116:5911 has no > read codec. > 2009-09-14 20:59:30.815547 [ERR] switch_core_io.c:118 sofia/external/ > 12323232 at 192.168.1.116:5911 has no read codec. > 2009-09-14 20:59:30.815547 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: > [] > 2009-09-14 20:59:45.349181 [ERR] switch_core_io.c:118 sofia/external/ > 12323232 at 192.168.1.116:5911 has no read codec. > 2009-09-14 20:59:45.350179 [ERR] sofia_glue.c:2566 AUDIO RTP REPORTS ERROR: > [] > 2009-09-14 21:00:00.104559 [ERR] switch_core_io.c:118 sofia/external/ > 12323232 at 192.168.1.116:5911 has no read codec. > 2009-09-14 21:00:04.495545 [ERR] switch_core_io.c:118 sofia/external/ > 12323232 at 192.168.1.116:5911 has no read codec. > 2009-09-14 21:00:16.996438 [ERR] switch_core_io.c:118 sofia/external/ > 12323232 at 192.168.1.116:5911 has no read codec. > 2009-09-14 21:00:25.905617 [ERR] switch_core_io.c:118 sofia/external/ > 12323232 at 192.168.1.116:5911 has no read codec. > > freeswitch at mycom.com> show channels count > API CALL [show(channels count)] output: > > 1875 total. > > freeswitch at mycom.com> 2009-09-14 21:00:36.212767 [ERR] > switch_core_io.c:118 sofia/external/12323232 at 192.168.1.116:5911 has no > read codec. > > > After I paused the traffic from sipp and when sipp finished, I still got a > bunch of zombie channels that are in CONSUME_MEDIA stage. What is the cause > of these zombies and how can I fix it? > > > > uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure > 5f013ece-d8a0-4ee0-bce0-3a56c05dc225,outbound,2009-09-14 > 20:53:55,1252932835,sofia/external/12323232 at 192.168.1.116:5911 > ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 > ,,,XML,default,PROXY,8000,PROXY,8000, > 9347ea12-7ffb-4e96-8ee9-b2ac1ee57752,outbound,2009-09-14 > 20:53:57,1252932837,sofia/external/12323232 at 192.168.1.116:5911 > ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 > ,,,XML,default,PROXY,8000,PROXY,8000, > 5dc2f6f2-7d58-4582-b680-40bb3ed330ef,outbound,2009-09-14 > 20:54:02,1252932842,sofia/external/12323232 at 192.168.1.116:5911 > ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 > ,,,XML,default,PROXY,8000,PROXY,8000, > 66361571-b022-42b9-9507-87d1bfc01b03,outbound,2009-09-14 > 20:54:03,1252932843,sofia/external/12323232 at 192.168.1.116:5911 > ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 > ,,,XML,default,PROXY,8000,PROXY,8000, > 7a075ea5-7c01-4951-b826-cf982df03501,outbound,2009-09-14 > 20:54:03,1252932843,sofia/external/12323232 at 192.168.1.116:5911 > ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 > ,,,XML,default,PROXY,8000,PROXY,8000, > bd8c9905-dbab-4ab7-a9af-b7e2b87cc0b0,outbound,2009-09-14 > 20:54:05,1252932845,sofia/external/12323232 at 192.168.1.116:5911 > ,CS_CONSUME_MEDIA,sipp,sipp,192.168.1.116,12323232 at 192.168.1.116:5911 > ,,,XML,default,PROXY,8000,PROXY,8000, > 3af84a6c-02f2-44ae-8a4b-1c7940522005,outbound,2009-09-14 > 20:54:06,1252932846,sofia/external > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/19b86564/attachment-0001.html From tculjaga at gmail.com Mon Sep 14 06:58:45 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 14 Sep 2009 15:58:45 +0200 Subject: [Freeswitch-users] FS create directory Message-ID: <65d96fc80909140658i51a9f053if97b04c8c5c517bb@mail.gmail.com> Hi, i just have a maybe dummy question but .... it is still a question :P ** in my case ${service_instance} is something dynamic and has to be created on the fly. Is there any way FS can create a directory prior to dump the file there? Tihomir. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/0fb2d65b/attachment.html From brian at freeswitch.org Mon Sep 14 07:13:16 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Sep 2009 09:13:16 -0500 Subject: [Freeswitch-users] FS create directory In-Reply-To: <65d96fc80909140658i51a9f053if97b04c8c5c517bb@mail.gmail.com> References: <65d96fc80909140658i51a9f053if97b04c8c5c517bb@mail.gmail.com> Message-ID: <5A7780B8-5B35-40A2-8148-807A62F44EFA@freeswitch.org> Run system(); from the dialplan... /b On Sep 14, 2009, at 8:58 AM, Tihomir Culjaga wrote: > Hi, > > i just have a maybe dummy question but .... it is still a question :P > > > > in my case ${service_instance} is something dynamic and has to be > created on the fly. > > Is there any way FS can create a directory prior to dump the file > there? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/74c5734a/attachment.html From mustafa.pk at gmail.com Mon Sep 14 07:03:56 2009 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Mon, 14 Sep 2009 20:03:56 +0600 Subject: [Freeswitch-users] How to filter the allowed string In-Reply-To: References: Message-ID: <4AAE4D4C.1080906@gmail.com> Ahmed, if you are talking about dial patterns then yes, freeswitch takes you a mile ahead and utilizes regular expressions for pattern matching, you could probably use something like this: "^([0-9]+)$" above simple regex will allow any digit from 0 to 9 and + indicates repetitive, so this regex is equal to following asterisk's pattern: "_X." -gm Jo?o Mesquita wrote: > Not sure I understand what you mean. Can you explain what you are > trying to achieve a little bit better? > > jmesquita > > On Mon, Sep 14, 2009 at 4:18 AM, Ahmed Munir > wrote: > > Hi, > > I'm newbie in FS. I want to know how to Filter the string to > include only the allowed characters in FS? > > Kindly advice me. > -- > Regards, > > Ahmed Munir > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From leon at scarlet-internet.nl Mon Sep 14 07:15:40 2009 From: leon at scarlet-internet.nl (Leon de Rooij) Date: Mon, 14 Sep 2009 16:15:40 +0200 Subject: [Freeswitch-users] FS create directory In-Reply-To: <65d96fc80909140658i51a9f053if97b04c8c5c517bb@mail.gmail.com> References: <65d96fc80909140658i51a9f053if97b04c8c5c517bb@mail.gmail.com> Message-ID: Hi, You could use a system call for that: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system regards, Leon On Sep 14, 2009, at 3:58 PM, Tihomir Culjaga wrote: > Hi, > > i just have a maybe dummy question but .... it is still a question :P > > > > in my case ${service_instance} is something dynamic and has to be > created on the fly. > > Is there any way FS can create a directory prior to dump the file > there? > > > Tihomir. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/3e320c24/attachment.html From paul.degt at gmail.com Mon Sep 14 07:20:04 2009 From: paul.degt at gmail.com (paul.degt at gmail.com) Date: Mon, 14 Sep 2009 10:20:04 -0400 (EDT) Subject: [Freeswitch-users] Asterisk 1.6 connecting to FS 1.4 In-Reply-To: Message-ID: <186671531.5.1252938004275.JavaMail.root@u15346123.onlinehome-server.com> Thank you for the hint. But.. why would I need a gateway in this case? I am just trying to ring an FS extension, right? Anybody has a clue how to make * not to send the challenge? >This means the far end is sending you a challenge and we do not know >how to answer it... please review how to setup a gateway on the Wiki >so you can authenticate. > >/b From tculjaga at gmail.com Mon Sep 14 07:23:47 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 14 Sep 2009 16:23:47 +0200 Subject: [Freeswitch-users] FS create directory In-Reply-To: References: <65d96fc80909140658i51a9f053if97b04c8c5c517bb@mail.gmail.com> Message-ID: <65d96fc80909140723ja3baa23gdfdf105f54b116ac@mail.gmail.com> yep, just sow it in the meantime... thanks. btw: can i use mod_shout to stream files to a server.. e.g. ** can it work? T. On Mon, Sep 14, 2009 at 4:15 PM, Leon de Rooij wrote: > Hi, > You could use a system call for that: > > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_system > > regards, > > Leon > > > On Sep 14, 2009, at 3:58 PM, Tihomir Culjaga wrote: > > Hi, > > i just have a maybe dummy question but .... it is still a question :P > > * data="${recordpath}/${service_instance}/${record_filename} 20 200"/>* > > in my case ${service_instance} is something dynamic and has to be created > on the fly. > > Is there any way FS can create a directory prior to dump the file there? > > > Tihomir. > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/1545e12d/attachment.html From zolotov at altron.ua Mon Sep 14 07:41:49 2009 From: zolotov at altron.ua (Evgeniy Zolotov) Date: Mon, 14 Sep 2009 17:41:49 +0300 Subject: [Freeswitch-users] FS create directory References: <65d96fc80909140658i51a9f053if97b04c8c5c517bb@mail.gmail.com> Message-ID: <00fb01ca3549$80049870$7602a8c0@opos20> This works for me: You must set ' filebase_dir ' before. ----- Original Message ----- From: Tihomir Culjaga To: freeswitch-users at lists.freeswitch.org Sent: Monday, September 14, 2009 4:58 PM Subject: [Freeswitch-users] FS create directory Hi, i just have a maybe dummy question but .... it is still a question :P in my case ${service_instance} is something dynamic and has to be created on the fly. Is there any way FS can create a directory prior to dump the file there? Tihomir. ------------------------------------------------------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/7f7fd1e1/attachment.html From tculjaga at gmail.com Mon Sep 14 08:02:40 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 14 Sep 2009 17:02:40 +0200 Subject: [Freeswitch-users] FS create directory In-Reply-To: <00fb01ca3549$80049870$7602a8c0@opos20> References: <65d96fc80909140658i51a9f053if97b04c8c5c517bb@mail.gmail.com> <00fb01ca3549$80049870$7602a8c0@opos20> Message-ID: <65d96fc80909140802l7e51c9bbv38b4b20903960979@mail.gmail.com> nice ... thx. T. On Mon, Sep 14, 2009 at 4:41 PM, Evgeniy Zolotov wrote: > This works for me: > > > > You must set ' filebase_dir ' before. > > ----- Original Message ----- > *From:* Tihomir Culjaga > *To:* freeswitch-users at lists.freeswitch.org > *Sent:* Monday, September 14, 2009 4:58 PM > *Subject:* [Freeswitch-users] FS create directory > > Hi, > > i just have a maybe dummy question but .... it is still a question :P > > * data="${recordpath}/${service_instance}/${record_filename} 20 200"/>* > > in my case ${service_instance} is something dynamic and has to be created > on the fly. > > Is there any way FS can create a directory prior to dump the file there? > > > Tihomir. > > ------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/72b1c908/attachment-0001.html From jerry.richards at teotech.com Mon Sep 14 08:13:15 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 14 Sep 2009 08:13:15 -0700 Subject: [Freeswitch-users] Pastebin Username/Password Not Accepted Message-ID: <1475D6AA3FE84CA9A30531E3CAC3910D@greyhawk.tonecommander.com> What account do I need to create to post logs in the Pastebin? I tried my mailing list username/password, and also tried a jira.freeswitch.org username/password. Neither of these were accepted. Best Regards, Jerry From anatoliy at kounitskiy.com Mon Sep 14 08:20:43 2009 From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy) Date: Mon, 14 Sep 2009 18:20:43 +0300 Subject: [Freeswitch-users] Pastebin Username/Password Not Accepted In-Reply-To: <1475D6AA3FE84CA9A30531E3CAC3910D@greyhawk.tonecommander.com> References: <1475D6AA3FE84CA9A30531E3CAC3910D@greyhawk.tonecommander.com> Message-ID: <4AAE5F4B.9090902@kounitskiy.com> Try username "pastebin" with pasword "freeswitch" (without ") Jerry Richards wrote: > What account do I need to create to post logs in the Pastebin? I tried my > mailing list username/password, and also tried a jira.freeswitch.org > username/password. Neither of these were accepted. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jerry.richards at teotech.com Mon Sep 14 08:29:22 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 14 Sep 2009 08:29:22 -0700 Subject: [Freeswitch-users] Pastebin Username/Password Not Accepted Message-ID: Aha... I have been notified that I failed the test. The username/password is given in the authentication pop-up itself. My bad... -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Monday, September 14, 2009 8:13 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: Pastebin Username/Password Not Accepted What account do I need to create to post logs in the Pastebin? I tried my mailing list username/password, and also tried a jira.freeswitch.org username/password. Neither of these were accepted. Best Regards, Jerry From pjintheusa at gmail.com Mon Sep 14 09:17:40 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Mon, 14 Sep 2009 12:17:40 -0400 Subject: [Freeswitch-users] Pastebin Username/Password Not Accepted In-Reply-To: References: Message-ID: <367751820909140917i20ef725u34559dd8736d1cb6@mail.gmail.com> it's a rite of passage :) On Mon, Sep 14, 2009 at 11:29 AM, Jerry Richards wrote: > > Aha... I have been notified that I failed the test. The username/password > is given in the authentication pop-up itself. My bad... > > > > -----Original Message----- > From: Jerry Richards [mailto:jerry.richards at teotech.com] > Sent: Monday, September 14, 2009 8:13 AM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: Pastebin Username/Password Not Accepted > > What account do I need to create to post logs in the Pastebin? I tried my > mailing list username/password, and also tried a jira.freeswitch.org > username/password. Neither of these were accepted. > > Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/b982a322/attachment.html From msc at freeswitch.org Mon Sep 14 09:33:33 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 14 Sep 2009 09:33:33 -0700 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects Message-ID: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> Hello FreeSWITCHers! We are looking for people who are in a position to help out with various subprojects that will help FreeSWITCH to keep growing. We need people to help out in these basic areas: Bug marshals (people who watch JIRA and test bug reports, patches, etc.) Documentation maintainers (people who update the wiki when new stuff comes out, also those familiar with mediawiki administration) Documentation authors (people who write new docs, how-to's, tutorials, examples, etc.) Package maintainers (people who manage Debian debs, RPMs, etc.) Additionally, we are always looking for more folks to assist with answering questions on IRC and the mailing list. It is definitely nice to have people who've gone through the pains of switching to FreeSWITCH (or learning it from scratch) who can assist the steady stream of new users. If you want to help and aren't sure where to go from here then please at least do the following: #1 - Join #freeswitch on irc.freenode.net and hang out as much as possible #2 - Check the recent changes link on wiki.freeswitch.org each day #3 - Join the Friday public conference call and listen in These three things, in addition to the mailing list, will keep you well in tune with the FreeSWITCH community and what's happening. Next, make a note of the parts of FS that you use frequently, know a lot about, or are particularly passionate about. Those are the items we'd love to have you help us with. For example: if you use mod_xml_curl frequently and have been through the set up process then you're a prime candidate to help answer questions, refine the mod_xml_curl wiki documentation, write up a tutorial, contribute a working example of a web server & database schema, etc. If you are good with a scripting language then we could definitely use help with rounding out the docs for your favorite language. We could also use code samples, so ask for a contrib folder if you have things you would like to share. Or how about this: you read something on the wiki, it doesn't quite work when you try, so you tinker until you figure it out. Now you're in a position to update the wiki for everyone else's benefit, too. As you can see, you don't have to be a FreeSWITCH expert before you can help the project. What we really need are people who care about the project and want to see it flourish. If you are such a person then please contact me off list. Tell me what you're good at or where you would like to help. Many thanks for all of your support! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/9f4874f7/attachment.html From diego.viola at gmail.com Mon Sep 14 10:04:38 2009 From: diego.viola at gmail.com (Diego Viola) Date: Mon, 14 Sep 2009 17:04:38 +0000 Subject: [Freeswitch-users] Asterisk 1.6 connecting to FS 1.4 In-Reply-To: <186671531.5.1252938004275.JavaMail.root@u15346123.onlinehome-server.com> References: <186671531.5.1252938004275.JavaMail.root@u15346123.onlinehome-server.com> Message-ID: <86a32abc0909141004kb6849f4w58ba431f429f8a3c@mail.gmail.com> There is no such things as FS 1.4, but 1.0.4 yes. On Mon, Sep 14, 2009 at 2:20 PM, wrote: > Thank you for the hint. > But.. why would I need a gateway in this case? I am just trying to ring an > FS extension, right? > Anybody has a clue how to make * not to send the challenge? > > > > >This means the far end is sending you a challenge and we do not know > >how to answer it... please review how to setup a gateway on the Wiki > >so you can authenticate. > > > >/b > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/eec158c3/attachment.html From mike at jerris.com Mon Sep 14 10:27:51 2009 From: mike at jerris.com (Michael Jerris) Date: Mon, 14 Sep 2009 13:27:51 -0400 Subject: [Freeswitch-users] 482 Request merged, in serial forking In-Reply-To: References: Message-ID: We currently don't support forked dialogs. Mike On Sep 8, 2009, at 12:16 PM, Humberto Quintana wrote: > Hi Brian, > > Thank you very much for your answer but both, Freeswitch and > Kamailio have public IPs, it's my NAT'd IP phone who has private IP > but this is fixed by Kamailio. > > The problem is not the 1st call is failing ( the test is set that > way), the problem is FS answers back 482 when Kamailio tries a 2nd > route ( or 3rd ) for the same call... > > > Freeswitch is configured to use the Requested-URI sent by Kamailio: > > > > > I noticed that there is no Log message in Freeswitch when receiving > the INVITE for the 2nd route. > The process in FS seems to be destroyed (11:46:21.396593) before the > 2nd INVITE is received (11:46:21.401419 > ). > > > U 2009/09/08 11:46:21.395702 freeswitch:5060 -> kamailio:5060 > SIP/2.0 503 Service Unavailable. > Call-ID: ba748cd27cd163b5 at 192.168.2.13 > > U 2009/09/08 11:46:21.395897 kamailio:5060 -> freeswitch:5060 > ACK sip:5145555555 at gw1:5060 SIP/2.0. > Call-ID: ba748cd27cd163b5 at 192.168.2.13 > > U 2009/09/08 11:46:21.401419 kamailio:5060 -> freeswitch:5060 > INVITE sip:15145555555 at gw2:5061 SIP/2.0. > Call-ID: ba748cd27cd163b5 at 192.168.2.13 > > U 2009/09/08 11:46:21.401845 freeswitch:5060 -> kamailio:5060 > SIP/2.0 482 Request merged. > Call-ID: ba748cd27cd163b5 at 192.168.2.13 > > > 2009-09-08 11:46:21.395503 [DEBUG] mod_sofia.c:417 Responding to > INVITE with: 503 > 2009-09-08 11:46:21.395503 [DEBUG] switch_core_state_machine.c:46 > sofia/external/10092020 at freeswitch Standard HANGUP, cause: > NORMAL_TEMPORARY_FAILURE > 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:434 > (sofia/external/10092020 at freeswitch) State HANGUP going to sleep > 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:476 > (sofia/external/10092020 at freeswitch) State Change CS_HANGUP -> > CS_REPORTING > 2009-09-08 11:46:21.396593 [DEBUG] switch_core_session.c:932 Send > signal sofia/external/10092020 at freeswitch [BREAK] > 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:398 > (sofia/external/10092020 at freeswitch) Running State Change CS_REPORTING > 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:612 > (sofia/external/10092020 at freeswitch) State REPORTING > 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:53 > sofia/external/10092020 at freeswitch Standard REPORTING, cause: > NORMAL_TEMPORARY_FAILURE > 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:612 > (sofia/external/10092020 at freeswitch) State REPORTING going to sleep > 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:411 > (sofia/external/10092020 at freeswitch) State Change CS_REPORTING -> > CS_DESTROY > 2009-09-08 11:46:21.396593 [DEBUG] switch_core_session.c:1068 > Session 3 (sofia/external/10092020 at freeswitch) Locked, Waiting on > external entities > 2009-09-08 11:46:21.396593 [NOTICE] switch_core_session.c:1086 > Session 3 (sofia/external/10092020 at freeswitch) Ended > 2009-09-08 11:46:21.396593 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/external/10092020 at freeswitch [CS_DESTROY] > 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:564 > (sofia/external/10092020 at freeswitch) State DESTROY > 2009-09-08 11:46:21.396593 [DEBUG] mod_sofia.c:255 sofia/external/ > 10092020 at freeswitch SOFIA DESTROY > 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:60 > sofia/external/10092020 at freeswitch Standard DESTROY > 2009-09-08 11:46:21.396593 [DEBUG] switch_core_state_machine.c:564 > (sofia/external/10092020 at freeswitch) State DESTROY going to sleep > > > Note: I'm using only the external sofia profile. > > > Thanks, > > Humberto > > > > > > > > > > > > ========================================== > Looks like FS is behind nat. You need to set local-network-acl and > the ext-rtp-ip and ext-sip-ip so FreeSWITCH properly puts in the right > IP's in the via headers and sdp. > > Please refer to internal.xml in the latest SVN for an example of how > to do this. > > /b > > New! Open Hotmail faster on the new MSN homepage! > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/94782edb/attachment-0001.html From aep.lists at it46.se Mon Sep 14 10:28:49 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Mon, 14 Sep 2009 19:28:49 +0200 Subject: [Freeswitch-users] DTMF CSeq: 505 INFO vs RTP DTMF Message-ID: <9ece78af3aedde36e10768727ab099cd.squirrel@correo.nodo50.org> Hi, I am using the function session.collectInput and session.streamFile to collect a number of DTMF digits. If the DTMF digits are sent in the RTP, i can collect several digits until timeout. No problem there! If the DTMFs are received as a sequence of SIP INFO packages, collectInput only receives the first one. Any ideas? -- Stopping junk mailers is good for the environment From jerry.richards at teotech.com Mon Sep 14 11:38:54 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 14 Sep 2009 11:38:54 -0700 Subject: [Freeswitch-users] Inbound Gateway Call Not Working Message-ID: <1FAF1147FDC04A5BB524E98666BE02AF@greyhawk.tonecommander.com> Okay. I got the Grandstream Gateway's 1-stage dialing working with Freeswitch (Thank You, Michael Collins and Thank All You Developers for creating this really slick Softswitch/PBX). Here are the changes/additions I made to the XML files: conf/sip_profiles/exernal/grandstreamGXW4104.xml (added file): conf/dialplan/default.xml (added to existing file): conf/dialplan/public.xml (added to existing file): conf/autoload_configs/acl.conf.xml (added to existing file): ... ... ... Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Friday, September 11, 2009 1:27 PM To: 'freeswitch-users at lists.freeswitch.org'; 'Michael Collins' Subject: RE: Inbound Gateway Call Not Working Thanks. I added the to both the "lan" list and "domain" list in the acl.conf.xml file and it does not try to authenticate anymore. However, now it replies to the INVITE with a 480 TEMPORARILY UNAVAILABLE. Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Friday, September 11, 2009 10:57 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: RE: Inbound Gateway Call Not Working By the way, the FS DEBUG console is saying the following when an inbound call is made: Rejected by acl "domains". Falling back to Digest auth. Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Friday, September 11, 2009 10:25 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: Inbound Gateway Call Not Working I am trying to configure a Grandstream gateway to work with FS. I can make outbound calls without a problem. However, inbound calls are getting a 403 Forbidden from FS in response to the INVITE from the gateway. Now, the INVITE's from address is the caller's number (e.g. 1112223333), which ofcourse, is foreign to the FS. So the FS sends a 407 Proxy Authentication Required and the gateway uses username "Anonymous" and the uri "sip:4000 at 192.168.72.38" (4000 is the destination for all calls from the gateway). Is there an example configuration for this scenario? Thanks and Best Regards, Jerry From jmesquita at freeswitch.org Mon Sep 14 12:13:03 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Mon, 14 Sep 2009 16:13:03 -0300 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> Message-ID: You can assign two things to me. 1. libesl code documentation (partially done and Doxygened - needs cleaning) 2. Bug marshal. I am setting up the proper lab environment here to be able to test most stuff. Count me in for any questions I can answer and I am _always_ on IRC jmesquita On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins wrote: > Hello FreeSWITCHers! > > We are looking for people who are in a position to help out with various > subprojects that will help FreeSWITCH to keep growing. We need people to > help out in these basic areas: > > Bug marshals (people who watch JIRA and test bug reports, patches, etc.) > Documentation maintainers (people who update the wiki when new stuff comes > out, also those familiar with mediawiki administration) > Documentation authors (people who write new docs, how-to's, tutorials, > examples, etc.) > Package maintainers (people who manage Debian debs, RPMs, etc.) > > Additionally, we are always looking for more folks to assist with answering > questions on IRC and the mailing list. It is definitely nice to have people > who've gone through the pains of switching to FreeSWITCH (or learning it > from scratch) who can assist the steady stream of new users. > > If you want to help and aren't sure where to go from here then please at > least do the following: > #1 - Join #freeswitch on irc.freenode.net and hang out as much as possible > #2 - Check the recent changes link on wiki.freeswitch.org each day > #3 - Join the Friday public conference call and listen in > These three things, in addition to the mailing list, will keep you well in > tune with the FreeSWITCH community and what's happening. > > Next, make a note of the parts of FS that you use frequently, know a lot > about, or are particularly passionate about. Those are the items we'd love > to have you help us with. For example: if you use mod_xml_curl frequently > and have been through the set up process then you're a prime candidate to > help answer questions, refine the mod_xml_curl wiki documentation, write up > a tutorial, contribute a working example of a web server & database schema, > etc. If you are good with a scripting language then we could definitely use > help with rounding out the docs for your favorite language. We could also > use code samples, so ask for a contrib folder if you have things you would > like to share. Or how about this: you read something on the wiki, it doesn't > quite work when you try, so you tinker until you figure it out. Now you're > in a position to update the wiki for everyone else's benefit, too. > > As you can see, you don't have to be a FreeSWITCH expert before you can > help the project. What we really need are people who care about the project > and want to see it flourish. If you are such a person then please contact me > off list. Tell me what you're good at or where you would like to help. > > Many thanks for all of your support! > -Michael > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/475d6fae/attachment.html From brian at freeswitch.org Mon Sep 14 12:37:04 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 14 Sep 2009 14:37:04 -0500 Subject: [Freeswitch-users] ATTENTION BEHAVIOR CHANGE of sip_invite_params variable. Message-ID: <94DD930D-4CAD-41A1-87BC-6FBD07C5AB99@freeswitch.org> I just committed revision 14849 to make sip_invite_params only apply to the RURI, If you wish to modify the To param son the invite you MUST use sip_invite_to_params moving forward. you have sip_invite_contact_params and sip_invite_from_params to work with also which were already there just making sure you know that now. Thanks, Brian West From rswagoner at gmail.com Mon Sep 14 12:46:21 2009 From: rswagoner at gmail.com (Ryan Wagoner) Date: Mon, 14 Sep 2009 15:46:21 -0400 Subject: [Freeswitch-users] DAHDI Dial 9 Receiving Setup Acknowledge Message-ID: <7d86ddb90909141246k557d1805t2e0132ea696451bc@mail.gmail.com> I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make calls from the Toshiba to Asterisk and internal calls from Asterisk to the Toshiba. What I can't do is make an call with an outside destination from Asterisk to the Toshiba. The Toshiba is looking for 9 to grab an outside line then it expects to see the 10 digits. In the FreePBX dial plan I use 9|. which sends 9 plus the 10 digit number. Using Wireshark to look at the QSIG commands coming from a Sangoma wanpipemon trace I see the following for an Asterisk to Toshiba internal call. Asterisk -> SETUP Toshiba -> CALL PROCESSING Toshiba -> CONNECT Asterisk -> CONNECT ACKNOWLEDGE However when trying to dial 9 + number I received the following Asterisk -> SETUP Toshiba -> SETUP ACKNOWLEDGE Looking at http://tools.ietf.org/html/rfc4497 I see the following On receipt of a QSIG SETUP message containing no Sending complete information element and a number in the Called party number information element that the gateway cannot determine to be complete, the gateway SHALL send back a QSIG SETUP ACKNOWLEDGE message, start QSIG timer T302, and await further number digits. Otherwise, the gateway SHALL wait for more digits to arrive in QSIG INFORMATION messages. Looking in the chan_dahdi.c code I see case PRI_EVENT_SETUP_ACK: chanpos = pri_find_principle(pri, e->setup_ack.channel); if (chanpos < 0) { ast_log(LOG_WARNING, "Received SETUP_ACKNOWLEDGE on unconfigured channel %d/%d span %d\n", PRI_SPAN(e->setup_ack.channel), PRI_CHANNEL(e->setup_ack.channel), pri->span); } else { chanpos = pri_fixup_principle(pri, chanpos, e->setup_ack.call); if (chanpos > -1) { ast_mutex_lock(&pri->pvts[chanpos]->lock); pri->pvts[chanpos]->setup_ack = 1; /* Send any queued digits */ for (x = 0;x < strlen(pri->pvts[chanpos]->dialdest); x++) { ast_debug(1, "Sending pending digit '%c'\n", pri->pvts[chanpos]->dialdest[x]); pri_information(pri->pri, pri->pvts[chanpos]->call, pri->pvts[chanpos]->dialdest[x]); } ast_mutex_unlock(&pri->pvts[chanpos]->lock); } else ast_log(LOG_WARNING, "Unable to move channel %d!\n", e->setup_ack.channel); } break; How do I get Asterisk to queue these digits so DAHDI can send them in response to the SETUP ACKNOWLEDGE message. What should be happening is Asterisk sends 9 via the SETUP message, waits for the SETUP ACKNOWLEDGE, then send the 10 digits number via a INFORMATION message. Ryan From rswagoner at gmail.com Mon Sep 14 12:48:04 2009 From: rswagoner at gmail.com (Ryan Wagoner) Date: Mon, 14 Sep 2009 15:48:04 -0400 Subject: [Freeswitch-users] DAHDI Dial 9 Receiving Setup Acknowledge In-Reply-To: <7d86ddb90909141246k557d1805t2e0132ea696451bc@mail.gmail.com> References: <7d86ddb90909141246k557d1805t2e0132ea696451bc@mail.gmail.com> Message-ID: <7d86ddb90909141248s6d4ee22rd9d2bbedb4cf481a@mail.gmail.com> Sorry this was meant for the Asterisk list. I wish FreeSWITCH had QSIG support so I could go that route. Ryan On Mon, Sep 14, 2009 at 3:46 PM, Ryan Wagoner wrote: > I have a Toshiba PBX connected via a QSIG PRI to Asterisk. I can make > calls from the Toshiba to Asterisk and internal calls from Asterisk to > the Toshiba. What I can't do is make an call with an outside > destination from Asterisk to the Toshiba. The Toshiba is looking for 9 > to grab an outside line then it expects to see the 10 digits. In the > FreePBX dial plan I use 9|. which sends 9 plus the 10 digit number. > > Using Wireshark to look at the QSIG commands coming from a Sangoma > wanpipemon trace I see the following for an Asterisk to Toshiba > internal call. > > Asterisk -> SETUP > Toshiba -> CALL PROCESSING > Toshiba -> CONNECT > Asterisk -> CONNECT ACKNOWLEDGE > > However when trying to dial 9 + number I received the following > > Asterisk -> SETUP > Toshiba -> SETUP ACKNOWLEDGE > > Looking at http://tools.ietf.org/html/rfc4497 I see the following > > ? On receipt of a QSIG SETUP message containing no Sending complete > ? information element and a number in the Called party number > ? information element that the gateway cannot determine to be complete, > ? the gateway SHALL send back a QSIG SETUP ACKNOWLEDGE message, start > ? QSIG timer T302, and await further number digits. > > ? Otherwise, the gateway SHALL wait for more digits > ? to arrive in QSIG INFORMATION messages. > > Looking in the chan_dahdi.c code I see > > ? ? ? ? ? ? ? ? ? ? ? ?case PRI_EVENT_SETUP_ACK: > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?chanpos = pri_find_principle(pri, > e->setup_ack.channel); > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?if (chanpos < 0) { > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?ast_log(LOG_WARNING, "Received > SETUP_ACKNOWLEDGE on unconfigured channel %d/%d span %d\n", > > PRI_SPAN(e->setup_ack.channel), PRI_CHANNEL(e->setup_ack.channel), > pri->span); > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?} else { > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?chanpos = > pri_fixup_principle(pri, chanpos, e->setup_ack.call); > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?if (chanpos > -1) { > > ast_mutex_lock(&pri->pvts[chanpos]->lock); > > pri->pvts[chanpos]->setup_ack = 1; > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?/* Send any queued digits */ > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?for (x = 0;x < > strlen(pri->pvts[chanpos]->dialdest); x++) { > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?ast_debug(1, > "Sending pending digit '%c'\n", pri->pvts[chanpos]->dialdest[x]); > > pri_information(pri->pri, pri->pvts[chanpos]->call, > > pri->pvts[chanpos]->dialdest[x]); > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?} > > ast_mutex_unlock(&pri->pvts[chanpos]->lock); > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?} else > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?ast_log(LOG_WARNING, > "Unable to move channel %d!\n", e->setup_ack.channel); > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?} > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?break; > > How do I get Asterisk to queue these digits so DAHDI can send them in > response to the SETUP ACKNOWLEDGE message. What should be happening is > Asterisk sends 9 via the SETUP message, waits for the SETUP > ACKNOWLEDGE, then send the 10 digits number via a INFORMATION message. > > Ryan > From m.sobkow at marketelsystems.com Mon Sep 14 12:54:31 2009 From: m.sobkow at marketelsystems.com (Mark Sobkow) Date: Mon, 14 Sep 2009 13:54:31 -0600 Subject: [Freeswitch-users] ERLang configuration callbacks Message-ID: <4AAE9F77.6090700@marketelsystems.com> I seem to be missing "something" in implementing the ERLang callbacks for Freeswitch. Our Freeswitch server is starting and getting registered with ERLang, we're invoking the bind for configuration, but I'm not seeing any of my callbacks fire. What am I missing? Sample code follows: -module(freeswitch_bind). -behaviour(gen_server). -record(st, {fsnode, pbxpid}). -export([start/3, terminate/2, code_change/3, init/1, handle_call/3, handle_cast/2, handle_info/2]). %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% %% gen_server methods start(Node, Section, Pid) -> gen_server:start(?MODULE, [Node, Section, Pid], []). init([Node, Section, Pid]) -> io:format( "freeswitch_bind:init( [Node=~w, Section=~w, Pid=~w])~n", [Node, Section, Pid] ), {api, Node} ! {bind, Section}, receive ok -> {ok, #st{fsnode=Node, pbxpid=Pid}}; {error, Reason} -> {stop, {error, {freeswitch_error, Reason}}} after 5000 -> {stop, {error, freeswitch_timeout}} end. terminate(_Reason, _State) -> ok. code_change(_OldVsn, State, _Extra) -> {ok, State}. %% %% Configuration handler replies that the requested document section, tag, and key are not %% found. %% handle_call({fetch, configuration, Tag, Key, Value, Params}, _From, State) -> io:format( "freeswitch_fetch:handle_call( {fetch, configuration, Tag=~w, Key=~w, Value=~w, Params=~w}, _From, State=~w)~n", [Tag, Key, Value, Params, State]), Xml = "
", { reply, {ok, Xml }, State }; %% %% Directory handler replies that the requested document section, tag, and key are not %% found. %% handle_call({fetch, directory, Tag, Key, Value, Params}, _From, State) -> io:format( "freeswitch_fetch:handle_call( {fetch, directory, Tag=~w, Key=~w, Value=~w, Params=~w}, _From, State=~w)~n", [Tag, Key, Value, Params, State]), Xml = "
", { reply, {ok, Xml }, State }; %% %% Dialplan handler replies that the requested document section, tag, and key are not %% found. %% handle_call({fetch, dialplan, Tag, Key, Value, Params}, _From, State) -> io:format( "freeswitch_fetch:handle_call( {fetch, dialplan, Tag=~w, Key=~w, Value=~w, Params=~w}, _From, State=~w)~n", [Tag, Key, Value, Params, State]), Xml = "
", { reply, {ok, Xml }, State }; %% %% Default handler replies that the requested document section, tag, and key are not %% found. %% handle_call({fetch, Section, Tag, Key, Value, Params}, _From, State) -> io:format( "freeswitch_fetch:handle_call( {fetch, Section=~w, Tag=~w, Key=~w, Value=~w, Params=~w}, _From, State=~w)~n", [Section, Tag, Key, Value, Params, State]), Xml = "
", { reply, {ok, Xml }, State }; %% %% If the request isn't recognized, just log it and do nothing. %% handle_call(Request, _From, State) -> io:format("freeswitch_bind:handle_call( ~w, _From, State) unrecognized request~n", [Request]), {reply, {error, unrecognized_request}, State}. handle_cast(Message, State) -> error_logger:error_msg("~p received unrecognized cast ~p~n", [self(), Message]), {noreply, State}. handle_info({fetch, Section, Tag, Key, Value, FetchID, Params}, #st{fsnode=Node, pbxpid=Pid}=State) -> {ok, XML} = gen_server:call(Pid, {fetch, Section, Tag, Key, Value, Params}), {api, Node} ! {fetch_reply, FetchID, XML}, receive ok -> {noreply, State}; {error, Reason} -> {stop, {error, Reason}, State} end. From diego.viola at gmail.com Mon Sep 14 14:32:01 2009 From: diego.viola at gmail.com (Diego Viola) Date: Mon, 14 Sep 2009 21:32:01 +0000 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> Message-ID: <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> Hi Michael, You can count with me for anything else, like documentation, coding/scripting, or any other FreeSWITCH related stuff. Regards, Diego 2009/9/14 Jo?o Mesquita > You can assign two things to me. > > 1. libesl code documentation (partially done and Doxygened - needs > cleaning) > 2. Bug marshal. I am setting up the proper lab environment here to be able > to test most stuff. > > Count me in for any questions I can answer and I am _always_ on IRC > > jmesquita > > On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins wrote: > >> Hello FreeSWITCHers! >> >> We are looking for people who are in a position to help out with various >> subprojects that will help FreeSWITCH to keep growing. We need people to >> help out in these basic areas: >> >> Bug marshals (people who watch JIRA and test bug reports, patches, etc.) >> Documentation maintainers (people who update the wiki when new stuff comes >> out, also those familiar with mediawiki administration) >> Documentation authors (people who write new docs, how-to's, tutorials, >> examples, etc.) >> Package maintainers (people who manage Debian debs, RPMs, etc.) >> >> Additionally, we are always looking for more folks to assist with >> answering questions on IRC and the mailing list. It is definitely nice to >> have people who've gone through the pains of switching to FreeSWITCH (or >> learning it from scratch) who can assist the steady stream of new users. >> >> If you want to help and aren't sure where to go from here then please at >> least do the following: >> #1 - Join #freeswitch on irc.freenode.net and hang out as much as >> possible >> #2 - Check the recent changes link on wiki.freeswitch.org each day >> #3 - Join the Friday public conference call and listen in >> These three things, in addition to the mailing list, will keep you well in >> tune with the FreeSWITCH community and what's happening. >> >> Next, make a note of the parts of FS that you use frequently, know a lot >> about, or are particularly passionate about. Those are the items we'd love >> to have you help us with. For example: if you use mod_xml_curl frequently >> and have been through the set up process then you're a prime candidate to >> help answer questions, refine the mod_xml_curl wiki documentation, write up >> a tutorial, contribute a working example of a web server & database schema, >> etc. If you are good with a scripting language then we could definitely use >> help with rounding out the docs for your favorite language. We could also >> use code samples, so ask for a contrib folder if you have things you would >> like to share. Or how about this: you read something on the wiki, it doesn't >> quite work when you try, so you tinker until you figure it out. Now you're >> in a position to update the wiki for everyone else's benefit, too. >> >> As you can see, you don't have to be a FreeSWITCH expert before you can >> help the project. What we really need are people who care about the project >> and want to see it flourish. If you are such a person then please contact me >> off list. Tell me what you're good at or where you would like to help. >> >> Many thanks for all of your support! >> -Michael >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/38dc7b08/attachment.html From mgende at gendesign.com Mon Sep 14 18:42:57 2009 From: mgende at gendesign.com (Michael Gende) Date: Mon, 14 Sep 2009 20:42:57 -0500 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> Message-ID: I'm good for coming up with some documentation (which I'm doing anyway for my guys at the office). Not that whats on the wiki isn't good and I'll likely steal; its all there if you read it. I'll submit this when I reach some measure of completeness. If its deemed good, great. If not, well, I probably need to know that anyway. Mike G. On Mon, Sep 14, 2009 at 4:32 PM, Diego Viola wrote: > Hi Michael, > > You can count with me for anything else, like documentation, > coding/scripting, or any other FreeSWITCH related stuff. > > Regards, > > Diego > > 2009/9/14 Jo?o Mesquita > >> You can assign two things to me. >> >> 1. libesl code documentation (partially done and Doxygened - needs >> cleaning) >> 2. Bug marshal. I am setting up the proper lab environment here to be able >> to test most stuff. >> >> Count me in for any questions I can answer and I am _always_ on IRC >> >> jmesquita >> >> On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins wrote: >> >>> Hello FreeSWITCHers! >>> >>> We are looking for people who are in a position to help out with various >>> subprojects that will help FreeSWITCH to keep growing. We need people to >>> help out in these basic areas: >>> >>> Bug marshals (people who watch JIRA and test bug reports, patches, etc.) >>> Documentation maintainers (people who update the wiki when new stuff >>> comes out, also those familiar with mediawiki administration) >>> Documentation authors (people who write new docs, how-to's, tutorials, >>> examples, etc.) >>> Package maintainers (people who manage Debian debs, RPMs, etc.) >>> >>> Additionally, we are always looking for more folks to assist with >>> answering questions on IRC and the mailing list. It is definitely nice to >>> have people who've gone through the pains of switching to FreeSWITCH (or >>> learning it from scratch) who can assist the steady stream of new users. >>> >>> If you want to help and aren't sure where to go from here then please at >>> least do the following: >>> #1 - Join #freeswitch on irc.freenode.net and hang out as much as >>> possible >>> #2 - Check the recent changes link on wiki.freeswitch.org each day >>> #3 - Join the Friday public conference call and listen in >>> These three things, in addition to the mailing list, will keep you well >>> in tune with the FreeSWITCH community and what's happening. >>> >>> Next, make a note of the parts of FS that you use frequently, know a lot >>> about, or are particularly passionate about. Those are the items we'd love >>> to have you help us with. For example: if you use mod_xml_curl frequently >>> and have been through the set up process then you're a prime candidate to >>> help answer questions, refine the mod_xml_curl wiki documentation, write up >>> a tutorial, contribute a working example of a web server & database schema, >>> etc. If you are good with a scripting language then we could definitely use >>> help with rounding out the docs for your favorite language. We could also >>> use code samples, so ask for a contrib folder if you have things you would >>> like to share. Or how about this: you read something on the wiki, it doesn't >>> quite work when you try, so you tinker until you figure it out. Now you're >>> in a position to update the wiki for everyone else's benefit, too. >>> >>> As you can see, you don't have to be a FreeSWITCH expert before you can >>> help the project. What we really need are people who care about the project >>> and want to see it flourish. If you are such a person then please contact me >>> off list. Tell me what you're good at or where you would like to help. >>> >>> Many thanks for all of your support! >>> -Michael >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090914/73bd9ca0/attachment.html From jingwei.yang at gmail.com Mon Sep 14 19:09:06 2009 From: jingwei.yang at gmail.com (Jingwei Yang) Date: Tue, 15 Sep 2009 10:09:06 +0800 Subject: [Freeswitch-users] compilation error with the latest codes In-Reply-To: <4D169DC3-E8A4-44A4-816A-52857772FBFB@freeswitch.org> References: <13529f9d0909140146r115db921l5a2f9bb92a582412@mail.gmail.com> <4D169DC3-E8A4-44A4-816A-52857772FBFB@freeswitch.org> Message-ID: <13529f9d0909141909s390e752at5546e0c27facfb7a@mail.gmail.com> Thanks Brian. On Mon, Sep 14, 2009 at 8:55 PM, Brian West wrote: > You have a merge conflict please svn revert sofia.c > > /b > > On Sep 14, 2009, at 3:46 AM, Jingwei Yang wrote: > > > Hi Folks, > > > > I've got a compilation error with the latest codes (r14842) > > > > Making all in packages > > Creating mod_sofia_la-mod_sofia.lo > > Compiling mod_sofia.c ... > > Creating mod_sofia_la-sofia.lo > > Compiling sofia.c ... > > sofia.c: In function ???sofia_handle_sip_r_invite???: > > sofia.c:3221: error: expected expression before ???< > make[5]: *** [mod_sofia_la-sofia.lo] Error 1 > > make[4]: *** [all] Error 2 > > make[3]: *** [mod_sofia-all] Error 1 > > make[2]: *** [all-recursive] Error 1 > > > > Does anyone have ideas about this? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/50fc7e21/attachment.html From yehavi.bourvine at gmail.com Mon Sep 14 23:55:51 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 15 Sep 2009 09:55:51 +0300 Subject: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec. In-Reply-To: <20090908071240.GA12470@jdc.jasonjgw.net> References: <20090908071240.GA12470@jdc.jasonjgw.net> Message-ID: Hello Jason, Sorry for the delay in answering - I saw your reply only now as it got burried with some other stuff... Anyway, I attach bellow the relevant sip trace. Phone 80678 (132.64.4.137) is calling 80679 (132.64.4.135) which answers. When 80679 presses the Hold or Transfer button the call is disconnected. Thanks! __Yehavi: 2009/9/8 Jason White > Yehavi Bourvine wrote: > > > > I have a problem when trying to put a call on hold: I get the above > > message and the call is disconnected. Any idea where to look for the > source > > of the problem? > > My next step in your situation would be to obtain a Sip trace and post > relevant details from it to the list. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/bab02b72/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: typescript.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/bab02b72/attachment-0001.txt From demuel at thephinix.org Tue Sep 15 00:38:54 2009 From: demuel at thephinix.org (demuel at thephinix.org) Date: Tue, 15 Sep 2009 08:38:54 +0100 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> Message-ID: <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> Hi Michael / FS enthusiast, In my opinion, it will be an undeniably perfect chaos if everyone wants to do everything. One way I can suggest out of this is the following: - sets of persons that does all the stuff on the FS core. I'm not sure if this one will work out but I think these are the only persons who has commit access to the source code and will also be willing to accept modifications, enhancements, etc. - individuals can ask if they can be the maintainer/tester for a particular module. Bug fixing, sending modifications and enhancements will still be subject to the approval of anthm. - an array of persons that will take ownership on what FS can do and provide realiable working examples for it. Currently, it takes amount of pain for a newbie on how to configure SIP because the examples are too confusing and sometimes there is not certainty if this will work on what release of FS or not. - any individual that has time and interest can start owning the porting of FS to other operating system. Like in my case, I am working on making FS ported to FreeBSD ports but I don't know if somebody did it or if he is currently doing it, as to what stage he is in? - there should be a release engineering team. In my opinion, a stable and current release will be much more sane. As with any other successful open source project, we won't be starting to say I want to contribute this and start working on that. We should indicate that you take ownership of this and any comments should be forwarded right unto you. Again, this is just my opinion. You can take some of it or leave it using "sudo rm -rf blah/*" . There and back again, Demuel I. Bendano a.k.a engrxyz > Hi Michael, > > You can count with me for anything else, like documentation, > coding/scripting, or any other FreeSWITCH related stuff. > > Regards, > > Diego > > 2009/9/14 Jo?o Mesquita > >> You can assign two things to me. >> >> 1. libesl code documentation (partially done and Doxygened - needs >> cleaning) >> 2. Bug marshal. I am setting up the proper lab environment here to be able >> to test most stuff. >> >> Count me in for any questions I can answer and I am _always_ on IRC >> >> jmesquita >> >> On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins wrote: >> >>> Hello FreeSWITCHers! >>> >>> We are looking for people who are in a position to help out with various >>> subprojects that will help FreeSWITCH to keep growing. We need people to >>> help out in these basic areas: >>> >>> Bug marshals (people who watch JIRA and test bug reports, patches, etc.) >>> Documentation maintainers (people who update the wiki when new stuff comes >>> out, also those familiar with mediawiki administration) >>> Documentation authors (people who write new docs, how-to's, tutorials, >>> examples, etc.) >>> Package maintainers (people who manage Debian debs, RPMs, etc.) >>> >>> Additionally, we are always looking for more folks to assist with >>> answering questions on IRC and the mailing list. It is definitely nice to >>> have people who've gone through the pains of switching to FreeSWITCH (or >>> learning it from scratch) who can assist the steady stream of new users. >>> >>> If you want to help and aren't sure where to go from here then please at >>> least do the following: >>> #1 - Join #freeswitch on irc.freenode.net and hang out as much as >>> possible >>> #2 - Check the recent changes link on wiki.freeswitch.org each day >>> #3 - Join the Friday public conference call and listen in >>> These three things, in addition to the mailing list, will keep you well in >>> tune with the FreeSWITCH community and what's happening. >>> >>> Next, make a note of the parts of FS that you use frequently, know a lot >>> about, or are particularly passionate about. Those are the items we'd love >>> to have you help us with. For example: if you use mod_xml_curl frequently >>> and have been through the set up process then you're a prime candidate to >>> help answer questions, refine the mod_xml_curl wiki documentation, write up >>> a tutorial, contribute a working example of a web server & database schema, >>> etc. If you are good with a scripting language then we could definitely use >>> help with rounding out the docs for your favorite language. We could also >>> use code samples, so ask for a contrib folder if you have things you would >>> like to share. Or how about this: you read something on the wiki, it doesn't >>> quite work when you try, so you tinker until you figure it out. Now you're >>> in a position to update the wiki for everyone else's benefit, too. >>> >>> As you can see, you don't have to be a FreeSWITCH expert before you can >>> help the project. What we really need are people who care about the project >>> and want to see it flourish. If you are such a person then please contact me >>> off list. Tell me what you're good at or where you would like to help. >>> >>> Many thanks for all of your support! >>> -Michael >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From cstomi.levlist at gmail.com Tue Sep 15 03:29:23 2009 From: cstomi.levlist at gmail.com (Tamas Cseke) Date: Tue, 15 Sep 2009 12:29:23 +0200 Subject: [Freeswitch-users] stange segfaults with log Message-ID: <4AAF6C83.2080203@gmail.com> Hello we have a strange problem with 14144 revision. It seems switch_log_printf got NULL pointer as data. It happens a few times. however in the previous frame session seems to be good for us. http://pastebin.freeswitch.org/10357 Could you please tell me what is the problem? Did we make some mistakes with building? Missed "make clean", or someting? Thanks in advance, Tamas From matt at venturevoip.com Tue Sep 15 04:10:05 2009 From: matt at venturevoip.com (Matt Riddell) Date: Tue, 15 Sep 2009 23:10:05 +1200 Subject: [Freeswitch-users] Limit_Hash Message-ID: <4AAF760D.3030105@venturevoip.com> Hi, I've moved this discussion to users as it seems my query is moving in that direction :) So, upon looking at limit_hash, it appears to do what I need to do. My question then becomes, how do I set a hash for an originated call? It seems that limit_hash is an application rather than a channel variable, and so far I've been doing most things without touching the dialplan. So, say I want to originate 9 calls, 3 from 3 customers. I would like to mark the calls with my_customer_group_1 through 3, and then use the limit_hash_usage command to verify the count of channels in each group. I therefore have a few questions: 1. Can I mark a call in the originate statement? 2. How do I use the limit_hash_usage command? The wiki states: You can verify the usage of any resource with the limit_hash_usage api call. limit_hash_usage Is realm the same as a SIP realm? Is id the hash that I have used to mark the call with? Just making sure :) -- Cheers, Matt Riddell Director From brian at freeswitch.org Tue Sep 15 05:44:19 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Sep 2009 07:44:19 -0500 Subject: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec. In-Reply-To: References: <20090908071240.GA12470@jdc.jasonjgw.net> Message-ID: <4C8A2EB0-7E3C-41C5-A085-743B82AE706A@freeswitch.org> Do you have Late Negotiation on? Also is this the only FreeSWITCH log output you have in this transfer? /b On Sep 15, 2009, at 1:55 AM, Yehavi Bourvine wrote: > Hello Jason, > > Sorry for the delay in answering - I saw your reply only now as it > got burried with some other stuff... > > Anyway, I attach bellow the relevant sip trace. Phone 80678 > (132.64.4.137) is calling 80679 (132.64.4.135) which answers. When > 80679 presses the Hold or Transfer button the call is disconnected. > > Thanks! __Yehavi: From yehavi.bourvine at gmail.com Tue Sep 15 06:17:27 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 15 Sep 2009 16:17:27 +0300 Subject: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec. In-Reply-To: <4C8A2EB0-7E3C-41C5-A085-743B82AE706A@freeswitch.org> References: <20090908071240.GA12470@jdc.jasonjgw.net> <4C8A2EB0-7E3C-41C5-A085-743B82AE706A@freeswitch.org> Message-ID: No, I have late negotiation commented out. This is the only log from the beginning of the session until it disconnects. Shall I turn on more debugging (if available)? Thanks, __Yehavi: 2009/9/15 Brian West > Do you have Late Negotiation on? Also is this the only FreeSWITCH log > output you have in this transfer? > > > On Sep 15, 2009, at 1:55 AM, Yehavi Bourvine wrote: > > > Hello Jason, > > > > Sorry for the delay in answering - I saw your reply only now as it > > got burried with some other stuff... > > > > Anyway, I attach bellow the relevant sip trace. Phone 80678 > > (132.64.4.137) is calling 80679 (132.64.4.135) which answers. When > > 80679 presses the Hold or Transfer button the call is disconnected. > > > > Thanks! __Yehavi: > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/56e9076a/attachment.html From shiyanov at gmail.com Tue Sep 15 06:43:38 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Tue, 15 Sep 2009 17:43:38 +0400 Subject: [Freeswitch-users] problem: no audio for one of the person in conference Message-ID: Hi there! The situation is: - Person A calls to the extension: - I bridge him with person B with help of mod_socket: SendMsg call-command: execute execute-app-name: bridge execute-app-arg: - A and B talks - Person C decides to barge in the call A<-->B (to become a third participator in the call) a) I send (mod_socket): api originate user/ &park() b) then I move A, B, C to the extension: conference profile "my_profile" is: The "moving" itself is done by sending this for each (A,B,C) channel SendMsg call-command: execute execute-app-name: execute_extension execute-app-arg: barge_in - Result: A, B, C are in the same conference with name "my_confname", A can hear B and vice verse, but both A and B can't hear C. C also doesn't hear neither A nor B. I also tried the "moving" to conference with api uuid_transfer -both barge_in api uuid_transfer barge_in but result is the same. Maybe someone already faced with such issue? Thanks, Artem -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/0787818d/attachment.html From yehavi.bourvine at gmail.com Tue Sep 15 06:49:05 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 15 Sep 2009 16:49:05 +0300 Subject: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec. In-Reply-To: References: <20090908071240.GA12470@jdc.jasonjgw.net> <4C8A2EB0-7E3C-41C5-A085-743B82AE706A@freeswitch.org> Message-ID: Sorry, seems that I did not turn-on debug (F8) in the previous logfile. here it is again (this time 80678 is calling 80675 which places it on hold and then the call disconnects). Thanks, __Yehavi: 2009/9/15 Yehavi Bourvine > No, I have late negotiation commented out. > > This is the only log from the beginning of the session until it > disconnects. Shall I turn on more debugging (if available)? > > Thanks, __Yehavi: > > 2009/9/15 Brian West > >> Do you have Late Negotiation on? Also is this the only FreeSWITCH log >> output you have in this transfer? > > > > >> >> >> On Sep 15, 2009, at 1:55 AM, Yehavi Bourvine wrote: >> >> > Hello Jason, >> > >> > Sorry for the delay in answering - I saw your reply only now as it >> > got burried with some other stuff... >> > >> > Anyway, I attach bellow the relevant sip trace. Phone 80678 >> > (132.64.4.137) is calling 80679 (132.64.4.135) which answers. When >> > 80679 presses the Hold or Transfer button the call is disconnected. >> > >> > Thanks! __Yehavi: >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/4ab38d79/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: typescript.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/4ab38d79/attachment-0001.txt From shiyanov at gmail.com Tue Sep 15 07:22:19 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Tue, 15 Sep 2009 18:22:19 +0400 Subject: [Freeswitch-users] "barge in" implementation with mod_socket and eavesdrop Message-ID: Hello! I'm trying to implement "barge in" functionality (see http://www.yourdictionary.com/telecom/barge-in) with "eavesdrop" but still with no success. The situation is: - Person A calls to the extension: - I bridge him with person B with help of mod_socket: SendMsg call-command: execute execute-app-name: bridge execute-app-arg: - A and B talks - Person C decides to barge in the call A<-->B (to become a third participator in the call) a) I send (mod_socket): SendMsg call-command: execute execute-app-name: eavesdrop execute-app-arg: b) Then, as the spec says ( http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop) I sent DTMF "3" with api uuid_send_dtmf 3 but it doesn't work. I mean that A can hear B and vice verse, but both A and B can't hear C. C also doesn't hear neither A nor B. If I press "3" on the C's softphone (latest X-Lite) then, really, C becomes a full-capabilities participator of the call. Instead of "uuid_send_dtmf" I tried: 1) sendevent DTMF Unique-ID: DTMF-Digit: 3 DTMF-Duration: 2000 2) first make queue_dtmf for the , and then eavesdrop 3) SendMsg call-command: execute execute-app-name: gentones execute-app-arg: 3 4) SendMsg call-command: execute execute-app-name: send_dtmf execute-app-arg: 3 And none of these methods leads to the "barged in" call. Anyone knows how to press "3" programmatically on behalf of the given channel with mod_socket?! Artem -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/75a9625c/attachment.html From m.sobkow at marketelsystems.com Tue Sep 15 07:38:41 2009 From: m.sobkow at marketelsystems.com (Mark Sobkow) Date: Tue, 15 Sep 2009 08:38:41 -0600 Subject: [Freeswitch-users] I got ERLang to fire a configuration request Message-ID: <4AAFA6F1.8050802@marketelsystems.com> I still need to stuff the Freeswitch PID into global storage somewhere so the process that's handling the configuration requests can send the reply without crashing (it's just getting a node id, not a Pid), but I seem to be on my way to configuring Freeswitch via ERLang. freeswitch_bind.erl has the calls added to register the ERLang callbacks. The callback function itself is in the aptly named freeswitch_callback.erl. -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... 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Name: freeswitch_callback.erl Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/e95c7d9a/attachment-0001.pl From mgende at gendesign.com Tue Sep 15 07:50:45 2009 From: mgende at gendesign.com (Michael Gende) Date: Tue, 15 Sep 2009 09:50:45 -0500 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> Message-ID: Hey Demuel, et al, I agree that without coordination, there will be duplication of effort, etc. How does one "find the line to get in" to help with, in my case, documentation? I don't want to, with the best of intentions, simply create confusion. I'm new, so if I've simply not yet RTFM that answers my question above, some kind person might nudge me in the right direction. Back to registering with my SIP provider... Regards, Mike G. On Tue, Sep 15, 2009 at 2:38 AM, wrote: > Hi Michael / FS enthusiast, > > In my opinion, it will be an undeniably perfect chaos if everyone wants to > do everything. One way > I can suggest out of this is the following: > > - sets of persons that does all the stuff on the FS core. I'm not sure if > this one will work out but > I think these are the only persons who has commit access to the source code > and will also be willing > to accept modifications, enhancements, etc. > > - individuals can ask if they can be the maintainer/tester for a particular > module. Bug fixing, > sending modifications and enhancements will still be subject to the > approval of anthm. > > - an array of persons that will take ownership on what FS can do and > provide realiable working > examples for it. Currently, it takes amount of pain for a newbie on how to > configure SIP because > the examples are too confusing and sometimes there is not certainty if this > will work on what > release of FS or not. > > - any individual that has time and interest can start owning the porting of > FS to other operating > system. Like in my case, I am working on making FS ported to FreeBSD ports > but I don't know if > somebody did it or if he is currently doing it, as to what stage he is in? > > - there should be a release engineering team. In my opinion, a stable and > current release will be > much more sane. > > As with any other successful open source project, we won't be starting to > say I want to contribute > this and start working on that. We should indicate that you take ownership > of this and any > comments should be forwarded right unto you. > > Again, this is just my opinion. You can take some of it or leave it using > "sudo rm -rf blah/*" . > > There and back again, > Demuel I. Bendano > a.k.a engrxyz > > > > Hi Michael, > > > > You can count with me for anything else, like documentation, > > coding/scripting, or any other FreeSWITCH related stuff. > > > > Regards, > > > > Diego > > > > 2009/9/14 Jo?o Mesquita > > > >> You can assign two things to me. > >> > >> 1. libesl code documentation (partially done and Doxygened - needs > >> cleaning) > >> 2. Bug marshal. I am setting up the proper lab environment here to be > able > >> to test most stuff. > >> > >> Count me in for any questions I can answer and I am _always_ on IRC > >> > >> jmesquita > >> > >> On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins >wrote: > >> > >>> Hello FreeSWITCHers! > >>> > >>> We are looking for people who are in a position to help out with > various > >>> subprojects that will help FreeSWITCH to keep growing. We need people > to > >>> help out in these basic areas: > >>> > >>> Bug marshals (people who watch JIRA and test bug reports, patches, > etc.) > >>> Documentation maintainers (people who update the wiki when new stuff > comes > >>> out, also those familiar with mediawiki administration) > >>> Documentation authors (people who write new docs, how-to's, tutorials, > >>> examples, etc.) > >>> Package maintainers (people who manage Debian debs, RPMs, etc.) > >>> > >>> Additionally, we are always looking for more folks to assist with > >>> answering questions on IRC and the mailing list. It is definitely nice > to > >>> have people who've gone through the pains of switching to FreeSWITCH > (or > >>> learning it from scratch) who can assist the steady stream of new > users. > >>> > >>> If you want to help and aren't sure where to go from here then please > at > >>> least do the following: > >>> #1 - Join #freeswitch on irc.freenode.net and hang out as much as > >>> possible > >>> #2 - Check the recent changes link on wiki.freeswitch.org each day > >>> #3 - Join the Friday public conference call and listen in > >>> These three things, in addition to the mailing list, will keep you well > in > >>> tune with the FreeSWITCH community and what's happening. > >>> > >>> Next, make a note of the parts of FS that you use frequently, know a > lot > >>> about, or are particularly passionate about. Those are the items we'd > love > >>> to have you help us with. For example: if you use mod_xml_curl > frequently > >>> and have been through the set up process then you're a prime candidate > to > >>> help answer questions, refine the mod_xml_curl wiki documentation, > write up > >>> a tutorial, contribute a working example of a web server & database > schema, > >>> etc. If you are good with a scripting language then we could definitely > use > >>> help with rounding out the docs for your favorite language. We could > also > >>> use code samples, so ask for a contrib folder if you have things you > would > >>> like to share. Or how about this: you read something on the wiki, it > doesn't > >>> quite work when you try, so you tinker until you figure it out. Now > you're > >>> in a position to update the wiki for everyone else's benefit, too. > >>> > >>> As you can see, you don't have to be a FreeSWITCH expert before you can > >>> help the project. What we really need are people who care about the > project > >>> and want to see it flourish. If you are such a person then please > contact me > >>> off list. Tell me what you're good at or where you would like to help. > >>> > >>> Many thanks for all of your support! > >>> -Michael > >>> > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/166ac3e5/attachment-0001.html From mgende at gendesign.com Tue Sep 15 07:56:09 2009 From: mgende at gendesign.com (Michael Gende) Date: Tue, 15 Sep 2009 09:56:09 -0500 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> Message-ID: Answered my own question. Will email Diego off-list. On Tue, Sep 15, 2009 at 9:50 AM, Michael Gende wrote: > Hey Demuel, et al, > > I agree that without coordination, there will be duplication of effort, > etc. > > How does one "find the line to get in" to help with, in my case, > documentation? I don't want to, with the best of intentions, simply create > confusion. > > I'm new, so if I've simply not yet RTFM that answers my question above, > some kind person might nudge me in the right direction. > > Back to registering with my SIP provider... > > Regards, > > Mike G. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/b1ec01dd/attachment.html From brian at freeswitch.org Tue Sep 15 07:58:35 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Sep 2009 09:58:35 -0500 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> Message-ID: The friday meetings are where we all collaborate on these group efforts and discuss project direction, goals and areas where people can help out more. Did we ever find someone to officially take over the Debian packages? /b On Sep 15, 2009, at 9:56 AM, Michael Gende wrote: > Answered my own question. Will email Diego off-list. > > On Tue, Sep 15, 2009 at 9:50 AM, Michael Gende > wrote: > Hey Demuel, et al, > > I agree that without coordination, there will be duplication of > effort, etc. > > How does one "find the line to get in" to help with, in my case, > documentation? I don't want to, with the best of intentions, simply > create confusion. > > I'm new, so if I've simply not yet RTFM that answers my question > above, some kind person might nudge me in the right direction. > > Back to registering with my SIP provider... > > Regards, > > Mike G. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/e4053a99/attachment.html From anthony.minessale at gmail.com Tue Sep 15 08:07:34 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Sep 2009 10:07:34 -0500 Subject: [Freeswitch-users] problem: no audio for one of the person in conference In-Reply-To: References: Message-ID: <191c3a030909150807v38ebc03eld0610da78968ad02@mail.gmail.com> replace "execute_extension" with "transfer" or use the application "three_way" On Tue, Sep 15, 2009 at 8:43 AM, Artem Shiyanov wrote: > Hi there! > > The situation is: > - Person A calls to the extension: > > > > > > > > - I bridge him with person B with help of mod_socket: > SendMsg > call-command: execute > execute-app-name: bridge > execute-app-arg: > > - A and B talks > > - Person C decides to barge in the call A<-->B (to become a third > participator in the call) > a) I send (mod_socket): > api originate user/ &park() > b) then I move A, B, C to the extension: > > > > > conference profile "my_profile" is: > > > > > > > > > > > > > > > The "moving" itself is done by sending this for each (A,B,C) channel > SendMsg > call-command: execute > execute-app-name: execute_extension > execute-app-arg: barge_in > > - Result: A, B, C are in the same conference with name "my_confname", A can > hear B and vice verse, but both A and B can't hear C. C also doesn't hear > neither A nor B. > > I also tried the "moving" to conference with > api uuid_transfer -both barge_in > api uuid_transfer barge_in > but result is the same. > > Maybe someone already faced with such issue? > > > Thanks, > Artem > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/669b7851/attachment.html From anthony.minessale at gmail.com Tue Sep 15 08:08:22 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Sep 2009 10:08:22 -0500 Subject: [Freeswitch-users] "barge in" implementation with mod_socket and eavesdrop In-Reply-To: References: Message-ID: <191c3a030909150808q282d2ee7y3cf9ab2e4c8d77ae@mail.gmail.com> yes call the app as "three_way" like i said in the other thread. On Tue, Sep 15, 2009 at 9:22 AM, Artem Shiyanov wrote: > Hello! > > I'm trying to implement "barge in" functionality (see > http://www.yourdictionary.com/telecom/barge-in) with "eavesdrop" but still > with no success. > > The situation is: > - Person A calls to the extension: > > > > > > > > - I bridge him with person B with help of mod_socket: > SendMsg > call-command: execute > execute-app-name: bridge > execute-app-arg: > > - A and B talks > > - Person C decides to barge in the call A<-->B (to become a third > participator in the call) > a) I send (mod_socket): > SendMsg > call-command: execute > execute-app-name: eavesdrop > execute-app-arg: > > b) Then, as the spec says ( > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop) > I sent DTMF "3" with > api uuid_send_dtmf 3 > but it doesn't work. I mean that A can hear B and vice verse, but both A > and B can't hear C. C also doesn't hear neither A nor B. > > If I press "3" on the C's softphone (latest X-Lite) then, really, C becomes > a full-capabilities participator of the call. > Instead of "uuid_send_dtmf" I tried: > 1) > sendevent DTMF > Unique-ID: > DTMF-Digit: 3 > DTMF-Duration: 2000 > > 2) first make queue_dtmf for the , and then eavesdrop > > 3) > SendMsg > call-command: execute > execute-app-name: gentones > execute-app-arg: 3 > > 4) > SendMsg > call-command: execute > execute-app-name: send_dtmf > execute-app-arg: 3 > > And none of these methods leads to the "barged in" call. > > Anyone knows how to press "3" programmatically on behalf of the given > channel with mod_socket?! > > > Artem > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/b0569452/attachment.html From msc at freeswitch.org Tue Sep 15 08:25:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Sep 2009 08:25:10 -0700 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> Message-ID: <87f2f3b90909150825o472ecf23h4088668266051d5a@mail.gmail.com> Demuel, Thanks for the input. Yes, we want to avoid chaos. I will work to keep everyone organized. I would welcome a mass of people all trying to do different things because the challenge would simply be to keep them organized. Right now the challenge is in recruiting people to stay with the not-so-glorious aspects of the project, namely documentation, maintenance, and janitorial style subprojects. Please keep the comments and suggestions coming! Thanks, MC On Tue, Sep 15, 2009 at 12:38 AM, wrote: > Hi Michael / FS enthusiast, > > In my opinion, it will be an undeniably perfect chaos if everyone wants to > do everything. One way > I can suggest out of this is the following: > > - sets of persons that does all the stuff on the FS core. I'm not sure if > this one will work out but > I think these are the only persons who has commit access to the source code > and will also be willing > to accept modifications, enhancements, etc. > > - individuals can ask if they can be the maintainer/tester for a particular > module. Bug fixing, > sending modifications and enhancements will still be subject to the > approval of anthm. > > - an array of persons that will take ownership on what FS can do and > provide realiable working > examples for it. Currently, it takes amount of pain for a newbie on how to > configure SIP because > the examples are too confusing and sometimes there is not certainty if this > will work on what > release of FS or not. > > - any individual that has time and interest can start owning the porting of > FS to other operating > system. Like in my case, I am working on making FS ported to FreeBSD ports > but I don't know if > somebody did it or if he is currently doing it, as to what stage he is in? > > - there should be a release engineering team. In my opinion, a stable and > current release will be > much more sane. > > As with any other successful open source project, we won't be starting to > say I want to contribute > this and start working on that. We should indicate that you take ownership > of this and any > comments should be forwarded right unto you. > > Again, this is just my opinion. You can take some of it or leave it using > "sudo rm -rf blah/*" . > > There and back again, > Demuel I. Bendano > a.k.a engrxyz > > > > Hi Michael, > > > > You can count with me for anything else, like documentation, > > coding/scripting, or any other FreeSWITCH related stuff. > > > > Regards, > > > > Diego > > > > 2009/9/14 Jo?o Mesquita > > > >> You can assign two things to me. > >> > >> 1. libesl code documentation (partially done and Doxygened - needs > >> cleaning) > >> 2. Bug marshal. I am setting up the proper lab environment here to be > able > >> to test most stuff. > >> > >> Count me in for any questions I can answer and I am _always_ on IRC > >> > >> jmesquita > >> > >> On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins >wrote: > >> > >>> Hello FreeSWITCHers! > >>> > >>> We are looking for people who are in a position to help out with > various > >>> subprojects that will help FreeSWITCH to keep growing. We need people > to > >>> help out in these basic areas: > >>> > >>> Bug marshals (people who watch JIRA and test bug reports, patches, > etc.) > >>> Documentation maintainers (people who update the wiki when new stuff > comes > >>> out, also those familiar with mediawiki administration) > >>> Documentation authors (people who write new docs, how-to's, tutorials, > >>> examples, etc.) > >>> Package maintainers (people who manage Debian debs, RPMs, etc.) > >>> > >>> Additionally, we are always looking for more folks to assist with > >>> answering questions on IRC and the mailing list. It is definitely nice > to > >>> have people who've gone through the pains of switching to FreeSWITCH > (or > >>> learning it from scratch) who can assist the steady stream of new > users. > >>> > >>> If you want to help and aren't sure where to go from here then please > at > >>> least do the following: > >>> #1 - Join #freeswitch on irc.freenode.net and hang out as much as > >>> possible > >>> #2 - Check the recent changes link on wiki.freeswitch.org each day > >>> #3 - Join the Friday public conference call and listen in > >>> These three things, in addition to the mailing list, will keep you well > in > >>> tune with the FreeSWITCH community and what's happening. > >>> > >>> Next, make a note of the parts of FS that you use frequently, know a > lot > >>> about, or are particularly passionate about. Those are the items we'd > love > >>> to have you help us with. For example: if you use mod_xml_curl > frequently > >>> and have been through the set up process then you're a prime candidate > to > >>> help answer questions, refine the mod_xml_curl wiki documentation, > write up > >>> a tutorial, contribute a working example of a web server & database > schema, > >>> etc. If you are good with a scripting language then we could definitely > use > >>> help with rounding out the docs for your favorite language. We could > also > >>> use code samples, so ask for a contrib folder if you have things you > would > >>> like to share. Or how about this: you read something on the wiki, it > doesn't > >>> quite work when you try, so you tinker until you figure it out. Now > you're > >>> in a position to update the wiki for everyone else's benefit, too. > >>> > >>> As you can see, you don't have to be a FreeSWITCH expert before you can > >>> help the project. What we really need are people who care about the > project > >>> and want to see it flourish. If you are such a person then please > contact me > >>> off list. Tell me what you're good at or where you would like to help. > >>> > >>> Many thanks for all of your support! > >>> -Michael > >>> > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/521f3a14/attachment-0001.html From anthony.minessale at gmail.com Tue Sep 15 08:32:07 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Sep 2009 10:32:07 -0500 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> Message-ID: <191c3a030909150832n68b1e53cu8b15621507f8e6a2@mail.gmail.com> ok so the first job we should fill is someone to manage what the other volunteers do. like community czar We can make that person or persons a manager on jira and create projects for the various duties. Everyone can get an account if they don't already have one and take advantage of some of the workflow features in jira. On Tue, Sep 15, 2009 at 9:58 AM, Brian West wrote: > The friday meetings are where we all collaborate on these group efforts and > discuss project direction, goals and areas where people can help out more. > Did we ever find someone to officially take over the Debian packages? > > /b > > > > On Sep 15, 2009, at 9:56 AM, Michael Gende wrote: > > Answered my own question. Will email Diego off-list. > > On Tue, Sep 15, 2009 at 9:50 AM, Michael Gende wrote: > >> Hey Demuel, et al, >> >> I agree that without coordination, there will be duplication of effort, >> etc. >> >> How does one "find the line to get in" to help with, in my case, >> documentation? I don't want to, with the best of intentions, simply create >> confusion. >> >> I'm new, so if I've simply not yet RTFM that answers my question above, >> some kind person might nudge me in the right direction. >> >> Back to registering with my SIP provider... >> >> Regards, >> >> Mike G. >> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/6f12428b/attachment.html From msc at freeswitch.org Tue Sep 15 08:35:19 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Sep 2009 08:35:19 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call Agenda For Friday September 18 Message-ID: <87f2f3b90909150835q569a5ac3s2f2655adf8b6d18@mail.gmail.com> Hello, The weekly conference call agenda page is ready for everyone to contribute: http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_18 Please add questions, comments, concerns, and ideas. As always, we are looking for volunteers to assist with various sub-projects. All are invited to listen in on the weekly call. It's a great way to get a feel for what's happening with the project and to communicate directly with the core developers. Thanks again for your support and for working so hard to make FreeSWITCH a successful project and for making the FreeSWITCH community such a great place to be. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/7777a92a/attachment.html From tculjaga at gmail.com Tue Sep 15 08:45:03 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 15 Sep 2009 17:45:03 +0200 Subject: [Freeswitch-users] fax detection Message-ID: <65d96fc80909150845i5b287bc2w5a04537c836b1b48@mail.gmail.com> Hi, is there any way to route fax calls according to the call capability? I mean .. if the fax call supports T.38 i'd like to route it to a T.38 capable gateway. All other fax calls (meaning inband) should be handled by FS/SpanDSP. Of course, I know that every fax call starts as a voice call and upon fax tone detection additional capabilities are being negotiated(T.38 or G711). Can it be done in early media, before the call is even answered? So, here the goal is to have a T.38 capable GW handling T.38 calls while SpanDSP handling T.30... Any chance to do that with FS? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/3a8280c4/attachment.html From jerry.richards at teotech.com Tue Sep 15 08:45:53 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 15 Sep 2009 08:45:53 -0700 Subject: [Freeswitch-users] FS Presence Implementation Message-ID: <7DD13BC3900947FC8114068C3DCE96D7@greyhawk.tonecommander.com> I would like to modify my SIP phone and my gateway to convey/exchange presence information. Could someone point me toward the FS presence documentation? I've seen bits and pieces. Also, I think presence can be communicated via more than one protocol. Thanks And Best Regards, Jerry From fraunhofer.lists.freeswitch-001 at traced.net Tue Sep 15 10:08:37 2009 From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer) Date: Tue, 15 Sep 2009 19:08:37 +0200 Subject: [Freeswitch-users] sending custom events from event_socket, channel_variable_changed event? Message-ID: Hello *, while trying to figure out how to send custom events from mod_socket with "sendmsg" (like a telnet connection or something) i only found how to do that from within javascript (with e.fire() and stuff). So... first of all, how's the correct syntax to do that from the event_socket? the wiki states --- sendmsg Send a message to the call of given uuid (call-command execute or hangup), see examples below. --- Is there a secret syntax i missed how to send "CUSTOM" events? I tried to work-around that by setting a channel-variable and then calling the info-app. That way, a channel_execute event is fired and shows up in my event_socket-app. If there's a prettier way... please ignore the following up to the mark :) so it looked to me that i can only send "command" messages and looking at the sources i found that mod_socket will hard-wire the event-name to "SWITCH_EVENT_COMMAND" (in read_packet() in ./mod/event_handlers/mod_event_socket/mod_event_socket.c). If i understand correctly how e.g.. the mod_spidermonkey-bindings enqueue events, the event-name is taken from the user and can therefor be "CUSTOM" (looks like it's being looked up in the switch_event_types_t-enum). --- if (switch_name_event(ename, &etype) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "Unknown event %s\n", ename); *rval = BOOLEAN_TO_JSVAL(JS_FALSE); return JS_TRUE; } if (etype == SWITCH_EVENT_CUSTOM) { [...] if (switch_event_create_subclass(&event, etype, subclass_name) != SWITCH_STATUS_SUCCESS) { --- If it's currently not possible, do you think this could be useful, too? I thought a bit about the syntax and came up with this idea to not break compatibility, but feel free to change that, you know it better :) if you want to send an event that's not COMMAND, use "event-name: CUSTOM" as the first line after the "sendmsg [uuid]" and change the event-name (if that's possible) of the event already created by read_packet() or reconstruct the event or ... Just another idea..... that could become handy is if there would be an event "CHANNEL_VAR_CHANGED", fired/enqueued by setvar(), setvar_multi() and friends. But as i write this i admit that one will be overwhelmed by the setvar massacre for example bridge() would do to set "disposition"-variables... Just a thought. Cheers Beni. From demuel at thephinix.org Tue Sep 15 10:11:01 2009 From: demuel at thephinix.org (demuel at thephinix.org) Date: Tue, 15 Sep 2009 18:11:01 +0100 Subject: [Freeswitch-users] FS Presence Implementation In-Reply-To: <7DD13BC3900947FC8114068C3DCE96D7@greyhawk.tonecommander.com> References: <7DD13BC3900947FC8114068C3DCE96D7@greyhawk.tonecommander.com> Message-ID: <779e0ed9572f0e9107a54028e906d228.squirrel@www.thephinix.org> I think you better start with the libjingle module for this one. > I would like to modify my SIP phone and my gateway to convey/exchange > presence information. Could someone point me toward the FS presence > documentation? I've seen bits and pieces. Also, I think presence can be > communicated via more than one protocol. > > Thanks And Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From andrew at hijacked.us Tue Sep 15 11:29:14 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 15 Sep 2009 14:29:14 -0400 Subject: [Freeswitch-users] ERLang configuration callbacks In-Reply-To: <4AAE9F77.6090700@marketelsystems.com> References: <4AAE9F77.6090700@marketelsystems.com> Message-ID: <20090915182913.GB20978@hijacked.us> On Mon, Sep 14, 2009 at 01:54:31PM -0600, Mark Sobkow wrote: > I seem to be missing "something" in implementing the ERLang callbacks > for Freeswitch. Our Freeswitch server is starting and getting > registered with ERLang, we're invoking the bind for configuration, but > I'm not seeing any of my callbacks fire. What am I missing? > The most obvious thing is that you're trying to catch info messages using handle_call. The erlang module doesn't use the OTP protocol for messages so handle_call/cast won't ever fire for messages sent from the freeswitch module. Andrew From jerry.richards at teotech.com Tue Sep 15 11:30:58 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Tue, 15 Sep 2009 11:30:58 -0700 Subject: [Freeswitch-users] FS Presence Implementation Message-ID: <65AA86A7106D4255A5734520F55D3C50@greyhawk.tonecommander.com> Also, is presence conveyed as any string? Or is presence a predefined list of status? Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Tuesday, September 15, 2009 8:46 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: FS Presence Implementation I would like to modify my SIP phone and my gateway to convey/exchange presence information. Could someone point me toward the FS presence documentation? I've seen bits and pieces. Also, I think presence can be communicated via more than one protocol. Thanks And Best Regards, Jerry From andrew at hijacked.us Tue Sep 15 11:39:35 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 15 Sep 2009 14:39:35 -0400 Subject: [Freeswitch-users] ERLang configuration callbacks In-Reply-To: <20090915182913.GB20978@hijacked.us> References: <4AAE9F77.6090700@marketelsystems.com> <20090915182913.GB20978@hijacked.us> Message-ID: <20090915183935.GC20978@hijacked.us> On Tue, Sep 15, 2009 at 02:29:14PM -0400, Andrew Thompson wrote: > On Mon, Sep 14, 2009 at 01:54:31PM -0600, Mark Sobkow wrote: Oh wait, I see what you're doing, you catch all the fetch requests in the handle_info and then make a call to another process to get the XML. My question is why are those both in the same process? The handle_info part is fine, but then the pid you make a gen_server:call to *must* be a different process or you'll hit a timeout and the process will exit. However, I can't start 2 copies of that process (one to catch requests, one to return XML) because your module always does a bind! Regardless, I at least get fetch requests when I run your module, I can't make it return something without refactoring it, but it does receive the requests at least. Andrew From anthony.minessale at gmail.com Tue Sep 15 11:53:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Sep 2009 13:53:20 -0500 Subject: [Freeswitch-users] FS Presence Implementation In-Reply-To: <65AA86A7106D4255A5734520F55D3C50@greyhawk.tonecommander.com> References: <65AA86A7106D4255A5734520F55D3C50@greyhawk.tonecommander.com> Message-ID: <191c3a030909151153i485e5f7v876db4cb6e95600a@mail.gmail.com> the default config ships with presence enabled for SIP if you have a phone that supports it, all you have to do is enable it on the phone. On Tue, Sep 15, 2009 at 1:30 PM, Jerry Richards wrote: > > Also, is presence conveyed as any string? Or is presence a predefined list > of status? > > Best Regards, > Jerry > > > -----Original Message----- > From: Jerry Richards [mailto:jerry.richards at teotech.com] > Sent: Tuesday, September 15, 2009 8:46 AM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: FS Presence Implementation > > I would like to modify my SIP phone and my gateway to convey/exchange > presence information. Could someone point me toward the FS presence > documentation? I've seen bits and pieces. Also, I think presence can be > communicated via more than one protocol. > > Thanks And Best Regards, > Jerry > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/d01223fb/attachment.html From anthony.minessale at gmail.com Tue Sep 15 12:19:36 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Sep 2009 14:19:36 -0500 Subject: [Freeswitch-users] sending custom events from event_socket, channel_variable_changed event? In-Reply-To: References: Message-ID: <191c3a030909151219y3ba5e5cp6de1d7bd745830e4@mail.gmail.com> I think we can't do it just because nobody needed it. I added a patch to r14874 to allow you to add a "unique-id" header to the sendevent command that should allow you to address and event right to a particular session rather than fire the event. On Tue, Sep 15, 2009 at 12:08 PM, Benedikt Fraunhofer < fraunhofer.lists.freeswitch-001 at traced.net> wrote: > Hello *, > > while trying to figure out how to send custom events from mod_socket > with "sendmsg" (like a telnet connection or something) i only found > how to do that from within javascript (with e.fire() and stuff). > > So... first of all, how's the correct syntax to do that from the > event_socket? the wiki states > --- > sendmsg > > Send a message to the call of given uuid (call-command execute or > hangup), see examples > below. > --- > > > Is there a secret syntax i missed how to send "CUSTOM" events? > > I tried to work-around that by setting a channel-variable and then > calling the info-app. That way, a channel_execute event is fired and > shows up in my event_socket-app. > If there's a prettier way... please ignore the following up to the > mark :) > > so it looked to me that i can only send "command" messages and looking > at the sources i found that mod_socket will hard-wire the event-name > to "SWITCH_EVENT_COMMAND" (in read_packet() in > ./mod/event_handlers/mod_event_socket/mod_event_socket.c). If i > understand correctly how e.g.. the mod_spidermonkey-bindings enqueue > events, the event-name is taken from the user and can therefor be > "CUSTOM" (looks like it's being looked up in the > switch_event_types_t-enum). > --- > if (switch_name_event(ename, &etype) != > SWITCH_STATUS_SUCCESS) { > switch_log_printf(SWITCH_CHANNEL_LOG, > SWITCH_LOG_WARNING, "Unknown event %s\n", ename); > *rval = BOOLEAN_TO_JSVAL(JS_FALSE); > return JS_TRUE; > } > > if (etype == SWITCH_EVENT_CUSTOM) { > [...] > if (switch_event_create_subclass(&event, etype, subclass_name) != > SWITCH_STATUS_SUCCESS) { > > --- > > > If it's currently not possible, do you think this could be useful, too? > I thought a bit about the syntax and came up with this idea to not > break compatibility, but feel free to change that, you know it better > :) > if you want to send an event that's not COMMAND, use "event-name: > CUSTOM" as the first line after the "sendmsg [uuid]" and change the > event-name (if that's possible) of the event already created by > read_packet() or reconstruct the event or ... > > > > Just another idea..... that could become handy is if there would be an > event "CHANNEL_VAR_CHANGED", fired/enqueued by setvar(), > setvar_multi() and friends. But as i write this i admit that one will > be overwhelmed by the setvar massacre for example bridge() would do to > set "disposition"-variables... Just a thought. > > Cheers > Beni. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/32679bc2/attachment.html From msc at freeswitch.org Tue Sep 15 12:51:25 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Sep 2009 12:51:25 -0700 Subject: [Freeswitch-users] I got ERLang to fire a configuration request In-Reply-To: <4AAFA6F1.8050802@marketelsystems.com> References: <4AAFA6F1.8050802@marketelsystems.com> Message-ID: <87f2f3b90909151251x2588da0em73a284238b4fe277@mail.gmail.com> Mark, You might want to send this question to freeswitch-dev at lists.freeswitch.orgas it's a bit intense for the users list. :) -MC On Tue, Sep 15, 2009 at 7:38 AM, Mark Sobkow wrote: > I still need to stuff the Freeswitch PID into global storage somewhere so > the process that's handling the configuration requests can send the reply > without crashing (it's just getting a node id, not a Pid), but I seem to be > on my way to configuring Freeswitch via ERLang. > > freeswitch_bind.erl has the calls added to register the ERLang callbacks. > The callback function itself is in the aptly named freeswitch_callback.erl. > > -module(freeswitch_bind). > > -behaviour(gen_server). > > -record(st, {fsnode, pbxpid, configpid, dirpid, dialpid}). > > -export([start/3, terminate/2, code_change/3, init/1, > handle_call/3, handle_cast/2, handle_info/2]). > > %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%% > %% gen_server methods > start(Node, Section, Pid) -> > gen_server:start(?MODULE, [Node, Section, Pid], []). > > init([Node, Section, Pid]) -> > io:format( "freeswitch_bind:init( [Node=~w, Section=~w, Pid=~w])~n", > [Node, Section, Pid] ), > {api, Node} ! {bind, Section}, > receive > ok -> > {ok, ConfigurationPid } = freeswitch:start_fetch_handler( > Node, configuration, freeswitch_callback, fetch_handler ), > {ok, DirectoryPid } = freeswitch:start_fetch_handler( Node, > directory, freeswitch_callback, fetch_handler ), > {ok, DialplanPid } = freeswitch:start_fetch_handler( Node, > dialplan, freeswitch_callback, fetch_handler ), > {ok, #st{fsnode=Node, pbxpid=Pid, configpid=ConfigurationPid, > dirpid=DirectoryPid, dialpid=DialplanPid}}; > {error, Reason} -> > {stop, {error, {freeswitch_error, Reason}}} > after 5000 -> > {stop, {error, freeswitch_timeout}} > end. > > terminate(_Reason, _State) -> > ok. > > code_change(_OldVsn, State, _Extra) -> > {ok, State}. > > %% > %% If the request isn't recognized, just log it and do nothing. > %% > handle_call(Request, _From, State) -> > io:format("freeswitch_bind:handle_call( ~w, _From, State) unrecognized > request~n", > [Request]), > {reply, {error, unrecognized_request}, State}. > > handle_cast(Message, State) -> > error_logger:error_msg("~p received unrecognized cast ~p~n", > [self(), Message]), > {noreply, State}. > > handle_info({fetch, Section, Tag, Key, Value, FetchID, Params}, > #st{fsnode=Node, pbxpid=Pid}=State) -> > {ok, XML} = gen_server:call(Pid, {fetch, Section, Tag, Key, Value, > Params}), > {api, Node} ! {fetch_reply, FetchID, XML}, > receive > ok -> > {noreply, State}; > {error, Reason} -> > {stop, {error, Reason}, State} > end. > > %% Author: mark > %% Created: Sep 15, 2009 > %% Description: TODO: Add description to freeswitch_callback > -module(freeswitch_callback). > > -behaviour(gen_server). > > -record(st, {fsnode, pbxpid}). > > -export([start/3, terminate/2, code_change/3, init/1, > handle_call/3, handle_cast/2, handle_info/2, fetch_handler/1]). > > start(Node, Section, Pid) -> > gen_server:start(?MODULE, [Node, Section, Pid], []). > > init([Node, Section, Pid]) -> > io:format( "freeswitch_callback:init( [Node=~w, Section=~w, > Pid=~w])~n", [Node, Section, Pid] ), > {ok, #st{fsnode=Node, pbxpid=Pid}}. > > terminate(_Reason, _State) -> > ok. > > code_change(_OldVsn, State, _Extra) -> > {ok, State}. > > %% > %% Callback for freeswitch:start_fetch_handler() called in > freeswitch_bind:init() > %% > fetch_handler( FreeswitchNode ) -> > receive > { nodedown, Node } -> > io:format( "freeswitch_callback:fetch_handler() Node > ~w is down~n", [Node] ), > ok; > { fetch, Section, Tag, Key, Value, FetchId, Params } -> > io:format( "freeswitch_callback:fetch_handler() > Invoking xml_fetch()~n" ), > {ok, Xml} = xml_fetch( {fetch, Section, Tag, Key, Value, > Params} ), > io:format( "freeswitch_callback:fetch_handler() > Sending reply to FreeswitchNode ~w: ~s~n", [FreeswitchNode, Xml] ), > FreeswitchNode ! { fetch_reply, FetchId, Xml }, > io:format( "freeswitch_callback:fetch_handler() > Reply sent~n" ), > ok > end, > { ok } = fetch_handler( FreeswitchNode ), > { ok }. > > %% > %% Configuration handler replies that the requested document section, > tag, and key are not > %% found. > %% > xml_fetch({fetch, configuration, Tag, Key, Value, Params}) -> > io:format( "freeswitch_callback:handle_call( {fetch, configuration, > Tag=~s, Key=~s, Value=~s, Params=~w} )~n", > [Tag, Key, Value, Params]), > Xml = > " >
> >
> ", > {ok, Xml }; > > %% > %% Directory handler replies that the requested document section, tag, > and key are not > %% found. > %% > xml_fetch({fetch, directory, Tag, Key, Value, Params}) -> > io:format( "freeswitch_callback:xml_fetch( {fetch, directory, > Tag=~s, Key=~s, Value=~s, Params=~w} )~n", > [Tag, Key, Value, Params]), > Xml = > " >
> >
> ", > {ok, Xml }; > > %% > %% Dialplan handler replies that the requested document section, tag, > and key are not > %% found. > %% > xml_fetch({fetch, dialplan, Tag, Key, Value, Params}) -> > io:format( "freeswitch_callback:xml_fetch( {fetch, dialplan, Tag=~s, > Key=~s, Value=~s, Params=~w} )~n", > [Tag, Key, Value, Params]), > Xml = > " >
> >
> ", > {ok, Xml }; > > %% > %% Default handler replies that the requested document section, tag, > and key are not > %% found. > %% > xml_fetch({fetch, Section, Tag, Key, Value, Params}) -> > io:format( "freeswitch_callback:xml_fetch( {fetch, Section=~w, > Tag=~s, Key=~s, Value=~s, Params=~w} )~n", > [Section, Tag, Key, Value, Params]), > Xml = > " >
> >
> ", > {ok, Xml }; > > %% > %% If the request isn't recognized, just log it. > %% > xml_fetch( Request ) -> > io:format( "freeswitch_callback:xml_fetch( Request=~w ) not > recognized~n", > [Request]), > Xml = > " >
> >
> ", > {ok, Xml }. > > > %% > %% If the request isn't recognized, just log it and do nothing. > %% > handle_call(Request, _From, State) -> > io:format("freeswitch_callback:handle_call( ~w, _From, State) > unrecognized request~n", > [Request]), > {reply, {error, unrecognized_request}, State}. > > handle_cast(Message, State) -> > error_logger:error_msg("~p received unrecognized cast ~p~n", > [self(), Message]), > {noreply, State}. > > handle_info({fetch, Section, Tag, Key, Value, FetchID, Params}, > #st{fsnode=Node, pbxpid=Pid}=State) -> > {ok, XML} = gen_server:call(Pid, {fetch, Section, Tag, Key, Value, > Params}), > {api, Node} ! {fetch_reply, FetchID, XML}, > receive > ok -> > {noreply, State}; > {error, Reason} -> > {stop, {error, Reason}, State} > end. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/ce6325b2/attachment-0001.html From andrew at hijacked.us Tue Sep 15 13:30:22 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 15 Sep 2009 16:30:22 -0400 Subject: [Freeswitch-users] I got ERLang to fire a configuration request In-Reply-To: <4AAFA6F1.8050802@marketelsystems.com> References: <4AAFA6F1.8050802@marketelsystems.com> Message-ID: <20090915203021.GD20978@hijacked.us> On Tue, Sep 15, 2009 at 08:38:41AM -0600, Mark Sobkow wrote: > I still need to stuff the Freeswitch PID into global storage somewhere > so the process that's handling the configuration requests can send the > reply without crashing (it's just getting a node id, not a Pid), but I > seem to be on my way to configuring Freeswitch via ERLang. > Looking at your code, assuming I read it right, you should be able to just replace: FreeSWITCHNode ! SomeMsg. with {api, FreeSWITCHNode} ! SomeMsg. The freeswitch module doesn't really have a pid, since it's not a real erlang node, it's all faked and all messages to a pid or a registered process on the C node go to the same place. Andrew From djbinter at yahoo.com Tue Sep 15 13:38:24 2009 From: djbinter at yahoo.com (DJB) Date: Tue, 15 Sep 2009 13:38:24 -0700 (PDT) Subject: [Freeswitch-users] Sip Allow Options Message-ID: <185098.74411.qm@web37505.mail.mud.yahoo.com> I wonder whether anyone can tell me why the latest trunk has no PRACK comparing to the 1.0.4. Here is the sip message: User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14877 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Compare to: User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Thank you, Dorn B. From miles.chet at gmail.com Tue Sep 15 13:46:54 2009 From: miles.chet at gmail.com (roberto) Date: Tue, 15 Sep 2009 17:46:54 -0300 Subject: [Freeswitch-users] A FreeSWITCH Calling Card Application written in Ruby In-Reply-To: <86a32abc0909061857u3bff1a3bkb16400f03f9b980c@mail.gmail.com> References: <86a32abc0909061836j4a4d756bnf5f1e675e1607cd6@mail.gmail.com> <86a32abc0909061857u3bff1a3bkb16400f03f9b980c@mail.gmail.com> Message-ID: Hello, Someone could tell me what happens to the project, it seems that is no longer available in github ? http://github.com/diego/freeswitch-card/ thanks, On Sun, Sep 6, 2009 at 10:57 PM, Diego Viola wrote: > I also have plans to add a GUI later, maybe I will merge my code and turn it > into a ramaze app, but it should be usable right now. > > Regards, > > Diego > > > On Mon, Sep 7, 2009 at 1:36 AM, Diego Viola wrote: >> >> Hello, >> >> I'm currently working on a calling card application written in Ruby, just >> a hobby, I currently have it on a usable state and I thought I would post it >> here in case if there is someone interested. >> >> It uses mod_nibblebill as the billing/rate engine and FSR (FreeSWITCHeR) >> as the event socket library. >> >> Here is the url of the project: >> >> http://github.com/diego/freeswitch-card/tree/master >> >> You can find a clone of the project on my FreeSWITCH contrib directory >> too. >> >> http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/diegoviola/ >> >> Regards, >> >> Diego >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From brian at freeswitch.org Tue Sep 15 13:55:32 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 15 Sep 2009 15:55:32 -0500 Subject: [Freeswitch-users] Sip Allow Options In-Reply-To: <185098.74411.qm@web37505.mail.mud.yahoo.com> References: <185098.74411.qm@web37505.mail.mud.yahoo.com> Message-ID: Because you have 100rel disabled so PRACK will NOT show up in the allow list. /b On Sep 15, 2009, at 3:38 PM, DJB wrote: > I wonder whether anyone can tell me why the latest trunk has no > PRACK comparing to the 1.0.4. > > Here is the sip message: > > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14877 > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > > Compare to: > > User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH > > > Thank you, > Dorn B. > From fraunhofer.lists.freeswitch-001 at traced.net Tue Sep 15 13:57:54 2009 From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer) Date: Tue, 15 Sep 2009 22:57:54 +0200 Subject: [Freeswitch-users] sending custom events from event_socket, channel_variable_changed event? In-Reply-To: <191c3a030909151219y3ba5e5cp6de1d7bd745830e4@mail.gmail.com> References: <191c3a030909151219y3ba5e5cp6de1d7bd745830e4@mail.gmail.com> Message-ID: Hi Anthony, 2009/9/15 Anthony Minessale : > I added a patch to r14874 to allow you to add a "unique-id" header to the > sendevent command > that should allow you to address and event right to a particular session > rather than fire the event. thx! While riding the train back from work i thought i could've missed that what i wanted to achieve was possible with sendevent instead of sendmsg but your reply made me calm down :) I won't be able to give it a try until this Friday, but I'll report back if this is the "new" way to send CUSTOM events from mod_event to mod_event and pay the wiki tax, accordingly :) Thx again! Beni. From email.list.subscriber at gmail.com Tue Sep 15 14:53:14 2009 From: email.list.subscriber at gmail.com (email lists) Date: Tue, 15 Sep 2009 17:53:14 -0400 Subject: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? In-Reply-To: <191c3a030909110831o4e0a4844obf3c339d58f5358a@mail.gmail.com> References: <4aa965af.161bf30a.61f8.7c7d@mx.google.com> <4AA9FBA5.5090403@kounitskiy.com> <191c3a030909110831o4e0a4844obf3c339d58f5358a@mail.gmail.com> Message-ID: <4ab00c82.161bf30a.6722.522e@mx.google.com> Thanks to those for the info and help on this issue. Ultimately ended up having to use alternative software for the radius piece (not related to any shortfalls by Freeswitch). Vlad From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, September 11, 2009 11:31 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? set the variable process_cdr=false on that a_leg first thing in your dialplan On Fri, Sep 11, 2009 at 2:26 AM, Anatoliy Kounitskiy wrote: It's normal to have to two records for a call - Start and Stop message. From what i see - you have one start and stop for each leg of the call. Regards, AK email lists wrote: > > Hello, > > > > Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate > RADIUS messages being generated for individual calls (sample messages > for one call below). Looking at the "Acct-Unique-Session-Id" and > "Acct-Session-Id" fields, it would appear that perhaps each call leg > results in a pair of start/stop RADIUS messages; is this the expected > behavior? If so, is there a way to disable RADIUS messaging for what > I presume is the "ingress" or A leg of the call? > > > > Any leads would be appreciated. > > > > Thanks in advance. > > > > Vladimir > > > > Thu Sep 10 10:37:25 2009 > > Acct-Status-Type = Start > > Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "097c8472ff7bcec7" > > Timestamp = 1252604245 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:37:25 2009 > > Acct-Status-Type = Start > > Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226 at x.x.x.x" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "53f729e173e8c0a9" > > Timestamp = 1252604245 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:37:57 2009 > > Acct-Status-Type = Stop > > Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" > > Freeswitch-Hangupcause = Normal-Unspecified > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Lastapp = "bridge" > > Freeswitch-Billusec = 32029926 > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" > > Freeswitch-Callanswerdate = "2009-09-10T10:22:00.319197-0700" > > Freeswitch-Callenddate = "2009-09-10T10:22:32.349123-0700" > > Acct-Session-Time = 32 > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "097c8472ff7bcec7" > > Timestamp = 1252604277 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:38:02 2009 > > Acct-Status-Type = Stop > > Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" > > Freeswitch-Hangupcause = Normal-Clearing > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226 at x.x.x.x" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Billusec = 32049973 > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" > > Freeswitch-Callanswerdate = "2009-09-10T10:22:00.289136-0700" > > Freeswitch-Callenddate = "2009-09-10T10:22:32.339109-0700" > > Acct-Session-Time = 32 > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "53f729e173e8c0a9" > > Timestamp = 1252604282 > > Request-Authenticator = Verified** > > > > ---------------------------------------------------------------------- -- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/1297d3f3/attachment-0001.html From msc at freeswitch.org Tue Sep 15 14:57:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Sep 2009 14:57:23 -0700 Subject: [Freeswitch-users] stange segfaults with log In-Reply-To: <4AAF6C83.2080203@gmail.com> References: <4AAF6C83.2080203@gmail.com> Message-ID: <87f2f3b90909151457s79464defh8fb0e9f6128845d8@mail.gmail.com> On Tue, Sep 15, 2009 at 3:29 AM, Tamas Cseke wrote: > Hello > > we have a strange problem with 14144 revision. It seems > switch_log_printf got NULL pointer as data. It happens a few times. > however in the previous frame session seems to be good for us. > http://pastebin.freeswitch.org/10357 > > Could you please tell me what is the problem? Did we make some mistakes > with building? Missed "make clean", or someting? > > Well, for one thing you are more than 800 revs behind current SVN. I strongly recommend you "make current" and let the system get properly updated. -MC > > Thanks in advance, > Tamas > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/11f6c278/attachment.html From email.list.subscriber at gmail.com Tue Sep 15 15:22:11 2009 From: email.list.subscriber at gmail.com (email lists) Date: Tue, 15 Sep 2009 18:22:11 -0400 Subject: [Freeswitch-users] 502 Bad Gateway: Destination out of order error Message-ID: <4ab0134e.141bf30a.483c.0b7c@mx.google.com> Hello All, Wondering if anyone has experienced this issue before. I've attached a snip of the log file where the error occurs and could use some leads on this. What's interesting is that the call appears to complete as normal, and a radius stop message even gets generated, though the duration is ~1 second. h323-disconnect-time = "h323-disconnect-time=14:35:01.000 UTC Tue Sep 15 2009" h323-connect-time = "h323-connect-time=14:34:59.000 UTC Tue Sep 15 2009" While there are a lot of pieces involved, the call scenario is pretty basic (no transfers, no holds, etc.), just a few redirects that Freeswitch appears to be able to handle without issue. Attached is a dumb'd down call ladder. I tried different rates at which I generate the calls, but it didn't seem to correlate to the amount of errors I am seeing. Sending a total of 100 calls, with a call duration of 10 seconds: @10 calls per second = 14 "502" errors. @5 calls per second = 4 "502" errors. @4 calls per second = NO ERRORS (1st run) @4 calls per second = 39 "502" errors. Please let me know if any additional information is needed. Thanks in advance for all help. Vladimir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/3b6288bd/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: fs.log Type: application/octet-stream Size: 4894 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/3b6288bd/attachment-0001.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: ladder.png Type: image/png Size: 12600 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/3b6288bd/attachment-0001.png From anthony.minessale at gmail.com Tue Sep 15 15:32:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 15 Sep 2009 17:32:13 -0500 Subject: [Freeswitch-users] 502 Bad Gateway: Destination out of order error In-Reply-To: <4ab0134e.141bf30a.483c.0b7c@mx.google.com> References: <4ab0134e.141bf30a.483c.0b7c@mx.google.com> Message-ID: <191c3a030909151532i286111dawa8f4f19e0b82985e@mail.gmail.com> It's probably from this 2009-09-15 14:34:55.838365 [ERR] sofia_glue.c:2503 AUDIO RTP REPORTS ERROR: [Socket Error!] 2009-09-15 14:34:55.838365 [NOTICE] sofia_glue.c:2504 Hangup sofia/external/ 411 at 192.168.0.150 [CS_CONSUME_MEDIA][DESTINATION_OUT_OF_ORDER ] 2009-09-15 14:34:55.838365 [ERR] sofia.c:3796 RTP Error! your machine failed to produce a socket when requested. My blind guess, you are on a 32 bit machine and you do not have the ulimits set for enough file descriptors etc.. ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 999999 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 244 ulimit -l unlimited On Tue, Sep 15, 2009 at 5:22 PM, email lists < email.list.subscriber at gmail.com> wrote: > Hello All, > > > > Wondering if anyone has experienced this issue before. I've attached a > snip of the log file where the error occurs and could use some leads on > this. What's interesting is that the call appears to complete as normal, > and a radius stop message even gets generated, though the duration is ~1 > second. > > > > > > h323-disconnect-time = "h323-disconnect-time=14:35:01.000 UTC Tue Sep 15 > 2009" > > h323-connect-time = "h323-connect-time=14:34:59.000 UTC Tue Sep 15 2009" > > > > > > While there are a lot of pieces involved, the call scenario is pretty basic > (no transfers, no holds, etc.), just a few redirects that Freeswitch appears > to be able to handle without issue. Attached is a dumb'd down call ladder. > > > > I tried different rates at which I generate the calls, but it didn?t seem > to correlate to the amount of errors I am seeing. > > > > Sending a total of 100 calls, with a call duration of 10 seconds: > > @10 calls per second = 14 "502" errors. > > @5 calls per second = 4 "502" errors. > > @4 calls per second = NO ERRORS (1st run) > > @4 calls per second = 39 "502" errors. > > > > Please let me know if any additional information is needed. > > > > Thanks in advance for all help. > > > > Vladimir > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/12ef7bf6/attachment.html From aep.lists at it46.se Tue Sep 15 15:37:01 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Wed, 16 Sep 2009 00:37:01 +0200 Subject: [Freeswitch-users] DTMF CSeq: 505 INFO vs RTP DTMF In-Reply-To: <9ece78af3aedde36e10768727ab099cd.squirrel@correo.nodo50.org> References: <9ece78af3aedde36e10768727ab099cd.squirrel@correo.nodo50.org> Message-ID: After digging into this issue, it might the case that the implementation of out-bound DTMF of the client i am using does not properly increments CSeq per DTMF. For those interested, i am currently integrating OpenBTS with Freeswitch! :) -aep -- Stopping junk mailers is good for the environment > Hi, > > I am using the function session.collectInput and session.streamFile to > collect a number of DTMF digits. > If the DTMF digits are sent in the RTP, i can collect several digits until > timeout. No problem there! If the DTMFs are received as a sequence of SIP > INFO packages, collectInput only receives the first one. > > Any ideas? > > > > > > -- > Stopping junk mailers is good for the environment > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From dujinfang at gmail.com Tue Sep 15 16:37:09 2009 From: dujinfang at gmail.com (Seven Du) Date: Wed, 16 Sep 2009 07:37:09 +0800 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <87f2f3b90909150825o472ecf23h4088668266051d5a@mail.gmail.com> References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> <87f2f3b90909150825o472ecf23h4088668266051d5a@mail.gmail.com> Message-ID: <563C8F4C-C0F1-4583-A003-DF0B65B373CB@gmail.com> Hi MC, Months ago we had tried the multi-language plugin on MediaWiki, I know you are still planning to do this but I just want how far it goes. Count me in when you are short of hand. On Sep 15, 2009, at 11:25 PM, Michael Collins wrote: > Demuel, > Thanks for the input. Yes, we want to avoid chaos. I will work to > keep everyone organized. I would welcome a mass of people all trying > to do different things because the challenge would simply be to keep > them organized. Right now the challenge is in recruiting people to > stay with the not-so-glorious aspects of the project, namely > documentation, maintenance, and janitorial style subprojects. > > Please keep the comments and suggestions coming! > > Thanks, > MC > > On Tue, Sep 15, 2009 at 12:38 AM, wrote: > Hi Michael / FS enthusiast, > > In my opinion, it will be an undeniably perfect chaos if everyone > wants to do everything. One way > I can suggest out of this is the following: > > - sets of persons that does all the stuff on the FS core. I'm not > sure if this one will work out but > I think these are the only persons who has commit access to the > source code and will also be willing > to accept modifications, enhancements, etc. > > - individuals can ask if they can be the maintainer/tester for a > particular module. Bug fixing, > sending modifications and enhancements will still be subject to the > approval of anthm. > > - an array of persons that will take ownership on what FS can do and > provide realiable working > examples for it. Currently, it takes amount of pain for a newbie on > how to configure SIP because > the examples are too confusing and sometimes there is not certainty > if this will work on what > release of FS or not. > > - any individual that has time and interest can start owning the > porting of FS to other operating > system. Like in my case, I am working on making FS ported to FreeBSD > ports but I don't know if > somebody did it or if he is currently doing it, as to what stage he > is in? > > - there should be a release engineering team. In my opinion, a > stable and current release will be > much more sane. > > As with any other successful open source project, we won't be > starting to say I want to contribute > this and start working on that. We should indicate that you take > ownership of this and any > comments should be forwarded right unto you. > > Again, this is just my opinion. You can take some of it or leave it > using "sudo rm -rf blah/*" . > > There and back again, > Demuel I. Bendano > a.k.a engrxyz > > > > Hi Michael, > > > > You can count with me for anything else, like documentation, > > coding/scripting, or any other FreeSWITCH related stuff. > > > > Regards, > > > > Diego > > > > 2009/9/14 Jo?o Mesquita > > > >> You can assign two things to me. > >> > >> 1. libesl code documentation (partially done and Doxygened - needs > >> cleaning) > >> 2. Bug marshal. I am setting up the proper lab environment here > to be able > >> to test most stuff. > >> > >> Count me in for any questions I can answer and I am _always_ on IRC > >> > >> jmesquita > >> > >> On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins >wrote: > >> > >>> Hello FreeSWITCHers! > >>> > >>> We are looking for people who are in a position to help out with > various > >>> subprojects that will help FreeSWITCH to keep growing. We need > people to > >>> help out in these basic areas: > >>> > >>> Bug marshals (people who watch JIRA and test bug reports, > patches, etc.) > >>> Documentation maintainers (people who update the wiki when new > stuff comes > >>> out, also those familiar with mediawiki administration) > >>> Documentation authors (people who write new docs, how-to's, > tutorials, > >>> examples, etc.) > >>> Package maintainers (people who manage Debian debs, RPMs, etc.) > >>> > >>> Additionally, we are always looking for more folks to assist with > >>> answering questions on IRC and the mailing list. It is > definitely nice to > >>> have people who've gone through the pains of switching to > FreeSWITCH (or > >>> learning it from scratch) who can assist the steady stream of > new users. > >>> > >>> If you want to help and aren't sure where to go from here then > please at > >>> least do the following: > >>> #1 - Join #freeswitch on irc.freenode.net and hang out as much as > >>> possible > >>> #2 - Check the recent changes link on wiki.freeswitch.org each day > >>> #3 - Join the Friday public conference call and listen in > >>> These three things, in addition to the mailing list, will keep > you well in > >>> tune with the FreeSWITCH community and what's happening. > >>> > >>> Next, make a note of the parts of FS that you use frequently, > know a lot > >>> about, or are particularly passionate about. Those are the items > we'd love > >>> to have you help us with. For example: if you use mod_xml_curl > frequently > >>> and have been through the set up process then you're a prime > candidate to > >>> help answer questions, refine the mod_xml_curl wiki > documentation, write up > >>> a tutorial, contribute a working example of a web server & > database schema, > >>> etc. If you are good with a scripting language then we could > definitely use > >>> help with rounding out the docs for your favorite language. We > could also > >>> use code samples, so ask for a contrib folder if you have things > you would > >>> like to share. Or how about this: you read something on the > wiki, it doesn't > >>> quite work when you try, so you tinker until you figure it out. > Now you're > >>> in a position to update the wiki for everyone else's benefit, too. > >>> > >>> As you can see, you don't have to be a FreeSWITCH expert before > you can > >>> help the project. What we really need are people who care about > the project > >>> and want to see it flourish. If you are such a person then > please contact me > >>> off list. Tell me what you're good at or where you would like to > help. > >>> > >>> Many thanks for all of your support! > >>> -Michael > >>> > >>> > >>> > >>> _______________________________________________ > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Tue Sep 15 16:59:05 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Sep 2009 16:59:05 -0700 Subject: [Freeswitch-users] DTMF CSeq: 505 INFO vs RTP DTMF In-Reply-To: References: <9ece78af3aedde36e10768727ab099cd.squirrel@correo.nodo50.org> Message-ID: <87f2f3b90909151659w56e7b605j4e61191c61473ac5@mail.gmail.com> On Tue, Sep 15, 2009 at 3:37 PM, Alberto Escudero wrote: > After digging into this issue, it might the case that the implementation > of out-bound DTMF of the client i am using does not properly increments > CSeq per DTMF. > > For those interested, i am currently integrating OpenBTS with Freeswitch! > :) > > -aep > We are very interested in seeing how this pans out. Please keep us posted on your progress and definitely come back when you have questions. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/296ec2be/attachment.html From msc at freeswitch.org Tue Sep 15 17:00:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 15 Sep 2009 17:00:10 -0700 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <563C8F4C-C0F1-4583-A003-DF0B65B373CB@gmail.com> References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> <87f2f3b90909150825o472ecf23h4088668266051d5a@mail.gmail.com> <563C8F4C-C0F1-4583-A003-DF0B65B373CB@gmail.com> Message-ID: <87f2f3b90909151700t6c93420euee612e88be8d1334@mail.gmail.com> On Tue, Sep 15, 2009 at 4:37 PM, Seven Du wrote: > Hi MC, > > Months ago we had tried the multi-language plugin on MediaWiki, I know > you are still planning to do this but I just want how far it goes. > Count me in when you are short of hand. > > :) Thanks, we are still tinkering. I haven't found anyone who is familiar with the multilang extension of mediawiki, so I'm learning it myself. I'll keep you updated on my progress. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/0c24f4b2/attachment-0001.html From nandy1925 at gmail.com Tue Sep 15 19:04:14 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 16 Sep 2009 10:04:14 +0800 Subject: [Freeswitch-users] FS 1.0.4 erl configure error In-Reply-To: <7d0bfd8c0909132320j1b2d1113j78da26b07a37e526@mail.gmail.com> References: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> <4AAD1611.2030803@gmail.com> <7d0bfd8c0909132320j1b2d1113j78da26b07a37e526@mail.gmail.com> Message-ID: <7d0bfd8c0909151904u2f3c35f0l36ab5d99f89b7845@mail.gmail.com> hi folks, anyone encountered this problem? tks. /nandy On Mon, Sep 14, 2009 at 2:20 PM, Nandy Dagondon wrote: > meftah, > > i disabled mod_erlang_event in modules.conf. unixodbc is installed already. > still ... the same error message. tks for your input. > > /nandy > > > On Sun, Sep 13, 2009 at 11:56 PM, Meftah Tayeb wrote: > >> hello, >> i think you enabled mod_erlang_event in the modules.conf >> install unixodbc if is not installed >> thanks >> >> Nandy Dagondon a ?crit : >> >> hi, >> >> i want to enable odbc support which is required in mod_lcr feature. >> however, i encounter ./configure problem after installing Erlang R13B01. >> this is the portion of the error messages: >> >> ....... >> checking for erl... /usr/local/bin/erl >> checking erlang version... 5.7.2 >> checking erlang libdir... >> /usr/local/lib/erlang/lib/erl_interface-3.6.2/lib >> checking erlang incdir... >> /usr/local/lib/erlang/lib/erl_interface-3.6.2/include >> checking ei.h usability... yes >> checking ei.h presence... no >> configure: WARNING: ei.h: accepted by the compiler, rejected by the >> preprocessor! >> configure: WARNING: ei.h: proceeding with the compiler's result >> checking for ei.h... yes >> checking for ei_encode_version in -lei... yes >> checking for ei_link_unlink in -lei... no >> configure: Your erlang seems OK, do not forget to enable mod_erlang_event >> in modules.conf >> configure: creating ./config.status >> config.status: creating src/include/switch_version.h.in >> .infig.status: error: cannot find input file: Makefile >> -------- END -------- >> >> i set ERL_TOP environment variable to the source directory. has anyone >> encountered this problem? can anyone give me a hint what's wrong. i'm >> compiling FS 1.0.4. >> >> thank you, >> /nandy >> >> ------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4421 (20090913) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> http://www.eset.com >> >> >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 4421 (20090913) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/90670568/attachment.html From andrew at hijacked.us Tue Sep 15 19:53:02 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Tue, 15 Sep 2009 22:53:02 -0400 Subject: [Freeswitch-users] FS 1.0.4 erl configure error In-Reply-To: <7d0bfd8c0909151904u2f3c35f0l36ab5d99f89b7845@mail.gmail.com> References: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> <4AAD1611.2030803@gmail.com> <7d0bfd8c0909132320j1b2d1113j78da26b07a37e526@mail.gmail.com> <7d0bfd8c0909151904u2f3c35f0l36ab5d99f89b7845@mail.gmail.com> Message-ID: <20090916025301.GH20978@hijacked.us> On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote: > hi folks, anyone encountered this problem? tks. I don't think this has anything to do with erlang or the freeswitch erlang module, it's simply that that module's config checks are run shortly before the real failure occurs. Andrew From nandy1925 at gmail.com Tue Sep 15 20:21:58 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 16 Sep 2009 11:21:58 +0800 Subject: [Freeswitch-users] FS 1.0.4 erl configure error In-Reply-To: <20090916025301.GH20978@hijacked.us> References: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> <4AAD1611.2030803@gmail.com> <7d0bfd8c0909132320j1b2d1113j78da26b07a37e526@mail.gmail.com> <7d0bfd8c0909151904u2f3c35f0l36ab5d99f89b7845@mail.gmail.com> <20090916025301.GH20978@hijacked.us> Message-ID: <7d0bfd8c0909152021i10a5a2dfta14fbff7fc100c89@mail.gmail.com> the ./configure script aborts after the last error message. any hint where to look for the problem? tks. /nandy On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson wrote: > On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote: > > hi folks, anyone encountered this problem? tks. > > I don't think this has anything to do with erlang or the freeswitch > erlang module, it's simply that that module's config checks are run > shortly before the real failure occurs. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/a6e99b5b/attachment.html From nandy1925 at gmail.com Tue Sep 15 20:26:06 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 16 Sep 2009 11:26:06 +0800 Subject: [Freeswitch-users] FS 1.0.4 erl configure error In-Reply-To: <7d0bfd8c0909152021i10a5a2dfta14fbff7fc100c89@mail.gmail.com> References: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> <4AAD1611.2030803@gmail.com> <7d0bfd8c0909132320j1b2d1113j78da26b07a37e526@mail.gmail.com> <7d0bfd8c0909151904u2f3c35f0l36ab5d99f89b7845@mail.gmail.com> <20090916025301.GH20978@hijacked.us> <7d0bfd8c0909152021i10a5a2dfta14fbff7fc100c89@mail.gmail.com> Message-ID: <7d0bfd8c0909152026h3f9acd27m33af33636e6362a5@mail.gmail.com> is the Erlang source needed in the FS source directory? /nandy On Wed, Sep 16, 2009 at 11:21 AM, Nandy Dagondon wrote: > the ./configure script aborts after the last error message. any hint where > to look for the problem? tks. > > /nandy > > > > On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson wrote: > >> On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote: >> > hi folks, anyone encountered this problem? tks. >> >> I don't think this has anything to do with erlang or the freeswitch >> erlang module, it's simply that that module's config checks are run >> shortly before the real failure occurs. >> >> Andrew >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/e36fe930/attachment.html From mayamatakeshi at gmail.com Tue Sep 15 20:35:23 2009 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Wed, 16 Sep 2009 12:35:23 +0900 Subject: [Freeswitch-users] No NOTIFY MWI when registering via proxy. In-Reply-To: <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> References: <15b9404e0909020359p1cb12023p7f33ed82da07bba1@mail.gmail.com> <15b9404e0909040328o457f3061ge1a1e3c9e8b49ed9@mail.gmail.com> <15b9404e0909042340g3d7db2b5x4f8aeed7b0811f6d@mail.gmail.com> <268C154B-944D-4909-B84A-CF379F275FA0@jerris.com> <15b9404e0909111903r36e1b4b0p267e3f9f0edb2ea6@mail.gmail.com> Message-ID: <15b9404e0909152035u2390478aud00c7caf72d62d6e@mail.gmail.com> On 9/12/09, mayamatakeshi wrote: > > > On Sat, Sep 12, 2009 at 1:45 AM, Michael Jerris wrote: > >> Following up, did a bug get created for this issue? >> > > Hello, > yes. > http://jira.freeswitch.org/browse/MODSOFIA-26 > Just to simplify things in case someone searches the list: Issue was solved on rev 14851. Thank you all. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/d83f4880/attachment.html From nandy1925 at gmail.com Tue Sep 15 20:47:32 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 16 Sep 2009 11:47:32 +0800 Subject: [Freeswitch-users] FS 1.0.4 erl configure error In-Reply-To: <7d0bfd8c0909152026h3f9acd27m33af33636e6362a5@mail.gmail.com> References: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> <4AAD1611.2030803@gmail.com> <7d0bfd8c0909132320j1b2d1113j78da26b07a37e526@mail.gmail.com> <7d0bfd8c0909151904u2f3c35f0l36ab5d99f89b7845@mail.gmail.com> <20090916025301.GH20978@hijacked.us> <7d0bfd8c0909152021i10a5a2dfta14fbff7fc100c89@mail.gmail.com> <7d0bfd8c0909152026h3f9acd27m33af33636e6362a5@mail.gmail.com> Message-ID: <7d0bfd8c0909152047w5d8ad521qc8df7be282557e4c@mail.gmail.com> it's working now. the problem? it's the configure script itself. some ^M characters somehow crept into the line containing ac_config_files. tks for the tip Andrew! /nandy On Wed, Sep 16, 2009 at 11:26 AM, Nandy Dagondon wrote: > is the Erlang source needed in the FS source directory? > > /nandy > > > > On Wed, Sep 16, 2009 at 11:21 AM, Nandy Dagondon wrote: > >> the ./configure script aborts after the last error message. any hint where >> to look for the problem? tks. >> >> /nandy >> >> >> >> On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson wrote: >> >>> On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote: >>> > hi folks, anyone encountered this problem? tks. >>> >>> I don't think this has anything to do with erlang or the freeswitch >>> erlang module, it's simply that that module's config checks are run >>> shortly before the real failure occurs. >>> >>> Andrew >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/3f1938e7/attachment-0001.html From mike at jerris.com Tue Sep 15 20:51:56 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 15 Sep 2009 23:51:56 -0400 Subject: [Freeswitch-users] FS 1.0.4 erl configure error In-Reply-To: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> References: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> Message-ID: <55EFB2D0-058B-4294-9697-4536FAA66D3D@jerris.com> something is messed up in your build environment, it has nothing to do with erlang. Is this with a fresh svn checkout or tarball? Mike On Sep 13, 2009, at 10:27 AM, Nandy Dagondon wrote: > hi, > > i want to enable odbc support which is required in mod_lcr feature. > however, i encounter ./configure problem after installing Erlang > R13B01. this is the portion of the error messages: > > ....... > checking for erl... /usr/local/bin/erl > checking erlang version... 5.7.2 > checking erlang libdir... /usr/local/lib/erlang/lib/ > erl_interface-3.6.2/lib > checking erlang incdir... /usr/local/lib/erlang/lib/ > erl_interface-3.6.2/include > checking ei.h usability... yes > checking ei.h presence... no > configure: WARNING: ei.h: accepted by the compiler, rejected by the > preprocessor! > configure: WARNING: ei.h: proceeding with the compiler's result > checking for ei.h... yes > checking for ei_encode_version in -lei... yes > checking for ei_link_unlink in -lei... no > configure: Your erlang seems OK, do not forget to enable > mod_erlang_event in modules.conf > configure: creating ./config.status > config.status: creating src/include/switch_version.h.in > .infig.status: error: cannot find input file: Makefile > -------- END -------- > > i set ERL_TOP environment variable to the source directory. has > anyone encountered this problem? can anyone give me a hint what's > wrong. i'm compiling FS 1.0.4. > > thank you, > /nandy > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090915/f94d4250/attachment.html From nandy1925 at gmail.com Tue Sep 15 20:56:12 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 16 Sep 2009 11:56:12 +0800 Subject: [Freeswitch-users] FS 1.0.4 erl configure error In-Reply-To: <55EFB2D0-058B-4294-9697-4536FAA66D3D@jerris.com> References: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> <55EFB2D0-058B-4294-9697-4536FAA66D3D@jerris.com> Message-ID: <7d0bfd8c0909152056s694a9b90wf2045b5574a467b4@mail.gmail.com> mike, got it from tarball. - nandy On Wed, Sep 16, 2009 at 11:51 AM, Michael Jerris wrote: > something is messed up in your build environment, it has nothing to do with > erlang. Is this with a fresh svn checkout or tarball? > Mike > > On Sep 13, 2009, at 10:27 AM, Nandy Dagondon wrote: > > hi, > > i want to enable odbc support which is required in mod_lcr feature. > however, i encounter ./configure problem after installing Erlang R13B01. > this is the portion of the error messages: > > ....... > checking for erl... /usr/local/bin/erl > checking erlang version... 5.7.2 > checking erlang libdir... /usr/local/lib/erlang/lib/erl_interface-3.6.2/lib > checking erlang incdir... > /usr/local/lib/erlang/lib/erl_interface-3.6.2/include > checking ei.h usability... yes > checking ei.h presence... no > configure: WARNING: ei.h: accepted by the compiler, rejected by the > preprocessor! > configure: WARNING: ei.h: proceeding with the compiler's result > checking for ei.h... yes > checking for ei_encode_version in -lei... yes > checking for ei_link_unlink in -lei... no > configure: Your erlang seems OK, do not forget to enable mod_erlang_event > in modules.conf > configure: creating ./config.status > config.status: creating src/include/switch_version.h.in > .infig.status: error: cannot find input file: Makefile > -------- END -------- > > i set ERL_TOP environment variable to the source directory. has anyone > encountered this problem? can anyone give me a hint what's wrong. i'm > compiling FS 1.0.4. > > thank you, > /nandy > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/740eafed/attachment.html From mkezys at gmail.com Tue Sep 15 22:29:13 2009 From: mkezys at gmail.com (Mindaugas Kezys) Date: Wed, 16 Sep 2009 08:29:13 +0300 Subject: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? In-Reply-To: <4ab00c82.161bf30a.6722.522e@mx.google.com> References: <4aa965af.161bf30a.61f8.7c7d@mx.google.com> <4AA9FBA5.5090403@kounitskiy.com> <191c3a030909110831o4e0a4844obf3c339d58f5358a@mail.gmail.com> <4ab00c82.161bf30a.6722.522e@mx.google.com> Message-ID: <032201ca368e$a1502300$e3f06900$@com> Can you tell why Freeswitch + mod_radius_cdr was not good for you? Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of email lists Sent: 2009 m. rugs?jo 16 d. 00:53 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? Thanks to those for the info and help on this issue. Ultimately ended up having to use alternative software for the radius piece (not related to any shortfalls by Freeswitch). Vlad From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, September 11, 2009 11:31 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? set the variable process_cdr=false on that a_leg first thing in your dialplan On Fri, Sep 11, 2009 at 2:26 AM, Anatoliy Kounitskiy wrote: It's normal to have to two records for a call - Start and Stop message. From what i see - you have one start and stop for each leg of the call. Regards, AK email lists wrote: > > Hello, > > > > Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate > RADIUS messages being generated for individual calls (sample messages > for one call below). Looking at the "Acct-Unique-Session-Id" and > "Acct-Session-Id" fields, it would appear that perhaps each call leg > results in a pair of start/stop RADIUS messages; is this the expected > behavior? If so, is there a way to disable RADIUS messaging for what > I presume is the "ingress" or A leg of the call? > > > > Any leads would be appreciated. > > > > Thanks in advance. > > > > Vladimir > > > > Thu Sep 10 10:37:25 2009 > > Acct-Status-Type = Start > > Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "097c8472ff7bcec7" > > Timestamp = 1252604245 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:37:25 2009 > > Acct-Status-Type = Start > > Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226 at x.x.x.x" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "53f729e173e8c0a9" > > Timestamp = 1252604245 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:37:57 2009 > > Acct-Status-Type = Stop > > Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" > > Freeswitch-Hangupcause = Normal-Unspecified > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Lastapp = "bridge" > > Freeswitch-Billusec = 32029926 > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" > > Freeswitch-Callanswerdate = "2009-09-10T10:22:00.319197-0700" > > Freeswitch-Callenddate = "2009-09-10T10:22:32.349123-0700" > > Acct-Session-Time = 32 > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "097c8472ff7bcec7" > > Timestamp = 1252604277 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:38:02 2009 > > Acct-Status-Type = Stop > > Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" > > Freeswitch-Hangupcause = Normal-Clearing > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226 at x.x.x.x" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Billusec = 32049973 > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" > > Freeswitch-Callanswerdate = "2009-09-10T10:22:00.289136-0700" > > Freeswitch-Callenddate = "2009-09-10T10:22:32.339109-0700" > > Acct-Session-Time = 32 > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "53f729e173e8c0a9" > > Timestamp = 1252604282 > > Request-Authenticator = Verified** > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/0e937e56/attachment-0001.html From mike at jerris.com Wed Sep 16 00:54:06 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 16 Sep 2009 03:54:06 -0400 Subject: [Freeswitch-users] FS 1.0.4 erl configure error In-Reply-To: <7d0bfd8c0909152047w5d8ad521qc8df7be282557e4c@mail.gmail.com> References: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> <4AAD1611.2030803@gmail.com> <7d0bfd8c0909132320j1b2d1113j78da26b07a37e526@mail.gmail.com> <7d0bfd8c0909151904u2f3c35f0l36ab5d99f89b7845@mail.gmail.com> <20090916025301.GH20978@hijacked.us> <7d0bfd8c0909152021i10a5a2dfta14fbff7fc100c89@mail.gmail.com> <7d0bfd8c0909152026h3f9acd27m33af33636e6362a5@mail.gmail.com> <7d0bfd8c0909152047w5d8ad521qc8df7be282557e4c@mail.gmail.com> Message-ID: <015CD67A-8F90-4004-8C0E-40BA6C072B5C@jerris.com> Are those in the Tarball? On Sep 15, 2009, at 11:47 PM, Nandy Dagondon wrote: > it's working now. the problem? it's the configure script itself. > some ^M characters somehow crept into the line containing > ac_config_files. tks for the tip Andrew! > > /nandy > > On Wed, Sep 16, 2009 at 11:26 AM, Nandy Dagondon > wrote: > is the Erlang source needed in the FS source directory? > > /nandy > > > > On Wed, Sep 16, 2009 at 11:21 AM, Nandy Dagondon > wrote: > the ./configure script aborts after the last error message. any hint > where to look for the problem? tks. > > /nandy > > > > On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson > wrote: > On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote: > > hi folks, anyone encountered this problem? tks. > > I don't think this has anything to do with erlang or the freeswitch > erlang module, it's simply that that module's config checks are run > shortly before the real failure occurs. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/78e82c3f/attachment.html From ahmedmunir007 at gmail.com Wed Sep 16 01:23:10 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Wed, 16 Sep 2009 14:23:10 +0600 Subject: [Freeswitch-users] How to Process Invalid extension in FS Message-ID: Hi, I'm newbie in FS. I want to know how to process invalid extension in FS? Because I want to prompt the IVR if invalid extension is dialled. Kindly advice me. -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/15a2cecc/attachment.html From jason at jasonjgw.net Wed Sep 16 01:36:38 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 16 Sep 2009 18:36:38 +1000 Subject: [Freeswitch-users] How to Process Invalid extension in FS In-Reply-To: References: Message-ID: <20090916083638.GA26867@jdc.jasonjgw.net> Ahmed Munir wrote: > I'm newbie in FS. I want to know how to process invalid extension in FS? > Because I want to prompt the IVR if invalid extension is dialled. You could write an entry at the end of the dial-plan that matches any extension and invokes the IVR. From nandy1925 at gmail.com Wed Sep 16 01:48:05 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 16 Sep 2009 16:48:05 +0800 Subject: [Freeswitch-users] FS 1.0.4 erl configure error In-Reply-To: <015CD67A-8F90-4004-8C0E-40BA6C072B5C@jerris.com> References: <7d0bfd8c0909130727uba86ea5u77deaf84d593e102@mail.gmail.com> <4AAD1611.2030803@gmail.com> <7d0bfd8c0909132320j1b2d1113j78da26b07a37e526@mail.gmail.com> <7d0bfd8c0909151904u2f3c35f0l36ab5d99f89b7845@mail.gmail.com> <20090916025301.GH20978@hijacked.us> <7d0bfd8c0909152021i10a5a2dfta14fbff7fc100c89@mail.gmail.com> <7d0bfd8c0909152026h3f9acd27m33af33636e6362a5@mail.gmail.com> <7d0bfd8c0909152047w5d8ad521qc8df7be282557e4c@mail.gmail.com> <015CD67A-8F90-4004-8C0E-40BA6C072B5C@jerris.com> Message-ID: <7d0bfd8c0909160148y59b04cdu3af3618674150bdf@mail.gmail.com> hi mike, i download the tarball file to check the configure script. it's clean. so, there must be an error during my first download or build. - nandy On Wed, Sep 16, 2009 at 3:54 PM, Michael Jerris wrote: > Are those in the Tarball? > > > On Sep 15, 2009, at 11:47 PM, Nandy Dagondon wrote: > > it's working now. the problem? it's the configure script itself. some ^M > characters somehow crept into the line containing ac_config_files. tks for > the tip Andrew! > > /nandy > > On Wed, Sep 16, 2009 at 11:26 AM, Nandy Dagondon < > nandy1925 at gmail.com> wrote: > >> is the Erlang source needed in the FS source directory? >> >> /nandy >> >> >> >> On Wed, Sep 16, 2009 at 11:21 AM, Nandy Dagondon < >> nandy1925 at gmail.com> wrote: >> >>> the ./configure script aborts after the last error message. any hint >>> where to look for the problem? tks. >>> >>> /nandy >>> >>> >>> >>> On Wed, Sep 16, 2009 at 10:53 AM, Andrew Thompson < >>> andrew at hijacked.us> wrote: >>> >>>> On Wed, Sep 16, 2009 at 10:04:14AM +0800, Nandy Dagondon wrote: >>>> > hi folks, anyone encountered this problem? tks. >>>> >>>> I don't think this has anything to do with erlang or the freeswitch >>>> erlang module, it's simply that that module's config checks are run >>>> shortly before the real failure occurs. >>>> >>>> Andrew >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> >>>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/92fd95a3/attachment.html From t.mahe at telemaque.fr Wed Sep 16 01:53:28 2009 From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=) Date: Wed, 16 Sep 2009 10:53:28 +0200 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> Message-ID: <4AB0A788.1000105@telemaque.fr> Hi, Count on me for answering questions on IRC when I'm in, and for subprojects I'm in too as you know ;) Regards, Gled Diego Viola a ?crit : > Hi Michael, > > You can count with me for anything else, like documentation, > coding/scripting, or any other FreeSWITCH related stuff. > > Regards, > > Diego > > 2009/9/14 Jo?o Mesquita > > > You can assign two things to me. > > 1. libesl code documentation (partially done and Doxygened - needs > cleaning) > 2. Bug marshal. I am setting up the proper lab environment here to > be able to test most stuff. > > Count me in for any questions I can answer and I am _always_ on IRC > > jmesquita > > On Mon, Sep 14, 2009 at 1:33 PM, Michael Collins > > wrote: > > Hello FreeSWITCHers! > > We are looking for people who are in a position to help out > with various subprojects that will help FreeSWITCH to keep > growing. We need people to help out in these basic areas: > > Bug marshals (people who watch JIRA and test bug reports, > patches, etc.) > Documentation maintainers (people who update the wiki when new > stuff comes out, also those familiar with mediawiki > administration) > Documentation authors (people who write new docs, how-to's, > tutorials, examples, etc.) > Package maintainers (people who manage Debian debs, RPMs, etc.) > > Additionally, we are always looking for more folks to assist > with answering questions on IRC and the mailing list. It is > definitely nice to have people who've gone through the pains > of switching to FreeSWITCH (or learning it from scratch) who > can assist the steady stream of new users. > > If you want to help and aren't sure where to go from here then > please at least do the following: > #1 - Join #freeswitch on irc.freenode.net > and hang out as much as possible > #2 - Check the recent changes link on wiki.freeswitch.org > each day > #3 - Join the Friday public conference call and listen in > These three things, in addition to the mailing list, will keep > you well in tune with the FreeSWITCH community and what's > happening. > > Next, make a note of the parts of FS that you use frequently, > know a lot about, or are particularly passionate about. Those > are the items we'd love to have you help us with. For example: > if you use mod_xml_curl frequently and have been through the > set up process then you're a prime candidate to help answer > questions, refine the mod_xml_curl wiki documentation, write > up a tutorial, contribute a working example of a web server & > database schema, etc. If you are good with a scripting > language then we could definitely use help with rounding out > the docs for your favorite language. We could also use code > samples, so ask for a contrib folder if you have things you > would like to share. Or how about this: you read something on > the wiki, it doesn't quite work when you try, so you tinker > until you figure it out. Now you're in a position to update > the wiki for everyone else's benefit, too. > > As you can see, you don't have to be a FreeSWITCH expert > before you can help the project. What we really need are > people who care about the project and want to see it flourish. > If you are such a person then please contact me off list. Tell > me what you're good at or where you would like to help. > > Many thanks for all of your support! > -Michael > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/6d6f14d0/attachment-0001.html From lists at venturevoip.com Wed Sep 16 03:40:15 2009 From: lists at venturevoip.com (Matt Riddell) Date: Wed, 16 Sep 2009 22:40:15 +1200 Subject: [Freeswitch-users] Limit_Hash Message-ID: <4AB0C08F.4040109@venturevoip.com> Hi, Didn't see this one come through before when I posted it, so sending it again - apologies if it did come through. I've moved this discussion to users as it seems my query is moving in that direction :) So, upon looking at limit_hash, it appears to do what I need to do. My question then becomes, how do I set a hash for an originated call? It seems that limit_hash is an application rather than a channel variable, and so far I've been doing most things without touching the dialplan. So, say I want to originate 9 calls, 3 from 3 customers. I would like to mark the calls with my_customer_group_1 through 3, and then use the limit_hash_usage command to verify the count of channels in each group. I therefore have a few questions: 1. Can I mark a call in the originate statement? 2. How do I use the limit_hash_usage command? The wiki states: You can verify the usage of any resource with the limit_hash_usage api call. limit_hash_usage Is realm the same as a SIP realm? Is id the hash that I have used to mark the call with? Just making sure :) -- Cheers, Matt Riddell Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) From tculjaga at gmail.com Wed Sep 16 06:49:43 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 16 Sep 2009 15:49:43 +0200 Subject: [Freeswitch-users] reloadxml question Message-ID: <65d96fc80909160649kd5ca200gdeed86239a0257d4@mail.gmail.com> hi, I've build a custom module for FS and everytihng work well except reloadxml command :P... m'I missing something in my module? ... i used mod_skeleton as a template when i started. When i start the FS without my module reloadxml works fine ... as soon as i include my module within modules.conf.xml and start FS .. it hangs. So, it is definitelly up to the custom module ... but what can it be? freeswitch at l01freeswitch1> freeswitch at l01freeswitch1> freeswitch at l01freeswitch1> reloadxml nothing happens ... i have to kill freeswitch (kill -9) to get the shell. T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/328bf316/attachment.html From tzury.by at reguluslabs.com Wed Sep 16 06:58:04 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Wed, 16 Sep 2009 16:58:04 +0300 Subject: [Freeswitch-users] How can I configure TO and FROM in the invite message Message-ID: <10128ef10909160658h2198c114o81c3c59f9f37c993@mail.gmail.com> Hi, Currently, the invite message looks as follows INVITE sip:1002 at CLIENT_IP:5060 SIP/2.0 Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se Max-Forwards: 69 From: "Extension 1001" ;tag=2rH67Q3aa1rpe To: Is there a way to configure FS so the message will look like this: INVITE sip:1002 at CLIENT_IP:5060 SIP/2.0 Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se Max-Forwards: 69 From: "Extension 1001" ;tag=2rH67Q3aa1rpe To: That is, at "From" having the account's domain name (e.g. sip:1001 at example.com) instead of the server's IP address. and having the same at "To" thanks @tzury From frank at carmickle.com Wed Sep 16 01:01:28 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 16 Sep 2009 04:01:28 -0400 Subject: [Freeswitch-users] shared mailbox, mwi Message-ID: <20090916080128.GF30343@base.carmickle.com> Hello I am trying to set up a shared mailbox per group of extensions where the mailboxes are [2-9][2-9][0,2,4,6,8]0 and the phones are [2-9][2-9]\d[1-9]. I mostly have it working except I can't seem to figure out how to get mwi for the phones in the group. I've tried a number of things. I haven't figured out how you can have a mailbox that doesn't have a user associated with it. When I use I get the mwi but I am not able to call that user any more. I have set for each user. What are my options for making this scenario work? Thank you --Frank From pankajanand18 at gmail.com Wed Sep 16 02:29:39 2009 From: pankajanand18 at gmail.com (pankaj anand) Date: Wed, 16 Sep 2009 14:59:39 +0530 Subject: [Freeswitch-users] how to add new user for external profile Message-ID: <809ad7ab0909160229u208733f2r776bc61b20bdc18a@mail.gmail.com> hi , i m very new to the FreeSwitch.. can any one tell me how to add a new user. i have already tried creating a new user by creating a $INSTALL_DIR/conf/directory/default/pankaj.xml : but when i try to connect it using , the softphone shows forbidden. Can anyone tell me where i am making a mistake. with regards Pankaj anand -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/9f1bb3b7/attachment.html From rupa at rupa.com Wed Sep 16 07:02:22 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 16 Sep 2009 09:02:22 -0500 Subject: [Freeswitch-users] Limit_Hash In-Reply-To: <4AB0C08F.4040109@venturevoip.com> References: <4AB0C08F.4040109@venturevoip.com> Message-ID: On Wed, Sep 16, 2009 at 5:40 AM, Matt Riddell wrote: > My question then becomes, how do I set a hash for an originated call? > > It seems that limit_hash is an application rather than a channel > variable, and so far I've been doing most things without touching the > dialplan. Yes, it is a dialplan app. So, you have to call it from a dialplan. > So, say I want to originate 9 calls, 3 from 3 customers. > > I would like to mark the calls with my_customer_group_1 through 3, and > then use the limit_hash_usage command to verify the count of channels in > each group. > > I therefore have a few questions: > > 1. Can I mark a call in the originate statement? If you don't want to bounce through the XML Dialplan, you can use an inline dialplan instead and specify the dialplan on the originate commandline. The wiki has an example: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_InlineDialplan#Originate > 2. How do I use the limit_hash_usage command? > > The wiki states: > > You can verify the usage of any resource with the limit_hash_usage api call. > > limit_hash_usage > > Is realm the same as a SIP realm? > > Is id the hash that I have used to mark the call with? > > Just making sure :) realm and id are just arbitrary strings. They can be useful if you want to do reporting out of the database (if using regular limit api, for limit_usage it wouldn't apply). You can provide your own meaning to realm and id. In a multi-tenant setup, you might use domain as the realm but that isn't necessary. > -- > Cheers, > > Matt Riddell > Director -- -Rupa From brian at freeswitch.org Wed Sep 16 07:05:00 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Sep 2009 09:05:00 -0500 Subject: [Freeswitch-users] reloadxml question In-Reply-To: <65d96fc80909160649kd5ca200gdeed86239a0257d4@mail.gmail.com> References: <65d96fc80909160649kd5ca200gdeed86239a0257d4@mail.gmail.com> Message-ID: <1FAB38C1-83CA-427E-ACB7-1E5CD55E9710@freeswitch.org> Yes you're missing a switch_xml_free(xml); some place. /b On Sep 16, 2009, at 8:49 AM, Tihomir Culjaga wrote: > hi, > > I've build a custom module for FS and everytihng work well except > reloadxml command :P... m'I missing something in my module? ... i > used mod_skeleton as a template when i started. > > > When i start the FS without my module reloadxml works fine ... as > soon as i include my module within modules.conf.xml and start FS .. > it hangs. > So, it is definitelly up to the custom module ... but what can it be? > > > > freeswitch at l01freeswitch1> > freeswitch at l01freeswitch1> > freeswitch at l01freeswitch1> reloadxml > > nothing happens ... i have to kill freeswitch (kill -9) to get the > shell. > > T. > From tculjaga at gmail.com Wed Sep 16 07:17:35 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 16 Sep 2009 16:17:35 +0200 Subject: [Freeswitch-users] how to add new user for external profile In-Reply-To: <809ad7ab0909160229u208733f2r776bc61b20bdc18a@mail.gmail.com> References: <809ad7ab0909160229u208733f2r776bc61b20bdc18a@mail.gmail.com> Message-ID: <65d96fc80909160717o49c677a3wb2945935b851206d@mail.gmail.com> FS loads all users from $INSTALL_DIR/conf/directory/ and you did it correct. freeswitch.xml:
Than, you need to check sip profiles. By default FS will accept registrations on internal profiles only... so you should enable it on the external as well. look at this portion of your adequate sip profile: Just make sure you use correct IP_ADDRESS:PORT to match the correct profile vars.xml: T. On Wed, Sep 16, 2009 at 11:29 AM, pankaj anand wrote: > hi , i m very new to the FreeSwitch.. > can any one tell me how to add a new user. > i have already tried creating a new user by creating a > $INSTALL_DIR/conf/directory/default/pankaj.xml : > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > but when i try to connect it using , the softphone shows forbidden. > Can anyone tell me where i am making a mistake. > > with regards > Pankaj anand > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/80a1fa79/attachment-0001.html From tculjaga at gmail.com Wed Sep 16 07:17:55 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 16 Sep 2009 16:17:55 +0200 Subject: [Freeswitch-users] reloadxml question In-Reply-To: <1FAB38C1-83CA-427E-ACB7-1E5CD55E9710@freeswitch.org> References: <65d96fc80909160649kd5ca200gdeed86239a0257d4@mail.gmail.com> <1FAB38C1-83CA-427E-ACB7-1E5CD55E9710@freeswitch.org> Message-ID: <65d96fc80909160717x40e8db16o302246b86ba1c@mail.gmail.com> perfect, thanks. T. On Wed, Sep 16, 2009 at 4:05 PM, Brian West wrote: > Yes you're missing a switch_xml_free(xml); some place. > > /b > > On Sep 16, 2009, at 8:49 AM, Tihomir Culjaga wrote: > > > hi, > > > > I've build a custom module for FS and everytihng work well except > > reloadxml command :P... m'I missing something in my module? ... i > > used mod_skeleton as a template when i started. > > > > > > When i start the FS without my module reloadxml works fine ... as > > soon as i include my module within modules.conf.xml and start FS .. > > it hangs. > > So, it is definitelly up to the custom module ... but what can it be? > > > > > > > > freeswitch at l01freeswitch1> > > freeswitch at l01freeswitch1> > > freeswitch at l01freeswitch1> reloadxml > > > > nothing happens ... i have to kill freeswitch (kill -9) to get the > > shell. > > > > T. > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/1e975ca6/attachment.html From aep.lists at it46.se Wed Sep 16 07:26:23 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Wed, 16 Sep 2009 16:26:23 +0200 Subject: [Freeswitch-users] originate command sofia behaviour Message-ID: <428d9a28388f5b40cf85a37bc100fc08.squirrel@correo.nodo50.org> I will like to update the wiki to spell out clearly the differences between this three commands I have a IVR running in 4600 and the FS box has IP address 192.168.46.15 originate sofia/192.168.46.15/1001 4600 originate sofia/internal/1001 at 192.168.46.15 4600 originate sofia/internal/1001%192.168.46.15 4600 The first originate places a call as a external gateway, not until registered phone 1001 answers the call is transfer to 4600 The second and third originate command triggers extension 4600 Javascript IVR although 1001 has not answer Can anyone clarify me if this is the intended behavior also including the difference between % and @ /aep From christian.loeschenkohl at xpirio.com Wed Sep 16 07:29:50 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Wed, 16 Sep 2009 16:29:50 +0200 Subject: [Freeswitch-users] memory leak - outbound socket Message-ID: <4AB0F65E.4060600@xpirio.com> hello version : 1.0.4 std. tarball - the wiki example for php outbound socket connection leaks memory without the async option - the memory used is never given back - async isn't that usefull for us - we want to query databases, set variables and so on no wait statements are possible <<<<---- no async !!!! the script is on the site http://wiki.freeswitch.org/wiki/PHP_ESL ------------------------------- what can i do? on our production server we use outbound socket connection and the 4 gig of memory are eaten up in less than a day br -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com From msc at freeswitch.org Wed Sep 16 07:56:59 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Sep 2009 07:56:59 -0700 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <4AB0A788.1000105@telemaque.fr> References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <4AB0A788.1000105@telemaque.fr> Message-ID: <87f2f3b90909160756i7af013dbie657ca02cd0dec6a@mail.gmail.com> On Wed, Sep 16, 2009 at 1:53 AM, Tristan Mah? wrote: > Hi, > > Count on me for answering questions on IRC when I'm in, and for subprojects > I'm in too as you know ;) > Merci! Okay, what's your IRC nick and when are you generally on line? Also, I'm pretty sure that you're fluent in French which is good because we need more multilingual people out there. Last question: what are your areas of expertise? I'd like to keep a list of people and what they're good at so we know whom to ask first when questions come up. Thanks again! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/2a5a7dd6/attachment.html From jmesquita at freeswitch.org Wed Sep 16 08:00:13 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 16 Sep 2009 12:00:13 -0300 Subject: [Freeswitch-users] How can I configure TO and FROM in the invite message In-Reply-To: <10128ef10909160658h2198c114o81c3c59f9f37c993@mail.gmail.com> References: <10128ef10909160658h2198c114o81c3c59f9f37c993@mail.gmail.com> Message-ID: Is this what you are looking for? http://wiki.freeswitch.org/wiki/Channel_Variables#sip_invite_domain jmesquita On Wed, Sep 16, 2009 at 10:58 AM, Tzury Bar Yochay wrote: > Hi, > Currently, the invite message looks as follows > > INVITE sip:1002 at CLIENT_IP:5060 SIP/2.0 > Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se > Max-Forwards: 69 > From: "Extension 1001" ;tag=2rH67Q3aa1rpe > To: > > Is there a way to configure FS so the message will look like this: > > INVITE sip:1002 at CLIENT_IP:5060 SIP/2.0 > Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se > Max-Forwards: 69 > From: "Extension 1001" ;tag=2rH67Q3aa1rpe > To: > > That is, at "From" having the account's domain name (e.g. > sip:1001 at example.com ) instead of the server's IP > address. > and having the same at "To" > > thanks > @tzury > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/f2c5c286/attachment.html From brian at freeswitch.org Wed Sep 16 08:00:39 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Sep 2009 10:00:39 -0500 Subject: [Freeswitch-users] reloadxml question In-Reply-To: <65d96fc80909160717x40e8db16o302246b86ba1c@mail.gmail.com> References: <65d96fc80909160649kd5ca200gdeed86239a0257d4@mail.gmail.com> <1FAB38C1-83CA-427E-ACB7-1E5CD55E9710@freeswitch.org> <65d96fc80909160717x40e8db16o302246b86ba1c@mail.gmail.com> Message-ID: Might I ask what you are working on? Its interesting to hear what people are doing with FreeSWITCH. /b On Sep 16, 2009, at 9:17 AM, Tihomir Culjaga wrote: > perfect, > > thanks. > > T. From rupa at rupa.com Wed Sep 16 08:05:30 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 16 Sep 2009 10:05:30 -0500 Subject: [Freeswitch-users] memory leak - outbound socket In-Reply-To: <4AB0F65E.4060600@xpirio.com> References: <4AB0F65E.4060600@xpirio.com> Message-ID: Either: 1) Provide a simple self-contained example that demonstrates the leak or 2) Run your application with FreeSWITCH under valgrind and provide the final output. To run freeswitch under valgrind: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Collection_Information_With_Valgrind_.28Linux.2FUnix.29 You should not have to run with high load to capture the behavior. Try with just 5 (in series) and then stop freeswitch. 2009/9/16 Christian L?schenkohl : > hello > > version : 1.0.4 std. tarball > > - the wiki example for php outbound socket connection leaks memory without the async option > - the memory used is never given back > - async isn't that usefull for us - we want to query databases, set variables and so on > ? no wait statements are possible > > > > > <<<<---- no async !!!! > > > > the script is on the site > http://wiki.freeswitch.org/wiki/PHP_ESL > > ------------------------------- > > what can i do? > on our production server we use outbound socket connection and the 4 gig of memory are > eaten up in less than a day > > br > > -- > Ing. Christian L?schenkohl > 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.loeschenkohl at xpirio.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From msc at freeswitch.org Wed Sep 16 08:16:15 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Sep 2009 08:16:15 -0700 Subject: [Freeswitch-users] A FreeSWITCH Calling Card Application written in Ruby In-Reply-To: References: <86a32abc0909061836j4a4d756bnf5f1e675e1607cd6@mail.gmail.com> <86a32abc0909061857u3bff1a3bkb16400f03f9b980c@mail.gmail.com> Message-ID: <87f2f3b90909160816s1e72508byb473df83db53da15@mail.gmail.com> On Tue, Sep 15, 2009 at 1:46 PM, roberto wrote: > Hello, > > Someone could tell me what happens to the project, it seems that is no > longer available in github ? > > http://github.com/diego/freeswitch-card/ > > thanks, > I haven't seen Diego Viola on line for a day or two. He can answer this when he's back online. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/76461210/attachment.html From brian at freeswitch.org Wed Sep 16 08:18:02 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Sep 2009 10:18:02 -0500 Subject: [Freeswitch-users] How can I configure TO and FROM in the invite message In-Reply-To: References: <10128ef10909160658h2198c114o81c3c59f9f37c993@mail.gmail.com> Message-ID: Or you setup a gateway and set the from-domain /b On Sep 16, 2009, at 10:00 AM, Jo?o Mesquita wrote: > Is this what you are looking for? > > http://wiki.freeswitch.org/wiki/Channel_Variables#sip_invite_domain > > jmesquita > > On Wed, Sep 16, 2009 at 10:58 AM, Tzury Bar Yochay > wrote: > Hi, > Currently, the invite message looks as follows > > INVITE sip:1002 at CLIENT_IP:5060 SIP/2.0 > Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se > Max-Forwards: 69 > From: "Extension 1001" ;tag=2rH67Q3aa1rpe > To: > > Is there a way to configure FS so the message will look like this: > > INVITE sip:1002 at CLIENT_IP:5060 SIP/2.0 > Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se > Max-Forwards: 69 > From: "Extension 1001" ;tag=2rH67Q3aa1rpe > To: > > That is, at "From" having the account's domain name (e.g. > sip:1001 at example.com) instead of the server's IP address. > and having the same at "To" > > thanks > @tzury > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/3a07c37f/attachment-0001.html From christian.loeschenkohl at xpirio.com Wed Sep 16 08:39:34 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Wed, 16 Sep 2009 17:39:34 +0200 Subject: [Freeswitch-users] memory leak - outbound socket In-Reply-To: References: <4AB0F65E.4060600@xpirio.com> Message-ID: <4AB106B6.7090305@xpirio.com> as a good fs user - of course i am :-) - i made a jira on this MODAPP-336 to be precise i hope this helps to solve my problem br On 2009-09-16 17:05, Rupa Schomaker wrote: > Either: > > 1) Provide a simple self-contained example that demonstrates the leak > > or > > 2) Run your application with FreeSWITCH under valgrind and provide the > final output. To run freeswitch under valgrind: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Collection_Information_With_Valgrind_.28Linux.2FUnix.29 > > You should not have to run with high load to capture the behavior. > Try with just 5 (in series) and then stop freeswitch. > > > 2009/9/16 Christian L?schenkohl: >> hello >> >> version : 1.0.4 std. tarball >> >> - the wiki example for php outbound socket connection leaks memory without the async option >> - the memory used is never given back >> - async isn't that usefull for us - we want to query databases, set variables and so on >> no wait statements are possible >> >> >> >> >> <<<<---- no async !!!! >> >> >> >> the script is on the site >> http://wiki.freeswitch.org/wiki/PHP_ESL >> >> ------------------------------- >> >> what can i do? >> on our production server we use outbound socket connection and the 4 gig of memory are >> eaten up in less than a day >> >> br >> >> -- >> Ing. Christian L?schenkohl >> 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.loeschenkohl at xpirio.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com From msc at freeswitch.org Wed Sep 16 08:48:23 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Sep 2009 08:48:23 -0700 Subject: [Freeswitch-users] ATTN: Debian gurus and enthusiasts Message-ID: <87f2f3b90909160848y26bdc08fk49ec36df6757dc1a@mail.gmail.com> If you are a Debian person and have experience creating .debs then read on... Frank Carmickle has graciously volunteered to assist with creating a FS deb config. I've heard that others have been doing something similar or are interested. If you are such a person then please email Frank and me off list so that we can organize the volunteers. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/7a277b91/attachment.html From jerry.richards at teotech.com Wed Sep 16 08:50:29 2009 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 16 Sep 2009 08:50:29 -0700 Subject: [Freeswitch-users] FS Presence Implementation In-Reply-To: <191c3a030909151153i485e5f7v876db4cb6e95600a@mail.gmail.com> References: <65AA86A7106D4255A5734520F55D3C50@greyhawk.tonecommander.com> <191c3a030909151153i485e5f7v876db4cb6e95600a@mail.gmail.com> Message-ID: I think you're referring to the SIP SIMPLE implementation as the default FS presence mechanism. This is fine and I can use that protocol. The question I still have regards the plain text content in the body of the SIP MESSAGE method. What is the format of this plain text for presence that is compatible with the FS implementation? Best Regards, Jerry _____ From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Tuesday, September 15, 2009 11:53 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FS Presence Implementation the default config ships with presence enabled for SIP if you have a phone that supports it, all you have to do is enable it on the phone. On Tue, Sep 15, 2009 at 1:30 PM, Jerry Richards wrote: Also, is presence conveyed as any string? Or is presence a predefined list of status? Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards at teotech.com] Sent: Tuesday, September 15, 2009 8:46 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: FS Presence Implementation I would like to modify my SIP phone and my gateway to convey/exchange presence information. Could someone point me toward the FS presence documentation? I've seen bits and pieces. Also, I think presence can be communicated via more than one protocol. Thanks And Best Regards, Jerry _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/413f145f/attachment.html From msc at freeswitch.org Wed Sep 16 08:52:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Sep 2009 08:52:03 -0700 Subject: [Freeswitch-users] How to Process Invalid extension in FS In-Reply-To: References: Message-ID: <87f2f3b90909160852n5951e720wf974e6ec3ef2a553@mail.gmail.com> On Wed, Sep 16, 2009 at 1:23 AM, Ahmed Munir wrote: > Hi, > > I'm newbie in FS. I want to know how to process invalid extension in FS? > Because I want to prompt the IVR if invalid extension is dialled. > > > Kindly advice me. > > Is this an invalid extension that was dialed by a registered SIP phone or by a caller who has already connected to an IVR? Just curious what your application is. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/8467c637/attachment.html From TStutsman at BossProductsInc.com Wed Sep 16 09:08:00 2009 From: TStutsman at BossProductsInc.com (Travis Stutsman) Date: Wed, 16 Sep 2009 12:08:00 -0400 Subject: [Freeswitch-users] faxrx error 13 Unexpected message received Message-ID: <5991020B1C443E459410B16D7B08F33E030879@dc1.bossproductsinc.com> In my attempts to receive a fax from a PSTN fax machine, the transaction fails with error code 13 "Unexpected message received". Verbose logging is on for mod_fax. Here is an exerpt: ################################################# 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: CFR with final frame tag 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: ff 13 84 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 In state 12 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected CFR received in state 12 ...... ======================================================================== ====== 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:180 Fax processing not successful - result (13) Unexpected message received. 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:185 Remote station id: ********** 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:186 Local station id: SpanDSP Fax Ident 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:187 Pages transferred: 0 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:189 Total fax pages: 0 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:190 Image resolution: 8031x3850 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:191 Transfer Rate: 14400 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:193 ECM status on 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:194 remote country: 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:195 remote vendor: 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:196 remote model: 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:198 ======================================================================== ====== ################################################# I have successfully received a fax from faxzero.com on this installation, as have I successfully sent a fax from freeswitch to the PSTN fax machine in question. I've been digging, but there doesn't seem to be a whole lot of information on faxing. I guess I could use a bit of direction. Any input is much appreciated. Thanks! -- Travis From steveu at coppice.org Wed Sep 16 09:50:58 2009 From: steveu at coppice.org (Steve Underwood) Date: Thu, 17 Sep 2009 00:50:58 +0800 Subject: [Freeswitch-users] faxrx error 13 Unexpected message received In-Reply-To: <5991020B1C443E459410B16D7B08F33E030879@dc1.bossproductsinc.com> References: <5991020B1C443E459410B16D7B08F33E030879@dc1.bossproductsinc.com> Message-ID: <4AB11772.4020904@coppice.org> On 09/17/2009 12:08 AM, Travis Stutsman wrote: > In my attempts to receive a fax from a PSTN fax machine, the transaction > fails with error code 13 "Unexpected message received". Verbose logging > is on for mod_fax. Here is an exerpt: > > ################################################# > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: CFR with > final frame tag > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: ff 13 84 > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 In state 12 > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected > CFR received in state 12 > ...... > ======================================================================== > ====== > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:180 Fax processing not > successful - result (13) Unexpected message received. > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:185 Remote station id: > ********** > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:186 Local station id: > SpanDSP Fax Ident > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:187 Pages transferred: 0 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:189 Total fax pages: 0 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:190 Image resolution: > 8031x3850 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:191 Transfer Rate: > 14400 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:193 ECM status on > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:194 remote country: > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:195 remote vendor: > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:196 remote model: > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:198 > ======================================================================== > ====== > ################################################# > > I have successfully received a fax from faxzero.com on this > installation, as have I successfully sent a fax from freeswitch to the > PSTN fax machine in question. > > I've been digging, but there doesn't seem to be a whole lot of > information on faxing. I guess I could use a bit of direction. Any > input is much appreciated. > You seem to have chopped out all the interesting parts of that log. A full log might say something interesting. Steve From rob4manhere at gmail.com Wed Sep 16 09:58:27 2009 From: rob4manhere at gmail.com (Rob Forman) Date: Wed, 16 Sep 2009 11:58:27 -0500 Subject: [Freeswitch-users] faxrx error 13 Unexpected message received In-Reply-To: <4AB11772.4020904@coppice.org> References: <5991020B1C443E459410B16D7B08F33E030879@dc1.bossproductsinc.com> <4AB11772.4020904@coppice.org> Message-ID: <6421BEAE-8A70-43FB-8A09-91299AB046DA@gmail.com> And make sure verbose is set to true in ./conf/autoload_configs/ fax.conf.xml. On Sep 16, 2009, at 11:50 AM, Steve Underwood wrote: > On 09/17/2009 12:08 AM, Travis Stutsman wrote: >> In my attempts to receive a fax from a PSTN fax machine, the >> transaction >> fails with error code 13 "Unexpected message received". Verbose >> logging >> is on for mod_fax. Here is an exerpt: >> >> ################################################# >> 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: CFR >> with >> final frame tag >> 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: ff >> 13 84 >> 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 In state >> 12 >> 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected >> CFR received in state 12 >> ...... >> = >> = >> = >> ===================================================================== >> ====== >> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:180 Fax processing not >> successful - result (13) Unexpected message received. >> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:185 Remote station id: >> ********** >> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:186 Local station id: >> SpanDSP Fax Ident >> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:187 Pages transferred: 0 >> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:189 Total fax pages: 0 >> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:190 Image resolution: >> 8031x3850 >> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:191 Transfer Rate: >> 14400 >> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:193 ECM status >> on >> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:194 remote country: >> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:195 remote vendor: >> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:196 remote model: >> 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:198 >> = >> = >> = >> ===================================================================== >> ====== >> ################################################# >> >> I have successfully received a fax from faxzero.com on this >> installation, as have I successfully sent a fax from freeswitch to >> the >> PSTN fax machine in question. >> >> I've been digging, but there doesn't seem to be a whole lot of >> information on faxing. I guess I could use a bit of direction. Any >> input is much appreciated. >> > You seem to have chopped out all the interesting parts of that log. A > full log might say something interesting. > > Steve > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From frank at impactfax.com Wed Sep 16 10:00:35 2009 From: frank at impactfax.com (Frank @ Impact) Date: Wed, 16 Sep 2009 13:00:35 -0400 Subject: [Freeswitch-users] session record does not for very short calls Message-ID: FreeSWITCH Version 1.0.trunk (12790M) I have this in my DP works fine as long as the call is long enough. But if the call is only, say, 3-4 seconds long (or something very short like that), then the wav file is never created with the audio in it. Is there a work around for this? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/9ccf5d5e/attachment.html From msc at freeswitch.org Wed Sep 16 10:10:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Sep 2009 10:10:42 -0700 Subject: [Freeswitch-users] ATTN: Debian gurus and enthusiasts In-Reply-To: <87f2f3b90909160848y26bdc08fk49ec36df6757dc1a@mail.gmail.com> References: <87f2f3b90909160848y26bdc08fk49ec36df6757dc1a@mail.gmail.com> Message-ID: <87f2f3b90909161010r7a3d170dqe999ab61b84d78c0@mail.gmail.com> FYI, MikeJ reminded me that we have .deb in tree already, so if you want to help maintain that then hop on IRC and hook up w/ MikeJ for more info. -MC On Wed, Sep 16, 2009 at 8:48 AM, Michael Collins wrote: > If you are a Debian person and have experience creating .debs then read > on... > > Frank Carmickle has graciously volunteered to assist with creating a FS deb > config. I've heard that others have been doing something similar or are > interested. If you are such a person then please email Frank and me off list > so that we can organize the volunteers. > > Thanks! > -Michael > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/1f98964e/attachment.html From TStutsman at BossProductsInc.com Wed Sep 16 10:21:02 2009 From: TStutsman at BossProductsInc.com (Travis Stutsman) Date: Wed, 16 Sep 2009 13:21:02 -0400 Subject: [Freeswitch-users] faxrx error 13 Unexpected message received In-Reply-To: <4AB11772.4020904@coppice.org> References: <5991020B1C443E459410B16D7B08F33E030879@dc1.bossproductsinc.com> <4AB11772.4020904@coppice.org> Message-ID: <5991020B1C443E459410B16D7B08F33E03087B@dc1.bossproductsinc.com> Alrighty. Here is mod_fax from beginning to end. ################################################# 2009-09-15 10:41:26.433382 [DEBUG] mod_fax.c:591 Raw read codec activation Success L16 20000 2009-09-15 10:41:26.433382 [DEBUG] mod_fax.c:607 Raw write codec activation Success L16 2009-09-15 10:41:26.433382 [DEBUG] switch_channel.c:182 sofia/external/**********@**.***.**.*** receive message [AUDIO_SYNC] 2009-09-15 10:41:26.464633 [DEBUG] switch_core_io.c:232 sofia/external/**********@**.***.**.*** receive message [TRANSCODING_NECESSARY] 2009-09-15 10:41:27.589676 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1 2009-09-15 10:41:27.761558 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 1 2009-09-15 10:41:27.792809 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1 2009-09-15 10:41:27.870937 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 1 2009-09-15 10:41:28.308454 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 1 2009-09-15 10:41:28.355331 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 1 2009-09-15 10:41:28.370956 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 1 2009-09-15 10:41:28.824099 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 1 2009-09-15 10:41:29.27231 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 1 2009-09-15 10:41:29.261615 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 1 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_A_CED, state 1 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Starting answer mode 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_A_CED to T30_PHASE_B_TX 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 1 to 17 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident 'SpanDSP Fax Ident' 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: CSI without final frame tag 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 DIS: 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= Store and forward Internet fax (T.37): Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= Real-time Internet fax (T.38): Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= 3G mobile network: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..0. ....= V.8 capabilities: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= Preferred octets: 256 octets 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= Ready to transmit a fax document (polling): Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..1.= Can receive fax: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= 2-D coding: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1% 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 10..= Recording length: Unlimited 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= Extension indicator: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= Compressed/uncompressed mode: Compressed 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .1..= Error correction mode (ECM): ECM 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .1.. ....= T.6 coding: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= Extension indicator: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= "Field not valid" supported: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= Multiple selective polling: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= Polled sub-address: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= T.43 coding: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ...0 ....= Plane interleave: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= Reserved for the use of extended voice coding set: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= Extension indicator: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...1= R8x15.4lines/mm: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= 300x300pels/25.4mm: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= Inch-based resolution preferred: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ...1 ....= Metric-based resolution preferred: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= Selective polling: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= Extension indicator: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= Sub-addressing: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= Password: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= Ready to transmit a data file (polling): Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ...0 ....= Binary file transfer (BFT): Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..0. ....= Document transfer mode (DTM): Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= Electronic data interchange (EDI): Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= Extension indicator: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= Basic transfer mode (BTM): Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= Ready to transfer a character or mixed mode document (polling): Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= Character mode: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..0. ....= Mixed mode (Annex E/T.4): Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= Extension indicator: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= Processable mode 26 (Rec. T.505): Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= Digital network capability: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= Duplex capability: Half only 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= JPEG coding: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ...0 ....= Full colour mode: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= 12bits/pel component: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= Extension indicator: Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= No subsampling (1:1:1): Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= Custom illuminant: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= Custom gamut range: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 1...= North American Letter (215.9mm x 279.4mm): Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ...1 ....= North American Legal (215.9mm x 355.6mm): Set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 0... ....= Extension indicator: Not set 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: DIS with final frame tag 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: ff 13 80 00 ee fa c4 80 95 80 80 80 18 2009-09-15 10:41:31.542953 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17 2009-09-15 10:41:31.621081 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17 2009-09-15 10:41:31.621081 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX 2009-09-15 10:41:31.621081 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 4 2009-09-15 10:41:31.621081 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0 2009-09-15 10:41:31.621081 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4 2009-09-15 10:41:31.652332 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2009-09-15 10:41:32.152351 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17 2009-09-15 10:41:32.261731 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17 2009-09-15 10:41:32.402361 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 17 2009-09-15 10:41:32.402361 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4A 2009-09-15 10:41:34.11798 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T4A (11200 remaining) 2009-09-15 10:41:34.11798 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: TSI without final frame tag 2009-09-15 10:41:34.11798 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: ff 03 43 38 37 34 31 32 36 38 34 37 35 31 20 20 20 20 20 20 20 20 20 2009-09-15 10:41:34.11798 [DEBUG] mod_fax.c:137 FLOW T.30 Remote gave TSI as: "***********" 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Stop none (0 remaining) 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: DCS with final frame tag 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: ff 13 83 00 22 f8 44 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 In state 17 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 DCS: 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= Store and forward Internet fax (T.37): Not set 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= Real-time Internet fax (T.38): Not set 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= 3G mobile network: Not set 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..1.= Receive fax: Set 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 ..10 00..= Selected data signalling rate: V.17 14400bps 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Not set 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 0... ....= 2-D coding: Not set 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..00= Recording width: 215mm +- 1% 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... 10..= Recording length: Unlimited 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .111 ....= Minimum scan line time: 0ms 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= Extension indicator: Set 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= Compressed/uncompressed mode: Compressed 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... .1..= Error correction mode (ECM): ECM 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= Frame size: 256 octets 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .1.. ....= T.6 coding: Set 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 0... ....= Extension indicator: Not set 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Selected compression 3 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Get document at 14400bps, modem 7 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 17 to 7 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2 2009-09-15 10:41:34.371187 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7 2009-09-15 10:41:34.371187 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_C_NON_ECM_RX 2009-09-15 10:41:34.371187 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0 2009-09-15 10:41:34.371187 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 7 2009-09-15 10:41:34.371187 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0 2009-09-15 10:41:34.402438 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7 2009-09-15 10:41:34.402438 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7 2009-09-15 10:41:34.574319 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7 2009-09-15 10:41:34.605571 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7 2009-09-15 10:41:34.605571 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7 2009-09-15 10:41:34.621196 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7 2009-09-15 10:41:34.621196 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7 2009-09-15 10:41:34.730575 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training in progress (-3) in state 7 2009-09-15 10:41:35.246220 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7 2009-09-15 10:41:36.27500 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training succeeded (-4) in state 7 2009-09-15 10:41:36.27500 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T2 (42080 remaining) 2009-09-15 10:41:36.27500 [DEBUG] mod_fax.c:137 FLOW FAX Switching from V.17 + V.21 to V.17 (-16.66dBm0) 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW T.30 Trainability (TCF) test result - 21822 total bits. longest run of zeros was 21600 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_C_NON_ECM_RX to T30_PHASE_B_TX 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 7 to 8 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: CFR with final frame tag 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: ff 13 84 2009-09-15 10:41:38.652601 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 8 2009-09-15 10:41:38.730729 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 8 2009-09-15 10:41:38.730729 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 8 to 12 2009-09-15 10:41:38.730729 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_C_ECM_RX 2009-09-15 10:41:38.730729 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 7 2009-09-15 10:41:38.730729 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0 2009-09-15 10:41:38.730729 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2 2009-09-15 10:41:38.746355 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 12 2009-09-15 10:41:38.793231 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Training failed (-5) in state 12 2009-09-15 10:41:38.871359 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 12 2009-09-15 10:41:38.871359 [DEBUG] mod_fax.c:137 FLOW T.30 Start T1A 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T1A (277280 remaining) 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: CFR with final frame tag 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: ff 13 84 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 In state 12 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected CFR received in state 12 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 12 to 3 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: DCN with final frame tag 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: ff 13 fa 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW FAX Switching from V.17 + V.21 to V.21 (-33.64dBm0) 2009-09-15 10:41:39.261999 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 3 2009-09-15 10:41:39.261999 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_C_ECM_RX to T30_PHASE_D_TX 2009-09-15 10:41:39.261999 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0 2009-09-15 10:41:39.261999 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4 2009-09-15 10:41:40.324540 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 2009-09-15 10:41:40.402668 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3 2009-09-15 10:41:40.402668 [DEBUG] mod_fax.c:137 FLOW T.30 Disconnecting 2009-09-15 10:41:40.402668 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E 2009-09-15 10:41:40.402668 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0 2009-09-15 10:41:40.402668 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 1 2009-09-15 10:41:40.402668 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 3 to 2 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_E, state 2 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:167 ======================================================================== ====== 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:180 Fax processing not successful - result (13) Unexpected message received. 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:185 Remote station id: *********** 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:186 Local station id: SpanDSP Fax Ident 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:187 Pages transferred: 0 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:189 Total fax pages: 0 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:190 Image resolution: 8031x3850 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:191 Transfer Rate: 14400 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:193 ECM status on 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:194 remote country: 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:195 remote vendor: 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:196 remote model: 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:198 ======================================================================== ====== 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 2 to 32 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 8 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 8 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete 2009-09-15 10:41:41.433958 [DEBUG] switch_core_codec.c:122 Restore original codec. ################################################# CFR in phase C? Hmm... Thanks again. -- Travis -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Steve Underwood Sent: Wednesday, September 16, 2009 12:51 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] faxrx error 13 Unexpected message received On 09/17/2009 12:08 AM, Travis Stutsman wrote: > In my attempts to receive a fax from a PSTN fax machine, the transaction > fails with error code 13 "Unexpected message received". Verbose logging > is on for mod_fax. Here is an exerpt: > > ################################################# > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: CFR with > final frame tag > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: ff 13 84 > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 In state 12 > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected > CFR received in state 12 > ...... > ======================================================================== > ====== > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:180 Fax processing not > successful - result (13) Unexpected message received. > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:185 Remote station id: > ********** > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:186 Local station id: > SpanDSP Fax Ident > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:187 Pages transferred: 0 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:189 Total fax pages: 0 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:190 Image resolution: > 8031x3850 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:191 Transfer Rate: > 14400 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:193 ECM status on > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:194 remote country: > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:195 remote vendor: > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:196 remote model: > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:198 > ======================================================================== > ====== > ################################################# > > I have successfully received a fax from faxzero.com on this > installation, as have I successfully sent a fax from freeswitch to the > PSTN fax machine in question. > > I've been digging, but there doesn't seem to be a whole lot of > information on faxing. I guess I could use a bit of direction. Any > input is much appreciated. > You seem to have chopped out all the interesting parts of that log. A full log might say something interesting. Steve _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jmesquita at freeswitch.org Wed Sep 16 10:27:17 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 16 Sep 2009 14:27:17 -0300 Subject: [Freeswitch-users] session record does not for very short calls In-Reply-To: References: Message-ID: I think you need to upgrade your version before we even take a look at that... You are so far behind trunk right now and lots of things have been changed since then. Not sure if this would solve your problem but not a lot of ppl will look at your problem when you run this version. jmesquita On Wed, Sep 16, 2009 at 2:00 PM, Frank @ Impact wrote: > FreeSWITCH Version 1.0.trunk (12790M) > > > > I have this in my DP > > > > > > > > > > works fine as long as the call is long enough. But if the call is only, > say, 3-4 seconds long (or something very short like that), then the wav file > is never created with the audio in it. > > > > Is there a work around for this? > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/abef54c8/attachment.html From dftoro at yahoo.com Wed Sep 16 10:44:19 2009 From: dftoro at yahoo.com (Diego Toro) Date: Wed, 16 Sep 2009 10:44:19 -0700 (PDT) Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <87f2f3b90909160756i7af013dbie657ca02cd0dec6a@mail.gmail.com> Message-ID: <644089.79676.qm@web33503.mail.mud.yahoo.com> Hi, count on me for testing and answering questions on Windows and spanish support. Diego http://lacarretade.blogspot.com/ --- On Wed, 9/16/09, Michael Collins wrote: From: Michael Collins Subject: Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects To: freeswitch-users at lists.freeswitch.org Date: Wednesday, September 16, 2009, 9:56 AM On Wed, Sep 16, 2009 at 1:53 AM, Tristan Mah? wrote: Hi, Count on me for answering questions on IRC when I'm in, and for subprojects I'm in too as you know ;) Merci! Okay, what's your IRC nick and when are you generally on line? Also, I'm pretty sure that you're fluent in French which is good because we need more multilingual people out there. Last question: what are your areas of expertise? I'd like to keep a list of people and what they're good at so we know whom to ask first when questions come up. Thanks again! -MC -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/67d1a661/attachment.html From tculjaga at gmail.com Wed Sep 16 10:55:57 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 16 Sep 2009 19:55:57 +0200 Subject: [Freeswitch-users] reloadxml question In-Reply-To: References: <65d96fc80909160649kd5ca200gdeed86239a0257d4@mail.gmail.com> <1FAB38C1-83CA-427E-ACB7-1E5CD55E9710@freeswitch.org> <65d96fc80909160717x40e8db16o302246b86ba1c@mail.gmail.com> Message-ID: <65d96fc80909161055l350471feya577bbb9fbd3f1f3@mail.gmail.com> well, it is a specific module for delivering some sort of services... Actually, we are trying to build a SIP Application Server based on freeswitch in core... the server will/should be in charge of delivering various services e.g. international call routing (route calls to international destinations according to src_number, dst_prefix, user_qos_group, asr, acd.... ) hubbing (route traffic between carriers according to whatever you want) local number portability (... no need to explain) location based services (special services e.g. call distribution for emergency services, local services ... lets say the user dials 101 and he always gets the closest/preferred store... ) legal intercept (not allowed to talk about :P) Voice VPNs Network Announcement Limited postpaid services radius authorization (no accounting) ... The idea is not to route calls through FS... it is going to be used just as a ticketing server. FS needs to respond with an appropriate SIP message containing routing decision. This way we make FS stateless as no calls are going through and in the same time it holds all the services/routing logic we need. so, there are 2 main things: 1. all relevant routing parameters are loaded into a DKA memory map 2. all user parameter queries are done towards an OpenLDAP database remember i was asking questions about performance and how to fine tune the monster :P... well that's it. i'm able to run at 400+ CPS on a single server. we will see how it will finish ... so far, we are at 80% of the project and I plan to switch some real traffic soon. T. On Wed, Sep 16, 2009 at 5:00 PM, Brian West wrote: > Might I ask what you are working on? Its interesting to hear what > people are doing with FreeSWITCH. > > /b > > On Sep 16, 2009, at 9:17 AM, Tihomir Culjaga wrote: > > > perfect, > > > > thanks. > > > > T. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/0c70e0bd/attachment.html From msc at freeswitch.org Wed Sep 16 11:23:54 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Sep 2009 11:23:54 -0700 Subject: [Freeswitch-users] originate command sofia behaviour In-Reply-To: <428d9a28388f5b40cf85a37bc100fc08.squirrel@correo.nodo50.org> References: <428d9a28388f5b40cf85a37bc100fc08.squirrel@correo.nodo50.org> Message-ID: <87f2f3b90909161123m3d8d41a7pbceae85da2697a58@mail.gmail.com> On Wed, Sep 16, 2009 at 7:26 AM, Alberto Escudero wrote: > > I will like to update the wiki to spell out clearly the differences > between this three commands > I have a IVR running in 4600 and the FS box has IP address 192.168.46.15 > > originate sofia/192.168.46.15/1001 4600 > originate sofia/internal/1001 at 192.168.46.15 4600 > originate sofia/internal/1001%192.168.46.15 4600 > > The first originate places a call as a external gateway, not until > registered phone 1001 answers the call is transfer to 4600 > > The second and third originate command triggers extension 4600 Javascript > IVR although 1001 has not answer > > Can anyone clarify me if this is the intended behavior also including the > difference between % and @ > The difference between % and @ is discussed here: http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/d0ee2a00/attachment.html From email.list.subscriber at gmail.com Wed Sep 16 11:39:23 2009 From: email.list.subscriber at gmail.com (email lists) Date: Wed, 16 Sep 2009 14:39:23 -0400 Subject: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? In-Reply-To: <032201ca368e$a1502300$e3f06900$@com> References: <4aa965af.161bf30a.61f8.7c7d@mx.google.com> <4AA9FBA5.5090403@kounitskiy.com> <191c3a030909110831o4e0a4844obf3c339d58f5358a@mail.gmail.com> <4ab00c82.161bf30a.6722.522e@mx.google.com> <032201ca368e$a1502300$e3f06900$@com> Message-ID: <4ab13092.151bf30a.3ba7.5c41@mx.google.com> Hello Mindaugas, It was not really a matter of Freeswitch + mod_radius_cdr not being good for me, or for what we needed it to do, but rather more of a resource and time constraint based decision. If we had 'C' knowledgeable resources readily available, the Freeswitch and Radius customizations required could've been completed, providing a more streamlined setup using Freeswitch alone. However, due to resource and (in this case more importantly) time constraints, I shifted to an alternative solution that I could more quickly implement to meet our immediate needs. I will eventually re-visit relying solely on Freeswitch to simplify our setup and probably ask a few more questions when that time comes. :) Vladimir From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mindaugas Kezys Sent: Wednesday, September 16, 2009 1:29 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? Can you tell why Freeswitch + mod_radius_cdr was not good for you? Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of email lists Sent: 2009 m. rugs?jo 16 d. 00:53 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? Thanks to those for the info and help on this issue. Ultimately ended up having to use alternative software for the radius piece (not related to any shortfalls by Freeswitch). Vlad From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, September 11, 2009 11:31 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? set the variable process_cdr=false on that a_leg first thing in your dialplan On Fri, Sep 11, 2009 at 2:26 AM, Anatoliy Kounitskiy wrote: It's normal to have to two records for a call - Start and Stop message. From what i see - you have one start and stop for each leg of the call. Regards, AK email lists wrote: > > Hello, > > > > Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate > RADIUS messages being generated for individual calls (sample messages > for one call below). Looking at the "Acct-Unique-Session-Id" and > "Acct-Session-Id" fields, it would appear that perhaps each call leg > results in a pair of start/stop RADIUS messages; is this the expected > behavior? If so, is there a way to disable RADIUS messaging for what > I presume is the "ingress" or A leg of the call? > > > > Any leads would be appreciated. > > > > Thanks in advance. > > > > Vladimir > > > > Thu Sep 10 10:37:25 2009 > > Acct-Status-Type = Start > > Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "097c8472ff7bcec7" > > Timestamp = 1252604245 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:37:25 2009 > > Acct-Status-Type = Start > > Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226 at x.x.x.x" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "53f729e173e8c0a9" > > Timestamp = 1252604245 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:37:57 2009 > > Acct-Status-Type = Stop > > Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" > > Freeswitch-Hangupcause = Normal-Unspecified > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Lastapp = "bridge" > > Freeswitch-Billusec = 32029926 > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" > > Freeswitch-Callanswerdate = "2009-09-10T10:22:00.319197-0700" > > Freeswitch-Callenddate = "2009-09-10T10:22:32.349123-0700" > > Acct-Session-Time = 32 > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "097c8472ff7bcec7" > > Timestamp = 1252604277 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:38:02 2009 > > Acct-Status-Type = Stop > > Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" > > Freeswitch-Hangupcause = Normal-Clearing > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226 at x.x.x.x" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Billusec = 32049973 > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" > > Freeswitch-Callanswerdate = "2009-09-10T10:22:00.289136-0700" > > Freeswitch-Callenddate = "2009-09-10T10:22:32.339109-0700" > > Acct-Session-Time = 32 > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "53f729e173e8c0a9" > > Timestamp = 1252604282 > > Request-Authenticator = Verified** > > > > ---------------------------------------------------------------------- -- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/c01adb59/attachment-0001.html From aep.lists at it46.se Wed Sep 16 11:43:44 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Wed, 16 Sep 2009 20:43:44 +0200 Subject: [Freeswitch-users] originate command sofia behaviour In-Reply-To: <87f2f3b90909161123m3d8d41a7pbceae85da2697a58@mail.gmail.com> References: <428d9a28388f5b40cf85a37bc100fc08.squirrel@correo.nodo50.org> <87f2f3b90909161123m3d8d41a7pbceae85da2697a58@mail.gmail.com> Message-ID: The problem i am facing is the following: Extension 4600 is a Javascript IVR that starts by session.aswer() I want to originate a call to leg 1 and then connected to the IVR when the leg 1 has answered. If I run originate sofia/192.168.46.15/1001 4600 call is transfer to extension 4600 *IVR* after 1001 answers the call If I run originate sofia/internal/1001 at 192.168.46.15 4600 the IVR starts BEFORE user 1001 has answered? What is the best way to: Initiate a call to leg 1 and connect it to leg 2 (the Javascript IVR) after leg 1 has answered the call? /aep -- Stopping junk mailers is good for the environment > On Wed, Sep 16, 2009 at 7:26 AM, Alberto Escudero > wrote: > >> >> I will like to update the wiki to spell out clearly the differences >> between this three commands >> I have a IVR running in 4600 and the FS box has IP address 192.168.46.15 >> >> originate sofia/192.168.46.15/1001 4600 >> originate sofia/internal/1001 at 192.168.46.15 4600 >> originate sofia/internal/1001%192.168.46.15 4600 >> >> The first originate places a call as a external gateway, not until >> registered phone 1001 answers the call is transfer to 4600 >> >> The second and third originate command triggers extension 4600 >> Javascript >> IVR although 1001 has not answer >> >> Can anyone clarify me if this is the intended behavior also including >> the >> difference between % and @ >> > > The difference between % and @ is discussed here: > http://wiki.freeswitch.org/wiki/Dialplan_XML#SIP-Specific_Dialstrings > -MC > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From shaheryarkh at googlemail.com Wed Sep 16 11:59:44 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Thu, 17 Sep 2009 00:59:44 +0600 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <644089.79676.qm@web33503.mail.mud.yahoo.com> References: <87f2f3b90909160756i7af013dbie657ca02cd0dec6a@mail.gmail.com> <644089.79676.qm@web33503.mail.mud.yahoo.com> Message-ID: I am also available for FS configuration on various Linux distributions and Wiki / documentation. Thank you. On Wed, Sep 16, 2009 at 11:44 PM, Diego Toro wrote: > Hi, count on me for testing and answering questions on Windows and spanish > support. > > Diego > http://lacarretade.blogspot.com/ > > --- On *Wed, 9/16/09, Michael Collins * wrote: > > > From: Michael Collins > Subject: Re: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With > FreeSWITCH Subprojects > To: freeswitch-users at lists.freeswitch.org > Date: Wednesday, September 16, 2009, 9:56 AM > > > > > On Wed, Sep 16, 2009 at 1:53 AM, Tristan Mah? > > wrote: > >> Hi, >> >> Count on me for answering questions on IRC when I'm in, and for >> subprojects I'm in too as you know ;) >> > Merci! > > Okay, what's your IRC nick and when are you generally on line? Also, I'm > pretty sure that you're fluent in French which is good because we need more > multilingual people out there. Last question: what are your areas of > expertise? I'd like to keep a list of people and what they're good at so we > know whom to ask first when questions come up. > > Thanks again! > -MC > > > -----Inline Attachment Follows----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/13796352/attachment.html From msc at freeswitch.org Wed Sep 16 12:01:11 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 16 Sep 2009 12:01:11 -0700 Subject: [Freeswitch-users] originate command sofia behaviour In-Reply-To: References: <428d9a28388f5b40cf85a37bc100fc08.squirrel@correo.nodo50.org> <87f2f3b90909161123m3d8d41a7pbceae85da2697a58@mail.gmail.com> Message-ID: <87f2f3b90909161201i3bb2ac5ag1e2613cfbf785d36@mail.gmail.com> On Wed, Sep 16, 2009 at 11:43 AM, Alberto Escudero wrote: > The problem i am facing is the following: > > Extension 4600 is a Javascript IVR that starts by session.aswer() > > I want to originate a call to leg 1 and then connected to the IVR when the > leg 1 has answered. > > If I run > > originate sofia/192.168.46.15/1001 4600 > call is transfer to extension 4600 *IVR* after 1001 answers the call > > If I run > originate sofia/internal/1001 at 192.168.46.15 4600 > the IVR starts BEFORE user 1001 has answered? > > What is the best way to: > > Initiate a call to leg 1 and connect it to leg 2 (the Javascript IVR) > after leg 1 has answered the call? > You can try ignoring early media to force the A-leg to answer before anything else happens. Try this and let us know if it does what you want: originate {ignore_early_media=true} sofia/internal/1001 at 192.168.46.15 4600 You can probably look at the SIP traces of the two options you've tried (without ignoring early media) to confirm that you're getting media prior to answer when doing "originate sofia/internal/1001 at 192.168.46.15 4600" - probably in one case you get a 180 and in the other a 183. Check it out and let us know. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/22e87ca8/attachment.html From mkezys at gmail.com Wed Sep 16 12:26:10 2009 From: mkezys at gmail.com (Mindaugas Kezys) Date: Wed, 16 Sep 2009 22:26:10 +0300 Subject: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? In-Reply-To: <4ab13092.151bf30a.3ba7.5c41@mx.google.com> References: <4aa965af.161bf30a.61f8.7c7d@mx.google.com> <4AA9FBA5.5090403@kounitskiy.com> <191c3a030909110831o4e0a4844obf3c339d58f5358a@mail.gmail.com> <4ab00c82.161bf30a.6722.522e@mx.google.com> <032201ca368e$a1502300$e3f06900$@com> <4ab13092.151bf30a.3ba7.5c41@mx.google.com> Message-ID: <008a01ca3703$8cd8d2d0$a68a7870$@com> Thank you for your answer. Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of email lists Sent: 2009 m. rugs?jo 16 d. 21:39 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? Hello Mindaugas, It was not really a matter of Freeswitch + mod_radius_cdr not being good for me, or for what we needed it to do, but rather more of a resource and time constraint based decision. If we had 'C' knowledgeable resources readily available, the Freeswitch and Radius customizations required could've been completed, providing a more streamlined setup using Freeswitch alone. However, due to resource and (in this case more importantly) time constraints, I shifted to an alternative solution that I could more quickly implement to meet our immediate needs. I will eventually re-visit relying solely on Freeswitch to simplify our setup and probably ask a few more questions when that time comes. :) Vladimir From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mindaugas Kezys Sent: Wednesday, September 16, 2009 1:29 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? Can you tell why Freeswitch + mod_radius_cdr was not good for you? Regards, Mindaugas Kezys http://www.kolmisoft.com VoIP Billing and Routing Solutions From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of email lists Sent: 2009 m. rugs?jo 16 d. 00:53 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? Thanks to those for the info and help on this issue. Ultimately ended up having to use alternative software for the radius piece (not related to any shortfalls by Freeswitch). Vlad From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, September 11, 2009 11:31 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_radius_cdr: Multiple RADIUS messages for single call? set the variable process_cdr=false on that a_leg first thing in your dialplan On Fri, Sep 11, 2009 at 2:26 AM, Anatoliy Kounitskiy wrote: It's normal to have to two records for a call - Start and Stop message. From what i see - you have one start and stop for each leg of the call. Regards, AK email lists wrote: > > Hello, > > > > Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate > RADIUS messages being generated for individual calls (sample messages > for one call below). Looking at the "Acct-Unique-Session-Id" and > "Acct-Session-Id" fields, it would appear that perhaps each call leg > results in a pair of start/stop RADIUS messages; is this the expected > behavior? If so, is there a way to disable RADIUS messaging for what > I presume is the "ingress" or A leg of the call? > > > > Any leads would be appreciated. > > > > Thanks in advance. > > > > Vladimir > > > > Thu Sep 10 10:37:25 2009 > > Acct-Status-Type = Start > > Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "097c8472ff7bcec7" > > Timestamp = 1252604245 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:37:25 2009 > > Acct-Status-Type = Start > > Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226 at x.x.x.x" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "53f729e173e8c0a9" > > Timestamp = 1252604245 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:37:57 2009 > > Acct-Status-Type = Stop > > Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" > > Freeswitch-Hangupcause = Normal-Unspecified > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Lastapp = "bridge" > > Freeswitch-Billusec = 32029926 > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" > > Freeswitch-Callanswerdate = "2009-09-10T10:22:00.319197-0700" > > Freeswitch-Callenddate = "2009-09-10T10:22:32.349123-0700" > > Acct-Session-Time = 32 > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "097c8472ff7bcec7" > > Timestamp = 1252604277 > > Request-Authenticator = Verified > > > > Thu Sep 10 10:38:02 2009 > > Acct-Status-Type = Stop > > Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" > > Freeswitch-Hangupcause = Normal-Clearing > > User-Name = "8135793256" > > Freeswitch-Src = "8135793256" > > Freeswitch-CLID = "sipp" > > Freeswitch-Dst = "14043297226 at x.x.x.x" > > Freeswitch-Dialplan = "XML" > > Framed-IP-Address = 50.46.50.55 > > Freeswitch-Context = "public" > > Freeswitch-Source = "mod_sofia" > > Freeswitch-Billusec = 32049973 > > Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" > > Freeswitch-Callanswerdate = "2009-09-10T10:22:00.289136-0700" > > Freeswitch-Callenddate = "2009-09-10T10:22:32.339109-0700" > > Acct-Session-Time = 32 > > NAS-Port = 0 > > Acct-Delay-Time = 0 > > NAS-IP-Address = 1.1.1.1 > > Acct-Unique-Session-Id = "53f729e173e8c0a9" > > Timestamp = 1252604282 > > Request-Authenticator = Verified** > > > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/0322417e/attachment-0001.html From siniypin at gmail.com Wed Sep 16 10:56:48 2009 From: siniypin at gmail.com (=?KOI8-R?B?8s/CxdLUIPTXxdLJ1M7F0g==?=) Date: Wed, 16 Sep 2009 21:56:48 +0400 Subject: [Freeswitch-users] mod_conference performance Message-ID: <2160023e0909161056p4338cc19rb982400ebf677c1d@mail.gmail.com> Hi guys! I've tested FreeSWITCH conference module performance trying to figure out maximum number of simultaneous calls my FS box can serve. It took all 100% of CPU with only 50 calls (in average depending on conference rate) and "leaking stream handle" messages started appearing. The environment I was testing in: OS - Windows Server 2007 SP1 64 Bit CPU - Dual-core AMD Opteron 1216 HE 2.4 GHz RAM - 2 GB FreeSwitch version 1.0.4 (14460) I've written a test program that used to originate calls once in 5 seconds from the other box. These calls were routed to particular conference room I was testing. I had a number of rooms with different rate (8000-32000) and interval (20,30) settings and with perpetual-sound turned on steraming music continiously. I've switched off all unnecessary modules, but left logging on in order to trace what was happening later. Client test softphone used respective speex codec according to conference room rate. This is a dialplan I used: My questions are: Do you know any way I can increase my FS conference capacity? What do I have to tune in FS or in my box? Best regards, Robert. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/549ebc2d/attachment.html From aep.lists at it46.se Wed Sep 16 13:34:50 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Wed, 16 Sep 2009 22:34:50 +0200 Subject: [Freeswitch-users] originate command sofia behaviour In-Reply-To: <87f2f3b90909161201i3bb2ac5ag1e2613cfbf785d36@mail.gmail.com> References: <428d9a28388f5b40cf85a37bc100fc08.squirrel@correo.nodo50.org> <87f2f3b90909161123m3d8d41a7pbceae85da2697a58@mail.gmail.com> <87f2f3b90909161201i3bb2ac5ag1e2613cfbf785d36@mail.gmail.com> Message-ID: Yes, it did work! No we do not need to pay for several GSM calls to test a IVR script! /aep and gmaruzz -- Stopping junk mailers is good for the environment > On Wed, Sep 16, 2009 at 11:43 AM, Alberto Escudero > wrote: > >> The problem i am facing is the following: >> >> Extension 4600 is a Javascript IVR that starts by session.aswer() >> >> I want to originate a call to leg 1 and then connected to the IVR when >> the >> leg 1 has answered. >> >> If I run >> >> originate sofia/192.168.46.15/1001 4600 >> call is transfer to extension 4600 *IVR* after 1001 answers the call >> >> If I run >> originate sofia/internal/1001 at 192.168.46.15 4600 >> the IVR starts BEFORE user 1001 has answered? >> >> What is the best way to: >> >> Initiate a call to leg 1 and connect it to leg 2 (the Javascript IVR) >> after leg 1 has answered the call? >> > > You can try ignoring early media to force the A-leg to answer before > anything else happens. Try this and let us know if it does what you want: > originate {ignore_early_media=true} sofia/internal/1001 at 192.168.46.15 4600 > > You can probably look at the SIP traces of the two options you've tried > (without ignoring early media) to confirm that you're getting media prior > to > answer when doing "originate sofia/internal/1001 at 192.168.46.15 4600" - > probably in one case you get a 180 and in the other a 183. Check it out > and > let us know. :) > -MC > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gerry at pstn2.net Wed Sep 16 14:21:07 2009 From: gerry at pstn2.net (Gerry Hull) Date: Wed, 16 Sep 2009 17:21:07 -0400 Subject: [Freeswitch-users] Vestec Speech Recognition Integration ? Message-ID: <98a86adf0909161421g5b51c434ofeb01f409b4a85f6@mail.gmail.com> Has anyone integrated Vestec Speech Recognition with FreeSwitch? It's $99/port... http://www.vestec.ca/ They have a C/C++ api, looks pretty simple. Alas, no MRCP until 2010. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/8ca7fc42/attachment.html From jmesquita at freeswitch.org Wed Sep 16 14:52:33 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Wed, 16 Sep 2009 18:52:33 -0300 Subject: [Freeswitch-users] mod_conference performance In-Reply-To: <2160023e0909161056p4338cc19rb982400ebf677c1d@mail.gmail.com> References: <2160023e0909161056p4338cc19rb982400ebf677c1d@mail.gmail.com> Message-ID: I would be really interested to replay your test on Linux. Would you be willing to provide me all the details and relevant files so I can reproduce the test with a Linux box here? If yes, contact me offlist and we can work together on this. Regards, jmesquita On Wed, Sep 16, 2009 at 2:56 PM, ?????? ????????? wrote: > Hi guys! > > I've tested FreeSWITCH conference module performance trying to figure out > maximum number of simultaneous calls my FS box can serve. It took all 100% > of CPU with only 50 calls (in average depending on conference rate) and > "leaking stream handle" messages started appearing. > > The environment I was testing in: > OS - Windows Server 2007 SP1 64 Bit > CPU - Dual-core AMD Opteron 1216 HE 2.4 GHz > RAM - 2 GB > FreeSwitch version 1.0.4 (14460) > > I've written a test program that used to originate calls once in 5 seconds > from the other box. These calls were routed to particular conference room I > was testing. I had a number of rooms with different rate (8000-32000) and > interval (20,30) settings and with perpetual-sound turned on steraming music > continiously. I've switched off all unnecessary modules, but left logging on > in order to trace what was happening later. Client test softphone used > respective speex codec according to conference room rate. > > This is a dialplan I used: > > break="on-true"> > > > break="on-true"> > > > break="on-true"> > > > break="on-true"> > > > break="on-true"> > > > break="on-true"> > > > > > My questions are: > Do you know any way I can increase my FS conference capacity? What do I > have to tune in FS or in my box? > > Best regards, Robert. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/dc08e33c/attachment.html From brian at freeswitch.org Wed Sep 16 15:04:34 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Sep 2009 17:04:34 -0500 Subject: [Freeswitch-users] mod_conference performance In-Reply-To: References: <2160023e0909161056p4338cc19rb982400ebf677c1d@mail.gmail.com> Message-ID: <39B852C6-5FF5-4568-898D-DF63C8ECC2B8@freeswitch.org> I would be very interested in this also. /b On Sep 16, 2009, at 4:52 PM, Jo?o Mesquita wrote: > I would be really interested to replay your test on Linux. Would you > be willing to provide me all the details and relevant files so I can > reproduce the test with a Linux box here? > > If yes, contact me offlist and we can work together on this. > > Regards, > > jmesquita From jaybinks at gmail.com Wed Sep 16 15:11:19 2009 From: jaybinks at gmail.com (Jay Binks) Date: Thu, 17 Sep 2009 08:11:19 +1000 Subject: [Freeswitch-users] mod_conference performance In-Reply-To: <2160023e0909161056p4338cc19rb982400ebf677c1d@mail.gmail.com> References: <2160023e0909161056p4338cc19rb982400ebf677c1d@mail.gmail.com> Message-ID: A few thing stuck out to me ... Mainly 50 calls and transcoding speex. Try it again with g711 and see how you go. Also not sure windows 7 is going to perform as good as other options, could be wrong though . Jay On 17/09/2009, at 3:56, ?????? ????????? wrote: > Hi guys! > > I've tested FreeSWITCH conference module performance trying to > figure out maximum number of simultaneous calls my FS box can serve. > It took all 100% of CPU with only 50 calls (in average depending on > conference rate) and "leaking stream handle" messages started > appearing. > > The environment I was testing in: > OS - Windows Server 2007 SP1 64 Bit > CPU - Dual-core AMD Opteron 1216 HE 2.4 GHz > RAM - 2 GB > FreeSwitch version 1.0.4 (14460) > > I've written a test program that used to originate calls once in 5 > seconds from the other box. These calls were routed to particular > conference room I was testing. I had a number of rooms with > different rate (8000-32000) and interval (20,30) settings and with > perpetual-sound turned on steraming music continiously. I've > switched off all unnecessary modules, but left logging on in order > to trace what was happening later. Client test softphone used > respective speex codec according to conference room rate. > > This is a dialplan I used: > > > > > > > > > > > > > > > data="$1 at ultrawideband20"/> > > > data="$1 at ultrawideband30"/> > > > > My questions are: > Do you know any way I can increase my FS conference capacity? What > do I have to tune in FS or in my box? > > Best regards, Robert. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Wed Sep 16 15:36:02 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 16 Sep 2009 17:36:02 -0500 Subject: [Freeswitch-users] mod_conference performance In-Reply-To: References: <2160023e0909161056p4338cc19rb982400ebf677c1d@mail.gmail.com> Message-ID: <0A6D2C0F-BCB8-4E54-B6A0-692AB95DAEE7@freeswitch.org> Yah speex is a cpu hog (while you can tune it to use less)! Granted it uses less bandwidth but on the server side it doesn't scale very well. /b On Sep 16, 2009, at 5:11 PM, Jay Binks wrote: > A few thing stuck out to me ... > Mainly 50 calls and transcoding speex. > > Try it again with g711 and see how you go. > > Also not sure windows 7 is going to perform as good as other options, > could be wrong though . > > Jay From christian.loeschenkohl at xpirio.com Wed Sep 16 16:20:26 2009 From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=) Date: Thu, 17 Sep 2009 01:20:26 +0200 Subject: [Freeswitch-users] local stream problem - moh Message-ID: <4AB172BA.2030202@xpirio.com> hello since installing the latest trunk 14894 my local streams / moh don't work anymore no config file has changed, the files are in place ---- show_local_stream outputs default,/opt/freeswitch/sounds/music/8000 moh/16000,/opt/freeswitch/sounds/music/16000 moh/8000,/opt/freeswitch/sounds/music/8000 ---- console prints 2009-09-17 01:15:01.496277 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.516281 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.516281 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.536282 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.536282 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.556286 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.556286 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.576286 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.576286 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.596283 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.596283 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.616261 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.616261 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.636304 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.636304 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.656273 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.656273 [ERR] switch_core_file.c:152 File [(null)] not created! ----- and yes, files are there ls /opt/freeswitch/sounds/music/8000 danza-espanola-op-37-h-142-xii-arabesca.wav partita-no-3-in-e-major-bwv-1006-1-preludio.wav ponce-preludio-in-e-major.wav suite-espanola-op-47-leyenda.wav ----- br -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com From dujinfang at gmail.com Wed Sep 16 16:46:17 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 17 Sep 2009 07:46:17 +0800 Subject: [Freeswitch-users] session record does not for very short calls In-Reply-To: References: Message-ID: <3CF5F946-4307-451E-8E6E-CBC597B5ABFD@gmail.com> I think the file was there but deleted by FreeSWITCH if it thinks it was too short (like 3 seconds?). If I'm not wrong, someone requested this feature becuase FreeSWITCH left too many small recordings. On Sep 17, 2009, at 1:27 AM, Jo?o Mesquita wrote: > I think you need to upgrade your version before we even take a look > at that... You are so far behind trunk right now and lots of things > have been changed since then. > > Not sure if this would solve your problem but not a lot of ppl will > look at your problem when you run this version. > > jmesquita > > On Wed, Sep 16, 2009 at 2:00 PM, Frank @ Impact > wrote: > FreeSWITCH Version 1.0.trunk (12790M) > > > I have this in my DP > > > > > > > > > works fine as long as the call is long enough. But if the call is > only, say, 3-4 seconds long (or something very short like that), > then the wav file is never created with the audio in it. > > > Is there a work around for this? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From steveu at coppice.org Wed Sep 16 16:51:09 2009 From: steveu at coppice.org (Steve Underwood) Date: Thu, 17 Sep 2009 07:51:09 +0800 Subject: [Freeswitch-users] faxrx error 13 Unexpected message received In-Reply-To: <5991020B1C443E459410B16D7B08F33E03087B@dc1.bossproductsinc.com> References: <5991020B1C443E459410B16D7B08F33E030879@dc1.bossproductsinc.com> <4AB11772.4020904@coppice.org> <5991020B1C443E459410B16D7B08F33E03087B@dc1.bossproductsinc.com> Message-ID: <4AB179ED.1060701@coppice.org> Hi Travis, That's a pretty weird call. It looks like you have a long delayed echo. See below. On 09/17/2009 01:21 AM, Travis Stutsman wrote: > Alrighty. Here is mod_fax from beginning to end. > > > ################################################# > 2009-09-15 10:41:26.433382 [DEBUG] mod_fax.c:591 Raw read codec > activation Success L16 20000 > 2009-09-15 10:41:26.433382 [DEBUG] mod_fax.c:607 Raw write codec > activation Success L16 > 2009-09-15 10:41:26.433382 [DEBUG] switch_channel.c:182 > sofia/external/**********@**.***.**.*** receive message [AUDIO_SYNC] > 2009-09-15 10:41:26.464633 [DEBUG] switch_core_io.c:232 > sofia/external/**********@**.***.**.*** receive message > [TRANSCODING_NECESSARY] > 2009-09-15 10:41:27.589676 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Carrier up (-2) in state 1 > 2009-09-15 10:41:27.761558 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Carrier down (-1) in state 1 > 2009-09-15 10:41:27.792809 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Carrier up (-2) in state 1 > 2009-09-15 10:41:27.870937 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Abort (-8) in state 1 > 2009-09-15 10:41:28.308454 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Abort (-8) in state 1 > 2009-09-15 10:41:28.355331 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Abort (-8) in state 1 > 2009-09-15 10:41:28.370956 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Abort (-8) in state 1 > 2009-09-15 10:41:28.824099 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Abort (-8) in state 1 > 2009-09-15 10:41:29.27231 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Abort (-8) in state 1 > 2009-09-15 10:41:29.261615 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Abort (-8) in state 1 > 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete > in phase T30_PHASE_A_CED, state 1 > 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Starting > answer mode > 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > phase T30_PHASE_A_CED to T30_PHASE_B_TX > 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0 > 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4 > 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2 > 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > state 1 to 17 > 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident > 'SpanDSP Fax Ident' > 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: CSI > without final frame tag > 2009-09-15 10:41:29.511625 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: ff 03 40 > 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20 > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete > in phase T30_PHASE_B_TX, state 17 > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 DIS: > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= > Store and forward Internet fax (T.37): Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= > Real-time Internet fax (T.38): Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= > 3G mobile network: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..0. ....= > V.8 capabilities: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= > Preferred octets: 256 octets > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= > Ready to transmit a fax document (polling): Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..1.= > Can receive fax: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..10 11..= > Supported data signalling rates: V.27 ter, V.29, and V.17 > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .1.. ....= > R8x7.7lines/mm and/or 200x200pels/25.4mm: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= > 2-D coding: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..10= > Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1% > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 10..= > Recording length: Unlimited > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .111 ....= > Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85 > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= > Extension indicator: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= > Compressed/uncompressed mode: Compressed > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .1..= > Error correction mode (ECM): ECM > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .1.. ....= > T.6 coding: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= > Extension indicator: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= > "Field not valid" supported: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= > Multiple selective polling: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= > Polled sub-address: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= > T.43 coding: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ...0 ....= > Plane interleave: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..0. ....= > Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= > Reserved for the use of extended voice coding set: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= > Extension indicator: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...1= > R8x15.4lines/mm: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= > 300x300pels/25.4mm: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .1..= > R16x15.4lines/mm and/or 400x400pels/25.4mm: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= > Inch-based resolution preferred: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ...1 ....= > Metric-based resolution preferred: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..0. ....= > Minimum scan line time for higher resolutions: T15.4 = T7.7 > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= > Selective polling: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= > Extension indicator: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= > Sub-addressing: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= > Password: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= > Ready to transmit a data file (polling): Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ...0 ....= > Binary file transfer (BFT): Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..0. ....= > Document transfer mode (DTM): Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= > Electronic data interchange (EDI): Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= > Extension indicator: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= > Basic transfer mode (BTM): Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= > Ready to transfer a character or mixed mode document (polling): Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= > Character mode: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..0. ....= > Mixed mode (Annex E/T.4): Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= > Extension indicator: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= > Processable mode 26 (Rec. T.505): Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= > Digital network capability: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= > Duplex capability: Half only > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= > JPEG coding: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ...0 ....= > Full colour mode: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= > 12bits/pel component: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= > Extension indicator: Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= > No subsampling (1:1:1): Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= > Custom illuminant: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= > Custom gamut range: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .... 1...= > North American Letter (215.9mm x 279.4mm): Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ...1 ....= > North American Legal (215.9mm x 355.6mm): Set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 ..0. ....= > Single-progression sequential coding (Rec. T.85) basic: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= > Single-progression sequential coding (Rec. T.85) optional L0: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 0... ....= > Extension indicator: Not set > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: DIS with > final frame tag > 2009-09-15 10:41:31.89811 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: ff 13 80 > 00 ee fa c4 80 95 80 80 80 18 > 2009-09-15 10:41:31.542953 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete > in phase T30_PHASE_B_TX, state 17 > 2009-09-15 10:41:31.621081 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete > in phase T30_PHASE_B_TX, state 17 > 2009-09-15 10:41:31.621081 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > phase T30_PHASE_B_TX to T30_PHASE_B_RX > 2009-09-15 10:41:31.621081 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 4 > 2009-09-15 10:41:31.621081 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0 > 2009-09-15 10:41:31.621081 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4 > 2009-09-15 10:41:31.652332 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Carrier up (-2) in state 17 > 2009-09-15 10:41:32.152351 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Carrier down (-1) in state 17 > 2009-09-15 10:41:32.261731 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Carrier up (-2) in state 17 > 2009-09-15 10:41:32.402361 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Framing OK (-6) in state 17 > 2009-09-15 10:41:32.402361 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4A > 2009-09-15 10:41:34.11798 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T4A > (11200 remaining) > 2009-09-15 10:41:34.11798 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: TSI > without final frame tag > 2009-09-15 10:41:34.11798 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: ff 03 43 > 38 37 34 31 32 36 38 34 37 35 31 20 20 20 20 20 20 20 20 20 > 2009-09-15 10:41:34.11798 [DEBUG] mod_fax.c:137 FLOW T.30 Remote gave > TSI as: "***********" > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Stop none (0 > remaining) > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: DCS with > final frame tag > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: ff 13 83 > 00 22 f8 44 > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 In state 17 > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 DCS: > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... ...0= > Store and forward Internet fax (T.37): Not set > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... .0..= > Real-time Internet fax (T.38): Not set > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= > 3G mobile network: Not set > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..1.= > Receive fax: Set > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 ..10 00..= > Selected data signalling rate: V.17 14400bps > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .0.. ....= > R8x7.7lines/mm and/or 200x200pels/25.4mm: Not set > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 0... ....= > 2-D coding: Not set > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..00= > Recording width: 215mm +- 1% > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... 10..= > Recording length: Unlimited > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .111 ....= > Minimum scan line time: 0ms > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 1... ....= > Extension indicator: Set > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... ..0.= > Compressed/uncompressed mode: Compressed > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... .1..= > Error correction mode (ECM): ECM > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .... 0...= > Frame size: 256 octets > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 .1.. ....= > T.6 coding: Set > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 0... ....= > Extension indicator: Not set > They want to send us a FAX in ECM mode at 14,400bps. > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Selected > compression 3 > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Get document > at 14400bps, modem 7 > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > state 17 to 7 > 2009-09-15 10:41:34.293059 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2 > 2009-09-15 10:41:34.371187 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Carrier down (-1) in state 7 > 2009-09-15 10:41:34.371187 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > phase T30_PHASE_B_RX to T30_PHASE_C_NON_ECM_RX > 2009-09-15 10:41:34.371187 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0 > 2009-09-15 10:41:34.371187 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 7 > 2009-09-15 10:41:34.371187 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0 > 2009-09-15 10:41:34.402438 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM > signal status is Carrier up (-2) in state 7 > 2009-09-15 10:41:34.402438 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Carrier up (-2) in state 7 > 2009-09-15 10:41:34.574319 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM > signal status is Training failed (-5) in state 7 > 2009-09-15 10:41:34.605571 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM > signal status is Carrier down (-1) in state 7 > 2009-09-15 10:41:34.605571 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Carrier down (-1) in state 7 > 2009-09-15 10:41:34.621196 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM > signal status is Carrier up (-2) in state 7 > 2009-09-15 10:41:34.621196 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Carrier up (-2) in state 7 > 2009-09-15 10:41:34.730575 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM > signal status is Training in progress (-3) in state 7 > 2009-09-15 10:41:35.246220 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Abort (-8) in state 7 > 2009-09-15 10:41:36.27500 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal > status is Training succeeded (-4) in state 7 > 2009-09-15 10:41:36.27500 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T2 (42080 > remaining) > 2009-09-15 10:41:36.27500 [DEBUG] mod_fax.c:137 FLOW FAX Switching from > V.17 + V.21 to V.17 (-16.66dBm0) > 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM > signal status is Carrier down (-1) in state 7 > 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW T.30 Trainability > (TCF) test result - 21822 total bits. longest run of zeros was 21600 > They've sent us perfectly good training test data (TCF) > 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > phase T30_PHASE_C_NON_ECM_RX to T30_PHASE_B_TX > 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0 > 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4 > 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > state 7 to 8 > 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: CFR with > final frame tag > 2009-09-15 10:41:37.574435 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: ff 13 84 > 2009-09-15 10:41:38.652601 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete > in phase T30_PHASE_B_TX, state 8 > We have sent the confirmation that the training test went OK > 2009-09-15 10:41:38.730729 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete > in phase T30_PHASE_B_TX, state 8 > 2009-09-15 10:41:38.730729 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > state 8 to 12 > 2009-09-15 10:41:38.730729 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > phase T30_PHASE_B_TX to T30_PHASE_C_ECM_RX > 2009-09-15 10:41:38.730729 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 7 > 2009-09-15 10:41:38.730729 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0 > 2009-09-15 10:41:38.730729 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2 > 2009-09-15 10:41:38.746355 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Carrier up (-2) in state 12 > 2009-09-15 10:41:38.793231 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Training failed (-5) in state 12 > 2009-09-15 10:41:38.871359 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Framing OK (-6) in state 12 > 2009-09-15 10:41:38.871359 [DEBUG] mod_fax.c:137 FLOW T.30 Start T1A > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T1A > (277280 remaining) > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: CFR with > final frame tag > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Rx: ff 13 84 > About one second later we have received a training confirmation message. Presumably that is an echo of the one we sent. The echo delay must have been quite big, though, or we would not have interpreted that echo as a received message. > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 In state 12 > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected > CFR received in state 12 > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > state 12 to 3 > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: DCN with > final frame tag > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW T.30 Tx: ff 13 fa > Having received a message that is out of sequence, we just disconnect. > 2009-09-15 10:41:39.215123 [DEBUG] mod_fax.c:137 FLOW FAX Switching from > V.17 + V.21 to V.21 (-33.64dBm0) > 2009-09-15 10:41:39.261999 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal > status is Carrier down (-1) in state 3 > 2009-09-15 10:41:39.261999 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > phase T30_PHASE_C_ECM_RX to T30_PHASE_D_TX > 2009-09-15 10:41:39.261999 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0 > 2009-09-15 10:41:39.261999 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4 > 2009-09-15 10:41:40.324540 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete > in phase T30_PHASE_D_TX, state 3 > 2009-09-15 10:41:40.402668 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete > in phase T30_PHASE_D_TX, state 3 > 2009-09-15 10:41:40.402668 [DEBUG] mod_fax.c:137 FLOW T.30 Disconnecting > 2009-09-15 10:41:40.402668 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > phase T30_PHASE_D_TX to T30_PHASE_E > 2009-09-15 10:41:40.402668 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0 > 2009-09-15 10:41:40.402668 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 1 > 2009-09-15 10:41:40.402668 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > state 3 to 2 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete > in phase T30_PHASE_E, state 2 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:167 > ======================================================================== > ====== > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:180 Fax processing not > successful - result (13) Unexpected message received. > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:185 Remote station id: > *********** > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:186 Local station id: > SpanDSP Fax Ident > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:187 Pages transferred: 0 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:189 Total fax pages: 0 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:190 Image resolution: > 8031x3850 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:191 Transfer Rate: > 14400 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:193 ECM status on > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:194 remote country: > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:195 remote vendor: > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:196 remote model: > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:198 > ======================================================================== > ====== > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > state 2 to 32 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from > phase T30_PHASE_E to T30_PHASE_CALL_FINISHED > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 8 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange > complete > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 8 > 2009-09-15 10:41:41.402707 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange > complete > 2009-09-15 10:41:41.433958 [DEBUG] switch_core_codec.c:122 Restore > original codec. > ################################################# > What kind of path was this call over? Steve From nandy1925 at gmail.com Wed Sep 16 17:05:49 2009 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Thu, 17 Sep 2009 08:05:49 +0800 Subject: [Freeswitch-users] session record does not for very short calls In-Reply-To: <3CF5F946-4307-451E-8E6E-CBC597B5ABFD@gmail.com> References: <3CF5F946-4307-451E-8E6E-CBC597B5ABFD@gmail.com> Message-ID: <7d0bfd8c0909161705y5e870521g8a6181d9d7715904@mail.gmail.com> this makes sense. a workaround would be to provide an optional variable to delete recording file if it's less than N seconds. otherwise, it defaults to a preset duration. /nandy On Thu, Sep 17, 2009 at 7:46 AM, Seven Du wrote: > I think the file was there but deleted by FreeSWITCH if it thinks it > was too short (like 3 seconds?). If I'm not wrong, someone requested > this feature becuase FreeSWITCH left too many small recordings. > > > On Sep 17, 2009, at 1:27 AM, Jo?o Mesquita wrote: > > I think you need to upgrade your version before we even take a look > > at that... You are so far behind trunk right now and lots of things > > have been changed since then. > > > > Not sure if this would solve your problem but not a lot of ppl will > > look at your problem when you run this version. > > > > jmesquita > > > > On Wed, Sep 16, 2009 at 2:00 PM, Frank @ Impact > > wrote: > > FreeSWITCH Version 1.0.trunk (12790M) > > > > > > I have this in my DP > > > > > > > > > > > > > > > > > > works fine as long as the call is long enough. But if the call is > > only, say, 3-4 seconds long (or something very short like that), > > then the wav file is never created with the audio in it. > > > > > > Is there a work around for this? > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/cef32f91/attachment.html From anthony.minessale at gmail.com Wed Sep 16 17:33:13 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Sep 2009 19:33:13 -0500 Subject: [Freeswitch-users] local stream problem - moh In-Reply-To: <4AB172BA.2030202@xpirio.com> References: <4AB172BA.2030202@xpirio.com> Message-ID: <191c3a030909161733j2de7f393j8a53f928b6c5e587@mail.gmail.com> Update again, already fixed On Sep 16, 2009 6:25 PM, "Christian L?schenkohl" < christian.loeschenkohl at xpirio.com> wrote: hello since installing the latest trunk 14894 my local streams / moh don't work anymore no config file has changed, the files are in place ---- show_local_stream outputs default,/opt/freeswitch/sounds/music/8000 moh/16000,/opt/freeswitch/sounds/music/16000 moh/8000,/opt/freeswitch/sounds/music/8000 ---- console prints 2009-09-17 01:15:01.496277 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.516281 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.516281 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.536282 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.536282 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.556286 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.556286 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.576286 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.576286 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.596283 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.596283 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.616261 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.616261 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.636304 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.636304 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.656273 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.656273 [ERR] switch_core_file.c:152 File [(null)] not created! ----- and yes, files are there ls /opt/freeswitch/sounds/music/8000 danza-espanola-op-37-h-142-xii-arabesca.wav partita-no-3-in-e-major-bwv-1006-1-preludio.wav ponce-preludio-in-e-major.wav suite-espanola-op-47-leyenda.wav ----- br -- Ing. Christian L?schenkohl 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.loeschenkohl at xpirio.com _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/1a7b5989/attachment.html From kjv at ken-ton.com Wed Sep 16 20:10:14 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Wed, 16 Sep 2009 23:10:14 -0400 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> <8A58CD3E-F0C3-4BAD-8DB9-3C9D3A22A70F@ken-ton.com> <191c3a030909140617v396063e7n86cd8fdb39d75fa9@mail.gmail.com> Message-ID: <0E10F348-B983-48F9-87DB-19BE52CEFD50@ken-ton.com> Folks; I give credit where credit is due, and I thank Brian K. West What For: This was found to be a compounded problem. (Cisco was part of it... But the real problem was the linux kernel...) Suffice it to say, without the kernel bug, the cisco bug wouldn't have been easily found. What Kernel Bug: It's a kernel bug that corrupted the sqlite database. This caused Freeswitch to refuse the phones registration request. This in turn caused the phones to re-register. Problem was, with 10 phones, 6 lines each, perpetually registering on a 100Mbps LAN, well, you can imagine the overhead. This created severe latency with Freeswitch, and manifested as dropped calls, one way audio, and the more phones you had, the worse the problem was. Workaround for problem was to use a ramdisk (tmpfs) for the database - (Big Thanks to bkw!) So far, it's been 24 hours, and all systems are nice and stable. (no negative reports yet (fingers crossed)). Brian (and Folks); If this is stable through Friday (and there's no reason to think it won't be), I will take the time to document the problem, basic configuration, and the workaround for the problem on the Wiki this weekend. Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Sep 14, 2009, at 9:22 AM, Brian West wrote: > HAHA I couldn't have said this better! > > /b > > On Sep 14, 2009, at 8:17 AM, Anthony Minessale wrote: > >> The first hint was when the firmware rev began with the letters POS > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/9cd80965/attachment.html From anthony.minessale at gmail.com Wed Sep 16 20:17:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 16 Sep 2009 22:17:03 -0500 Subject: [Freeswitch-users] FS Presence Implementation In-Reply-To: References: <65AA86A7106D4255A5734520F55D3C50@greyhawk.tonecommander.com> <191c3a030909151153i485e5f7v876db4cb6e95600a@mail.gmail.com> Message-ID: <191c3a030909162017l5e978238lde467ebdde098ee7@mail.gmail.com> we support both application/dialog-info+xml (snom maybe a few others, I can't keep track) and pidf used by polycom and eyebeam On Wed, Sep 16, 2009 at 10:50 AM, Jerry Richards wrote: > I think you're referring to the SIP SIMPLE implementation as the default > FS presence mechanism. This is fine and I can use that protocol. The > question I still have regards the plain text content in the body of the SIP > MESSAGE method. What is the format of this plain text for presence that is > compatible with the FS implementation? > > Best Regards, > Jerry > > > ------------------------------ > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* Tuesday, September 15, 2009 11:53 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] FS Presence Implementation > > the default config ships with presence enabled for SIP > if you have a phone that supports it, all you have to do is enable it on > the phone. > > > On Tue, Sep 15, 2009 at 1:30 PM, Jerry Richards < > jerry.richards at teotech.com> wrote: > >> >> Also, is presence conveyed as any string? Or is presence a predefined >> list >> of status? >> >> Best Regards, >> Jerry >> >> >> -----Original Message----- >> From: Jerry Richards [mailto:jerry.richards at teotech.com] >> Sent: Tuesday, September 15, 2009 8:46 AM >> To: 'freeswitch-users at lists.freeswitch.org' >> Subject: FS Presence Implementation >> >> I would like to modify my SIP phone and my gateway to convey/exchange >> presence information. Could someone point me toward the FS presence >> documentation? I've seen bits and pieces. Also, I think presence can be >> communicated via more than one protocol. >> >> Thanks And Best Regards, >> Jerry >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090916/0c698722/attachment-0001.html From jason at jasonjgw.net Wed Sep 16 21:07:42 2009 From: jason at jasonjgw.net (Jason White) Date: Thu, 17 Sep 2009 14:07:42 +1000 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: <0E10F348-B983-48F9-87DB-19BE52CEFD50@ken-ton.com> References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> <8A58CD3E-F0C3-4BAD-8DB9-3C9D3A22A70F@ken-ton.com> <191c3a030909140617v396063e7n86cd8fdb39d75fa9@mail.gmail.com> <0E10F348-B983-48F9-87DB-19BE52CEFD50@ken-ton.com> Message-ID: <20090917040742.GA24786@jdc.jasonjgw.net> Karl Vesterling wrote: > What Kernel Bug: > It's a kernel bug that corrupted the sqlite database. > This caused Freeswitch to refuse the phones registration request. Please take this up with your Linux distribution as a bug report related to the kernel, and persist with it until it's sorted out. The more that users do this, the more kernel bugs will get fixed. We're all responsible to some extent for the quality of our free/open-source operating systems. From luismzuccolo at yahoo.com.ar Wed Sep 16 21:34:30 2009 From: luismzuccolo at yahoo.com.ar (Luis M. Zuccolo) Date: Thu, 17 Sep 2009 01:34:30 -0300 Subject: [Freeswitch-users] Compile error Message-ID: <1253162070.3583.14.camel@localhost.localdomain> Hi: Since svn version 13523 to current I get this error: make[5]: swig: Command not found make[5]: *** [mod_lua_wrap.cpp] Error 127 make[4]: *** [all] Error 1 make[3]: *** [mod_lua-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 Was there any change from that version? Thanks in advance __________________________________________________ Correo Yahoo! Espacio para todos tus mensajes, antivirus y antispam ?gratis! ?Abr? tu cuenta ya! - http://correo.yahoo.com.ar From tuyanozipek at gmail.com Wed Sep 16 21:53:23 2009 From: tuyanozipek at gmail.com (=?ISO-8859-1?Q?Tuyan_=D6zipek?=) Date: Thu, 17 Sep 2009 00:53:23 -0400 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: <20090917040742.GA24786@jdc.jasonjgw.net> References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> <8A58CD3E-F0C3-4BAD-8DB9-3C9D3A22A70F@ken-ton.com> <191c3a030909140617v396063e7n86cd8fdb39d75fa9@mail.gmail.com> <0E10F348-B983-48F9-87DB-19BE52CEFD50@ken-ton.com> <20090917040742.GA24786@jdc.jasonjgw.net> Message-ID: <881082ae0909162153p38935967xa3ca3e2772c6a1d@mail.gmail.com> Which distro is this? /tyn On Thu, Sep 17, 2009 at 12:07 AM, Jason White wrote: > Karl Vesterling wrote: > >> What Kernel Bug: >> It's a kernel bug that corrupted the sqlite database. >> This caused Freeswitch to refuse the phones registration request. > > Please take this up with your Linux distribution as a bug report related to > the kernel, and persist with it until it's sorted out. > > The more that users do this, the more kernel bugs will get fixed. > > We're all responsible to some extent for the quality of our free/open-source > operating systems. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gabe at gundy.org Wed Sep 16 22:36:52 2009 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 16 Sep 2009 23:36:52 -0600 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: <0E10F348-B983-48F9-87DB-19BE52CEFD50@ken-ton.com> References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> <8A58CD3E-F0C3-4BAD-8DB9-3C9D3A22A70F@ken-ton.com> <191c3a030909140617v396063e7n86cd8fdb39d75fa9@mail.gmail.com> <0E10F348-B983-48F9-87DB-19BE52CEFD50@ken-ton.com> Message-ID: <903da5680909162236i60fd2648s4a67751cdc8eae46@mail.gmail.com> On Wed, Sep 16, 2009 at 9:10 PM, Karl Vesterling wrote: > It's a kernel bug that corrupted the sqlite database. > This caused Freeswitch to refuse the phones registration request. > This in turn caused the phones to re-register. > Problem was, with 10 phones, 6 lines each, perpetually registering on a > 100Mbps LAN, well, you can imagine the overhead. > This created severe latency with Freeswitch, and manifested as dropped > calls, one way audio, and the more phones you had, the worse the problem > was. > Workaround for problem was to use a ramdisk (tmpfs) ?for the database - (Big > Thanks to bkw!) The data in the sqlite db doesn't need to survive a reboot? Any benefit if it does? Also, I've read that the ramdisk isn't that different than what the kernel already does to keep things in memory (yielding very little gain). Thoughts on this? Gabe From frank at carmickle.com Wed Sep 16 22:42:13 2009 From: frank at carmickle.com (Frank Carmickle) Date: Thu, 17 Sep 2009 01:42:13 -0400 Subject: [Freeswitch-users] Compile error In-Reply-To: <1253162070.3583.14.camel@localhost.localdomain> References: <1253162070.3583.14.camel@localhost.localdomain> Message-ID: <20090917054213.GH30343@base.carmickle.com> On Thu, Sep 17, Luis M. Zuccolo wrote: > Hi: > > Since svn version 13523 to current I get this error: > > make[5]: swig: Command not found You must install swig. If your on debian apt-get install swig. If your not see http://www.swig.org/ HTH --FC From kokoska.rokoska at post.cz Thu Sep 17 00:51:57 2009 From: kokoska.rokoska at post.cz (kokoska rokoska) Date: Thu, 17 Sep 2009 09:51:57 +0200 Subject: [Freeswitch-users] DB table sip_dialogs always empty Message-ID: <4AB1EA9D.9030804@post.cz> Hello, I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and based on logs FS connected to the dsns correctly. But during the calls I can't see any rows in sip_dialogs table. When I run "show channels" from console it works fine. All other tables are "populated and maintained" as I expect :-) I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB... Could you, please, point what I'm missing, or where I'm wrong? Thank you very much! Best regards, kokoska.rokoska From t.mahe at telemaque.fr Thu Sep 17 01:21:56 2009 From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=) Date: Thu, 17 Sep 2009 10:21:56 +0200 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <87f2f3b90909160756i7af013dbie657ca02cd0dec6a@mail.gmail.com> References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <4AB0A788.1000105@telemaque.fr> <87f2f3b90909160756i7af013dbie657ca02cd0dec6a@mail.gmail.com> Message-ID: <4AB1F1A4.3040409@telemaque.fr> Hi Michael, I'm gled on IRC, always connected so ping me if you wanna talk ;) Michael Collins a ?crit : > > > On Wed, Sep 16, 2009 at 1:53 AM, Tristan Mah? > wrote: > > Hi, > > Count on me for answering questions on IRC when I'm in, and for > subprojects I'm in too as you know ;) > > Merci! > > Okay, what's your IRC nick and when are you generally on line? Also, > I'm pretty sure that you're fluent in French which is good because we > need more multilingual people out there. Last question: what are your > areas of expertise? I'd like to keep a list of people and what they're > good at so we know whom to ask first when questions come up. > > Thanks again! > -MC > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/9dff416d/attachment.html From yehavi.bourvine at gmail.com Thu Sep 17 01:56:03 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 17 Sep 2009 11:56:03 +0300 Subject: [Freeswitch-users] switch_core_io.c:118 sofia/internal/XXXXX has no read codec. In-Reply-To: References: <20090908071240.GA12470@jdc.jasonjgw.net> <4C8A2EB0-7E3C-41C5-A085-743B82AE706A@freeswitch.org> Message-ID: I've solved the problem: I am running it on a Fedora-10 system. Once I've installed a vanilla kernel (from kernel.org) the problem went away. BTW, can someone shed the light on the kernel's bug which I see mentions of it in this list? Thanks! __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/1c1cb605/attachment.html From gmaruzz at celliax.org Thu Sep 17 02:11:33 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 17 Sep 2009 11:11:33 +0200 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <4AB1F1A4.3040409@telemaque.fr> References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <4AB0A788.1000105@telemaque.fr> <87f2f3b90909160756i7af013dbie657ca02cd0dec6a@mail.gmail.com> <4AB1F1A4.3040409@telemaque.fr> Message-ID: <7b197bef0909170211q6b8c1e11v1ec737c8b90ced00@mail.gmail.com> I'm gmaruzz on IRC, for GSM, Skype, Italian language, audio stuff, etc... -giovanni On Thu, Sep 17, 2009 at 10:21 AM, Tristan Mah? wrote: > Hi Michael, > > I'm gled on IRC, always connected so ping me if you wanna talk ;) > > Michael Collins a ?crit?: > > On Wed, Sep 16, 2009 at 1:53 AM, Tristan Mah? wrote: >> >> Hi, >> >> Count on me for answering questions on IRC when I'm in, and for >> subprojects I'm in too as you know ;) > > Merci! > > Okay, what's your IRC nick and when are you generally on line? Also, I'm > pretty sure that you're fluent in French which is good because we need more > multilingual people out there. Last question: what are your areas of > expertise? I'd like to keep a list of people and what they're good at so we > know whom to ask first when questions come up. > > Thanks again! > -MC > > ________________________________ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From tzury.by at reguluslabs.com Thu Sep 17 02:19:33 2009 From: tzury.by at reguluslabs.com (Tzury Bar Yochay) Date: Thu, 17 Sep 2009 12:19:33 +0300 Subject: [Freeswitch-users] How can I configure TO and FROM in the invite message In-Reply-To: References: <10128ef10909160658h2198c114o81c3c59f9f37c993@mail.gmail.com> Message-ID: <10128ef10909170219n5990e86asd548e1c8f00fb25c@mail.gmail.com> in which file under which section should I specify this From sias at cpdata.co.za Thu Sep 17 03:23:50 2009 From: sias at cpdata.co.za (Sias Mey) Date: Thu, 17 Sep 2009 12:23:50 +0200 Subject: [Freeswitch-users] Delay when transferring call Message-ID: <20090917102350.GA15903@sias-laptop.cpdata.co.za> Hi, Im having a strange issue with a api triggered call transfer. There seems to be a long delay between when the transfer is triggered and when it actually happens. 2009-09-17 11:36:26.995001 [NOTICE] switch_ivr.c:1350 Transfer sofia/internal/1004 at 192.168.0.10 to xml [incust-camp=lucidlive-call=78-conf=41 at default] Error in my_thread_global_end(): 26 threads didn't exit 2009-09-17 11:36:31.997191 [INFO] mod_dialplan_xml.c:315 Processing 1004->incust-camp=lucidlive-call=7 8-conf=41 in context default 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Execution start 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Connecting to Ringback to add call 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Finished adding calls 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Connecting to database lucidlive to update call 2009-09-17 11:36:32.37433 [INFO] regin.js:1 Finished updateing call I though it was my regin.js script causing the delay since it runs a couple of database queries and other things, but the output above show that runs fine. My question is about the delay between 11:36:26 -> 11:36:31. The call is being transfered out of a fifo, but for those 5 seconds there is no MOH or anything else. Just silence. The transfer is triggered via a xml rpc call. But since the delay is between the switch_ivr and mod_dialplan_xml somewhere I doubt that that has much to do with it. Any clues or other places I can go look?. Cheers, Sias From kokoska.rokoska at post.cz Thu Sep 17 04:08:49 2009 From: kokoska.rokoska at post.cz (kokoska.rokoska) Date: Thu, 17 Sep 2009 13:08:49 +0200 Subject: [Freeswitch-users] DB table sip_dialogs is always empty Message-ID: <4AB218C1.1010002@post.cz> I'm sorry to resend this post, but even after few hours I can't see it in the mailing-list... Thanks. Best regards, kokoska.rokoska kokoska rokoska napsal(a): > Hello, > > I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and based > on logs FS connected to the dsns correctly. But during the calls I can't > see any rows in sip_dialogs table. When I run "show channels" from > console it works fine. > All other tables are "populated and maintained" as I expect :-) > > I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB... > > Could you, please, point what I'm missing, or where I'm wrong? Thank you > very much! > > > Best regards, > > kokoska.rokoska > From ahmedmunir007 at gmail.com Thu Sep 17 04:25:55 2009 From: ahmedmunir007 at gmail.com (Ahmed Munir) Date: Thu, 17 Sep 2009 17:25:55 +0600 Subject: [Freeswitch-users] How to process s extension in FS Message-ID: Hi, How can I process s extension in FS? Is there other way around of doing it? Kindly advice me. -- Regards, Ahmed Munir -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/19804fb2/attachment.html From aep.lists at it46.se Thu Sep 17 05:18:01 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Thu, 17 Sep 2009 14:18:01 +0200 Subject: [Freeswitch-users] FreeSWITCH and OpenBTS integration Message-ID: I am happy to let you know that FreeSWITCH route calls from OpenBTS, the open base station based on the Universal Software Radio USRP. Yes! Calls from a standard handset to a GSM base station connected to FreeSWITCH If you want to read more about the idea check: http://openbts.sourceforge.net/ http://www.it46.se/entry/380 (our effort to deploy the technology in a developing region) I have put a few notes for others to give it a try available here: https://sourceforge.net/apps/trac/openbts/wiki/OpenBTS/SettingUpFreeSWITCH Let me know what is the best place in FreeSWITCH wiki to add and keep updated this information -- Stopping junk mailers is good for the environment From aep.lists at it46.se Thu Sep 17 05:46:03 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Thu, 17 Sep 2009 14:46:03 +0200 Subject: [Freeswitch-users] FreeSWITCH and OpenBTS integration In-Reply-To: References: Message-ID: <599dca04711d6c3180b63f36e220dd86.squirrel@correo.nodo50.org> Sorry, just realized that the sourceforge page is protected by password. I am happy to put the info in FreeSWITCH wiki, where does it make sense to add this project info? -aep -- Stopping junk mailers is good for the environment > I am happy to let you know that FreeSWITCH route calls from OpenBTS, the > open base station based on the Universal Software Radio USRP. Yes! Calls > from a standard handset to a GSM base station connected to FreeSWITCH > > If you want to read more about the idea check: > http://openbts.sourceforge.net/ > http://www.it46.se/entry/380 (our effort to deploy the technology in a > developing region) > > I have put a few notes for others to give it a try available here: > https://sourceforge.net/apps/trac/openbts/wiki/OpenBTS/SettingUpFreeSWITCH > > Let me know what is the best place in FreeSWITCH wiki to add and keep > updated this information > > -- > Stopping junk mailers is good for the environment > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mcampbellsmith at gmail.com Thu Sep 17 05:58:38 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Thu, 17 Sep 2009 22:58:38 +1000 Subject: [Freeswitch-users] Simple call waiting question Message-ID: <33c87fa30909170558v4b032da2w6f8202b187d3bcf6@mail.gmail.com> HI All, I am trying to create a simple call waiting dialplan as my phone does not have Recall button. The simple scenario is: 1. B calls user A and is answered 2. C calls user A 3. A puts B on hold 4. A answers C 4. A then recalls first call from B I was going to use fifo for step 3. I am doing: and then My question is then, how do I get A to answer the call from C? Step 4 and then step 5? On another note, if I put I get the following in the console: Transfer sofia/internal_nat/sip:1000 at xx.xx.xx.xx:5060 to 5900[-aleg at XML] Shouldn't that be XML[5900 at default] ? Thanks! From brian at freeswitch.org Thu Sep 17 06:11:57 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Sep 2009 08:11:57 -0500 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: <20090917040742.GA24786@jdc.jasonjgw.net> References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> <8A58CD3E-F0C3-4BAD-8DB9-3C9D3A22A70F@ken-ton.com> <191c3a030909140617v396063e7n86cd8fdb39d75fa9@mail.gmail.com> <0E10F348-B983-48F9-87DB-19BE52CEFD50@ken-ton.com> <20090917040742.GA24786@jdc.jasonjgw.net> Message-ID: <9AC967E2-2807-4FB7-BA8E-F5C128BCDE76@freeswitch.org> Its a bug in 2.6.26 thru 2.6.28 kernels that impact the performance of SQLite. He was specifically running SUSE. /b On Sep 16, 2009, at 11:07 PM, Jason White wrote: > Please take this up with your Linux distribution as a bug report > related to > the kernel, and persist with it until it's sorted out. > > The more that users do this, the more kernel bugs will get fixed. > > We're all responsible to some extent for the quality of our free/ > open-source > operating systems. From brian at freeswitch.org Thu Sep 17 06:12:36 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Sep 2009 08:12:36 -0500 Subject: [Freeswitch-users] Compile error In-Reply-To: <20090917054213.GH30343@base.carmickle.com> References: <1253162070.3583.14.camel@localhost.localdomain> <20090917054213.GH30343@base.carmickle.com> Message-ID: <19FE43B3-7C34-49C5-8E10-DA66B66197BA@freeswitch.org> NO you must not. The issue has been fixed in svn already please start with a fresh tree. /b PS: end users should NEVER have to reswig. On Sep 17, 2009, at 12:42 AM, Frank Carmickle wrote: > On Thu, Sep 17, Luis M. Zuccolo wrote: >> Hi: >> >> Since svn version 13523 to current I get this error: >> >> make[5]: swig: Command not found > > You must install swig. If your on debian apt-get install swig. If > your not see http://www.swig.org/ > > HTH > --FC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/060e8297/attachment.html From brian at freeswitch.org Thu Sep 17 06:13:33 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Sep 2009 08:13:33 -0500 Subject: [Freeswitch-users] DB table sip_dialogs always empty In-Reply-To: <4AB1EA9D.9030804@post.cz> References: <4AB1EA9D.9030804@post.cz> Message-ID: <9751AA65-D077-4E13-AA14-B20EDDE1B2B8@freeswitch.org> These are not sip_dialogs as you think they are. These are used for dialing-info dialogs for sip subscriptions. /b On Sep 17, 2009, at 2:51 AM, kokoska rokoska wrote: > > Hello, > > I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and > based > on logs FS connected to the dsns correctly. But during the calls I > can't > see any rows in sip_dialogs table. When I run "show channels" from > console it works fine. > All other tables are "populated and maintained" as I expect :-) > > I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB... > > Could you, please, point what I'm missing, or where I'm wrong? Thank > you > very much! > > > Best regards, > > kokoska.rokoska From brian at freeswitch.org Thu Sep 17 06:15:06 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Sep 2009 08:15:06 -0500 Subject: [Freeswitch-users] How can I configure TO and FROM in the invite message In-Reply-To: <10128ef10909170219n5990e86asd548e1c8f00fb25c@mail.gmail.com> References: <10128ef10909160658h2198c114o81c3c59f9f37c993@mail.gmail.com> <10128ef10909170219n5990e86asd548e1c8f00fb25c@mail.gmail.com> Message-ID: You can do that in your dialplan. Not sure it'll fix all your options unless you do both the gateway and the sip_invite_domain. /b On Sep 17, 2009, at 4:19 AM, Tzury Bar Yochay wrote: > in which file under which section should I specify this > > data="{sip_invite_domain=${sip_from_host}}sofia/gateway/ > gw1/$1 at domain.org"/> From brian at freeswitch.org Thu Sep 17 06:19:00 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Sep 2009 08:19:00 -0500 Subject: [Freeswitch-users] How to process s extension in FS In-Reply-To: References: Message-ID: Since we have no 's' extension or anything similar maybe if you tell me what you're trying to do I can tell you how to do it.a /b On Sep 17, 2009, at 6:25 AM, Ahmed Munir wrote: > > Hi, > > How can I process s extension in FS? Is there other way around of > doing it? Kindly advice me. > > -- > Regards, > > Ahmed Munir From brian at freeswitch.org Thu Sep 17 06:21:01 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Sep 2009 08:21:01 -0500 Subject: [Freeswitch-users] Simple call waiting question In-Reply-To: <33c87fa30909170558v4b032da2w6f8202b187d3bcf6@mail.gmail.com> References: <33c87fa30909170558v4b032da2w6f8202b187d3bcf6@mail.gmail.com> Message-ID: Personally I would throw the phone in the trash. :P In the default dialplan look at 5900 for park and 5901 for unpark. /b On Sep 17, 2009, at 7:58 AM, Mark Campbell-Smith wrote: > I am trying to create a simple call waiting dialplan as my phone does > not have Recall button. From brian at freeswitch.org Thu Sep 17 06:21:48 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Sep 2009 08:21:48 -0500 Subject: [Freeswitch-users] session record does not for very short calls In-Reply-To: <7d0bfd8c0909161705y5e870521g8a6181d9d7715904@mail.gmail.com> References: <3CF5F946-4307-451E-8E6E-CBC597B5ABFD@gmail.com> <7d0bfd8c0909161705y5e870521g8a6181d9d7715904@mail.gmail.com> Message-ID: I think if you remove it'll do what you want. /b On Sep 16, 2009, at 7:05 PM, Nandy Dagondon wrote: > this makes sense. a workaround would be to provide an optional > variable to delete recording file if it's less than N seconds. > otherwise, it defaults to a preset duration. > > /nandy > > > On Thu, Sep 17, 2009 at 7:46 AM, Seven Du wrote: > I think the file was there but deleted by FreeSWITCH if it thinks it > was too short (like 3 seconds?). If I'm not wrong, someone requested > this feature becuase FreeSWITCH left too many small recordings. > > > On Sep 17, 2009, at 1:27 AM, Jo?o Mesquita wrote: > > I think you need to upgrade your version before we even take a look > > at that... You are so far behind trunk right now and lots of things > > have been changed since then. > > > > Not sure if this would solve your problem but not a lot of ppl will > > look at your problem when you run this version. > > > > jmesquita > > > > On Wed, Sep 16, 2009 at 2:00 PM, Frank @ Impact > > wrote: > > FreeSWITCH Version 1.0.trunk (12790M) > > > > > > I have this in my DP > > > > > > > > > > > > > > > > > > works fine as long as the call is long enough. But if the call is > > only, say, 3-4 seconds long (or something very short like that), > > then the wav file is never created with the audio in it. > > > > > > Is there a work around for this? > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Thu Sep 17 06:19:32 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Sep 2009 08:19:32 -0500 Subject: [Freeswitch-users] FreeSWITCH and OpenBTS integration In-Reply-To: <599dca04711d6c3180b63f36e220dd86.squirrel@correo.nodo50.org> References: <599dca04711d6c3180b63f36e220dd86.squirrel@correo.nodo50.org> Message-ID: Just create an OpenBTS page on our wiki. /b On Sep 17, 2009, at 7:46 AM, Alberto Escudero wrote: > Sorry, just realized that the sourceforge page is protected by > password. I > am happy to put the info in FreeSWITCH wiki, where does it make > sense to > add this project info? From email.list.subscriber at gmail.com Thu Sep 17 06:33:20 2009 From: email.list.subscriber at gmail.com (email lists) Date: Thu, 17 Sep 2009 09:33:20 -0400 Subject: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error) In-Reply-To: <4EF4BF1E8F43894386584BE36354494A13E94D77@ZANEMS01.cc-ntd1.covad.com> References: <4EF4BF1E8F43894386584BE36354494A13E94D77@ZANEMS01.cc-ntd1.covad.com> Message-ID: <4ab23a57.151bf30a.1393.1f39@mx.google.com> Anthony, I increased the ulimit parameters as suggested. I then ran into a few other Freeswitch errors that required me to edit switch.conf.xml to increase the max sessions, and max sessions per second parameters (to 100000, and 100, respectively). Now, after a few minutes of running calls at 20 calls per second with a 3 min. duration, I start seeing an influx of "480 Temporarily Unavailable" SIP error messages. Here is a snip of the log: ------------------------------------------- 2009-09-16 19:35:19.18798 [DEBUG] switch_core_session.c:932 Send signal sofia/external/2015692700 at 72.245.84.150 [BREAK] 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/9726074300 at 72.245.84.24:5060) State HANGUP 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:398 (sofia/external/2015692700 at 72.245.84.150) Running State Change CS_REPORTING 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:338 Channel sofia/internal/9726074300 at 72.245.84.24:5060 hanging up, cause: NORMAL_CLEARING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:612 (sofia/external/2015692700 at 72.245.84.150) State REPORTING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:53 sofia/external/2015692700 at 72.245.84.150 Standard REPORTING, cause: NO_ANSWER 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:417 Responding to INVITE with: 480 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:612 (sofia/external/2015692700 at 72.245.84.150) State REPORTING going to sleep 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:411 (sofia/external/2015692700 at 72.245.84.150) State Change CS_REPORTING -> CS_DESTROY 2009-09-16 19:35:19.18798 [DEBUG] switch_core_session.c:1068 Session 3395 (sofia/external/2015692700 at 72.245.84.150) Locked, Waiting on external entities 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:46 sofia/internal/9726074300 at 72.245.84.24:5060 Standard HANGUP, cause: NORMAL_CLEARING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/9726074300 at 72.245.84.24:5060) State HANGUP going to sleep 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/external/2015692915 at 72.245.84.150) State HANGUP 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:46 sofia/external/2015692915 at 72.245.84.150 Standard HANGUP, cause: NO_ANSWER 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/external/2015692915 at 72.245.84.150) State HANGUP going to sleep 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:476 (sofia/internal/9726074300 at 72.245.84.24:5060) State Change CS_HANGUP -> CS_REPORTING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/9726074300 at 72.245.84.24:5060 [BREAK] 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/9726074300 at 72.245.84.24:5060) Running State Change CS_REPORTING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/9726074300 at 72.245.84.24:5060) State REPORTING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:53 sofia/internal/9726074300 at 72.245.84.24:5060 Standard REPORTING, cause: NORMAL_CLEARING 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:612 (sofia/internal/9726074300 at 72.245.84.24:5060) State REPORTING going to sleep 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:411 (sofia/internal/9726074300 at 72.245.84.24:5060) State Change CS_REPORTING -> CS_DESTROY 2009-09-16 19:35:19.18798 [DEBUG] switch_core_session.c:1068 Session 3392 (sofia/internal/9726074300 at 72.245.84.24:5060) Locked, Waiting on external entities 2009-09-16 19:35:19.18798 [NOTICE] switch_core_session.c:1086 Session 3392 (sofia/internal/9726074300 at 72.245.84.24:5060) Ended 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/external/2015692915 at 72.245.84.150) State HANGUP 2009-09-16 19:35:19.18798 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/9726074300 at 72.245.84.24:5060 [CS_DESTROY] 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:338 Channel sofia/external/2015692915 at 72.245.84.150 hanging up, cause: NO_ANSWER 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:406 Sending CANCEL to sofia/external/2015692915 at 72.245.84.150 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:46 sofia/external/2015692915 at 72.245.84.150 Standard HANGUP, cause: NO_ANSWER 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/external/2015692915 at 72.245.84.150) State HANGUP going to sleep 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/9726074300 at 72.245.84.24:5060) State DESTROY 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:255 sofia/internal/9726074300 at 72.245.84.24:5060 SOFIA DESTROY 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:60 sofia/internal/9726074300 at 72.245.84.24:5060 Standard DESTROY 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:564 (sofia/internal/9726074300 at 72.245.84.24:5060) State DESTROY going to sleep 2009-09-16 19:35:19.18798 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/2126822269 at 72.245.84.24:5060) State HANGUP 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:338 Channel sofia/internal/2126822269 at 72.245.84.24:5060 hanging up, cause: NORMAL_CLEARING 2009-09-16 19:35:19.18798 [DEBUG] mod_sofia.c:417 Responding to INVITE with: 480 ------------------------------------------- Is there some other limitation on freeswitch I may be hitting? I tried throttling down to 10 calls per second (3 min. duration) and here are some snippets from tailing the log file: ------------------------------------------- 2009-09-16 20:01:47.422656 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE] 2009-09-16 20:01:47.496066 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE] 2009-09-16 20:01:47.623316 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE] 2009-09-16 20:02:59.777664 [CRIT] mod_sofia.c:2601 Error Creating Session 2009-09-16 20:02:59.777664 [CRIT] mod_sofia.c:2601 Error Creating Session 2009-09-16 20:02:59.777664 [CRIT] mod_sofia.c:2601 Error Creating Session 2009-09-16 20:02:59.788134 [CRIT] mod_sofia.c:2601 Error Creating Session nta: sent 503 Maximum Calls In Progress for INVITE (1) nta: sent 503 Maximum Calls In Progress for INVITE (1) nta: sent 503 Maximum Calls In Progress for INVITE (1) nta: sent 503 Maximum Calls In Progress for INVITE (1) nta: sent 502 Bad Gateway for INVITE (1) nta: sent 502 Bad Gateway for INVITE (1) nta: sent 502 Bad Gateway for INVITE (1) nta: sent 503 Maximum Calls In Progress for INVITE (1) 2009-09-16 20:02:59.788134 [CRIT] mod_sofia.c:2601 Error Creating Session 2009-09-16 20:02:59.788134 [CRIT] mod_sofia.c:2601 Error Creating Session 2009-09-16 20:02:59.788134 [CRIT] mod_sofia.c:2601 Error Creating Session nta: sent 502 Bad Gateway for INVITE (1) nta: sent 502 Bad Gateway for INVITE (1) nta: sent 502 Bad Gateway for INVITE (1) nta: sent 502 Bad Gateway for INVITE (1) 2009-09-16 20:02:59.861110 [CRIT] switch_time.c:500 Over Session Rate of 100! nta: sent 480 Temporarily Unavailable for INVITE (1) nta: sent 480 Temporarily Unavailable for INVITE (1) nta: sent 480 Temporarily Unavailable for INVITE (1) nta: sent 480 Temporarily Unavailable for INVITE (1) nta: sent 480 Temporarily Unavailable for INVITE (1) nta: sent 480 Temporarily Unavailable for INVITE (1) ------------------------------------------- I'm going to try increasing the "session rate", but I'm wondering if there are other limitations I'm running into. Any thoughts/suggestions? Vladimir ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale [anthony.minessale at gmail.com] Sent: Tuesday, September 15, 2009 6:32 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] 502 Bad Gateway: Destination out of order error It's probably from this 2009-09-15 14:34:55.838365 [ERR] sofia_glue.c:2503 AUDIO RTP REPORTS ERROR: [Socket Error!] 2009-09-15 14:34:55.838365 [NOTICE] sofia_glue.c:2504 Hangup sofia/external/411 at 192.168.0.150 [CS_CONSUME_MEDIA][DESTINATION_OUT_OF_ORDER ] 2009-09-15 14:34:55.838365 [ERR] sofia.c:3796 RTP Error! your machine failed to produce a socket when requested. My blind guess, you are on a 32 bit machine and you do not have the ulimits set for enough file descriptors etc.. ulimit -c unlimited ulimit -d unlimited ulimit -f unlimited ulimit -i unlimited ulimit -n 999999 ulimit -q unlimited ulimit -u unlimited ulimit -v unlimited ulimit -x unlimited ulimit -s 244 ulimit -l unlimited On Tue, Sep 15, 2009 at 5:22 PM, email lists > wrote: Hello All, Wondering if anyone has experienced this issue before. I've attached a snip of the log file where the error occurs and could use some leads on this. What's interesting is that the call appears to complete as normal, and a radius stop message even gets generated, though the duration is ~1 second. h323-disconnect-time = "h323-disconnect-time=14:35:01.000 UTC Tue Sep 15 2009" h323-connect-time = "h323-connect-time=14:34:59.000 UTC Tue Sep 15 2009" While there are a lot of pieces involved, the call scenario is pretty basic (no transfers, no holds, etc.), just a few redirects that Freeswitch appears to be able to handle without issue. Attached is a dumb'd down call ladder. I tried different rates at which I generate the calls, but it didn't seem to correlate to the amount of errors I am seeing. Sending a total of 100 calls, with a call duration of 10 seconds: @10 calls per second = 14 "502" errors. @5 calls per second = 4 "502" errors. @4 calls per second = NO ERRORS (1st run) @4 calls per second = 39 "502" errors. Please let me know if any additional information is needed. Thanks in advance for all help. Vladimir _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 From brian at freeswitch.org Thu Sep 17 07:07:26 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Sep 2009 09:07:26 -0500 Subject: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error) In-Reply-To: <4ab23a57.151bf30a.1393.1f39@mx.google.com> References: <4EF4BF1E8F43894386584BE36354494A13E94D77@ZANEMS01.cc-ntd1.covad.com> <4ab23a57.151bf30a.1393.1f39@mx.google.com> Message-ID: <9859212C-CDCC-4F49-9055-5B309D79D331@freeswitch.org> Are you using SIPP? /b On Sep 17, 2009, at 8:33 AM, email lists wrote: > Now, after a few minutes of running calls at 20 calls per second with > a 3 min. duration, I start seeing an influx of "480 Temporarily > Unavailable" SIP error messages. From irmatov at gmail.com Thu Sep 17 07:08:09 2009 From: irmatov at gmail.com (Timur Irmatov) Date: Thu, 17 Sep 2009 19:08:09 +0500 Subject: [Freeswitch-users] Limit_Hash In-Reply-To: <4AAF760D.3030105@venturevoip.com> References: <4AAF760D.3030105@venturevoip.com> Message-ID: <241d382f0909170708i6d60613ar1589e0974671c7@mail.gmail.com> Hi, On Tue, Sep 15, 2009 at 4:10 PM, Matt Riddell wrote: > My question then becomes, how do I set a hash for an originated call? You set a limit by executing limit_hash application at appropriate place in your dial plan. > It seems that limit_hash is an application rather than a channel > variable, and so far I've been doing most things without touching the > dialplan. Correct, limit_hash is an application. > So, say I want to originate 9 calls, 3 from 3 customers. > > I would like to mark the calls with my_customer_group_1 through 3, and > then use the limit_hash_usage command to verify the count of channels in > each group. > > I therefore have a few questions: > > 1. Can I mark a call in the originate statement? Somehow you need an ability to distinguish one customer from another. For example, if each customer has its own number and originates calls only from this number, you can use this number for limiting their usage. > 2. How do I use the limit_hash_usage command? > The wiki states: > You can verify the usage of any resource with the limit_hash_usage api call. > limit_hash_usage > Is realm the same as a SIP realm? > Is id the hash that I have used to mark the call with? No, is just an arbitrary string to allow you have same s in different realms. For example, realm can be a name of your customer, and can be a direction of call, so you can set a limit of 5 outbound and 3 inbound calls per customer: .... .... .... Of course, those two lines should be placed to appropriate (and most probably separate) places of your dialplan. So, back to your example. Three customers, 3 outbound calls per each. Let's suppose customer 1, 2 and 3 have numbers 1001, 1002, 1003 assigned to them, and customer1 originates calls from 1001, and so on.. You place the limit like this: You should see messages in console about limits during live calls. Also, bear in mind that if you place a limit in one dialplan context and then transfer your call to another, limit will be reset. Does anyone know, why it is like this? Is it possible to allow limits persist across dialplan contexts? -- Timur Irmatov, xmpp:irmatov at jabber.ru From steveu at coppice.org Thu Sep 17 07:23:28 2009 From: steveu at coppice.org (Steve Underwood) Date: Thu, 17 Sep 2009 22:23:28 +0800 Subject: [Freeswitch-users] FreeSWITCH and OpenBTS integration In-Reply-To: References: <599dca04711d6c3180b63f36e220dd86.squirrel@correo.nodo50.org> Message-ID: <4AB24660.6080006@coppice.org> On 09/17/2009 09:19 PM, Brian West wrote: > Just create an OpenBTS page on our wiki. > > /b > > On Sep 17, 2009, at 7:46 AM, Alberto Escudero wrote: > > >> Sorry, just realized that the sourceforge page is protected by >> password. I >> am happy to put the info in FreeSWITCH wiki, where does it make >> sense to >> add this project info? >> > Isn't there still some legal wrangling over openBTS? Steve From email.list.subscriber at gmail.com Thu Sep 17 07:31:30 2009 From: email.list.subscriber at gmail.com (email lists) Date: Thu, 17 Sep 2009 10:31:30 -0400 Subject: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error) In-Reply-To: <9859212C-CDCC-4F49-9055-5B309D79D331@freeswitch.org> References: <4EF4BF1E8F43894386584BE36354494A13E94D77@ZANEMS01.cc-ntd1.covad.com> <4ab23a57.151bf30a.1393.1f39@mx.google.com> <9859212C-CDCC-4F49-9055-5B309D79D331@freeswitch.org> Message-ID: <4ab247f8.171bf30a.41b9.1c84@mx.google.com> Yes. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, September 17, 2009 10:07 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error) Are you using SIPP? /b On Sep 17, 2009, at 8:33 AM, email lists wrote: > Now, after a few minutes of running calls at 20 calls per second with > a 3 min. duration, I start seeing an influx of "480 Temporarily > Unavailable" SIP error messages. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org From anthony.minessale at gmail.com Thu Sep 17 07:44:24 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Sep 2009 09:44:24 -0500 Subject: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error) In-Reply-To: <4ab247f8.171bf30a.41b9.1c84@mx.google.com> References: <4EF4BF1E8F43894386584BE36354494A13E94D77@ZANEMS01.cc-ntd1.covad.com> <4ab23a57.151bf30a.1393.1f39@mx.google.com> <9859212C-CDCC-4F49-9055-5B309D79D331@freeswitch.org> <4ab247f8.171bf30a.41b9.1c84@mx.google.com> Message-ID: <191c3a030909170744t6bb456a5wbd53fe61a6efbe04@mail.gmail.com> You should get a more powerful machine if you intend to sustain upwards of 3600 calls at 20cps We recommend a 64 bit multi-core machine running 64bit Centos5.3 On Thu, Sep 17, 2009 at 9:31 AM, email lists < email.list.subscriber at gmail.com> wrote: > Yes. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian West > Sent: Thursday, September 17, 2009 10:07 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max > Sessions (was 502 Bad Gateway: Destination out of order error) > > Are you using SIPP? > > /b > > On Sep 17, 2009, at 8:33 AM, email lists wrote: > > > Now, after a few minutes of running calls at 20 calls per second > with > > a 3 min. duration, I start seeing an influx of "480 Temporarily > > Unavailable" SIP error messages. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/52bf2d80/attachment.html From stkn at freeswitch.org Thu Sep 17 07:52:28 2009 From: stkn at freeswitch.org (Stefan Knoblich) Date: Thu, 17 Sep 2009 16:52:28 +0200 Subject: [Freeswitch-users] FreeSWITCH and OpenBTS integration In-Reply-To: <4AB24660.6080006@coppice.org> References: <4AB24660.6080006@coppice.org> Message-ID: <200909171652.29203.stkn@freeswitch.org> Am Thursday 17 September 2009 schrieb Steve Underwood: > On 09/17/2009 09:19 PM, Brian West wrote: > > Just create an OpenBTS page on our wiki. > > > > /b > > > > On Sep 17, 2009, at 7:46 AM, Alberto Escudero wrote: > > > > > >> Sorry, just realized that the sourceforge page is protected by > >> password. I > >> am happy to put the info in FreeSWITCH wiki, where does it make > >> sense to > >> add this project info? > >> > > > Isn't there still some legal wrangling over openBTS? > > Steve nope, that was resolved: http://openbts.blogspot.com/2009/07/three-quotes.html -- ------------------------------------------------------------------------------- Stefan Knoblich | Web: http://www.axsentis.de/ axsentis GmbH | http://oss.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de UST-ID: DE244977565 | JID: s.knoblich at jabber.axsentis.de ------------------------------------------------------------------------------- Web: http://stkn.techmage.de/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net From aep.lists at it46.se Thu Sep 17 08:31:33 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Thu, 17 Sep 2009 17:31:33 +0200 Subject: [Freeswitch-users] FreeSWITCH and OpenBTS integration In-Reply-To: <200909171652.29203.stkn@freeswitch.org> References: <4AB24660.6080006@coppice.org> <200909171652.29203.stkn@freeswitch.org> Message-ID: The real challenge at the moment is to find adequate regulatory scenarios to run this technology. In many parts of the world where we works operators/governments have literally locked the spectrum so others can not run anything on it. Community networks are left with garbage bands to operate 802.11 devices. We welcome any scenarios where to operate an open gsm infrastructure connected to IP. /aep -- Stopping junk mailers is good for the environment > Am Thursday 17 September 2009 schrieb Steve Underwood: >> On 09/17/2009 09:19 PM, Brian West wrote: >> > Just create an OpenBTS page on our wiki. >> > >> > /b >> > >> > On Sep 17, 2009, at 7:46 AM, Alberto Escudero wrote: >> > >> > >> >> Sorry, just realized that the sourceforge page is protected by >> >> password. I >> >> am happy to put the info in FreeSWITCH wiki, where does it make >> >> sense to >> >> add this project info? >> >> >> > >> Isn't there still some legal wrangling over openBTS? >> >> Steve > > > nope, that was resolved: > > http://openbts.blogspot.com/2009/07/three-quotes.html > > > -- > ------------------------------------------------------------------------------- > Stefan Knoblich | Web: http://www.axsentis.de/ > axsentis GmbH | http://oss.axsentis.de/ > Eupener Str. 74, 50933 Koeln, Germany | > Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de > UST-ID: DE244977565 | JID: > s.knoblich at jabber.axsentis.de > ------------------------------------------------------------------------------- > Web: http://stkn.techmage.de/ > Email: stkn at freeswitch.org > IRC: #freeswitch-de @ irc.freenode.net > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From aep.lists at it46.se Thu Sep 17 08:39:51 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Thu, 17 Sep 2009 17:39:51 +0200 Subject: [Freeswitch-users] FreeSWITCH and OpenBTS integration In-Reply-To: References: <599dca04711d6c3180b63f36e220dd86.squirrel@correo.nodo50.org> Message-ID: <69792d30f30205f33eb3585412d966f9.squirrel@correo.nodo50.org> Done! http://wiki.freeswitch.org/wiki/OpenBTS -- Stopping junk mailers is good for the environment > Just create an OpenBTS page on our wiki. > > /b > > On Sep 17, 2009, at 7:46 AM, Alberto Escudero wrote: > >> Sorry, just realized that the sourceforge page is protected by >> password. I >> am happy to put the info in FreeSWITCH wiki, where does it make >> sense to >> add this project info? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From TStutsman at BossProductsInc.com Thu Sep 17 09:01:51 2009 From: TStutsman at BossProductsInc.com (Travis Stutsman) Date: Thu, 17 Sep 2009 12:01:51 -0400 Subject: [Freeswitch-users] faxrx error 13 Unexpected message received In-Reply-To: <4AB179ED.1060701@coppice.org> References: <5991020B1C443E459410B16D7B08F33E030879@dc1.bossproductsinc.com> <4AB11772.4020904@coppice.org><5991020B1C443E459410B16D7B08F33E03087B@dc1.bossproductsinc.com> <4AB179ED.1060701@coppice.org> Message-ID: <5991020B1C443E459410B16D7B08F33E030887@dc1.bossproductsinc.com> Thanks Steve. More tests today. Consistent results. The Fax machine is in the same building as the FS server. It goes out over the PSTN to my carrier, then carrier to me over IP to the server. I need to try more fax machines and more locations. -- Travis From msc at freeswitch.org Thu Sep 17 10:40:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 10:40:21 -0700 Subject: [Freeswitch-users] Delay when transferring call In-Reply-To: <20090917102350.GA15903@sias-laptop.cpdata.co.za> References: <20090917102350.GA15903@sias-laptop.cpdata.co.za> Message-ID: <87f2f3b90909171040n1844d4ccu44d070a785cc0d6a@mail.gmail.com> Turn on debug (press F8) level logging and capture the output, put in pastebin.freeswitch.org. Hopefully the debug output will shed some light on where the delay is occurring. Also, see this page for some tips on how to collect information for debugging purposes: http://wiki.freeswitch.org/wiki/Reporting_Bugs It will give you handy tips on collecting information, posting to pastebin, asking community for help, etc. etc. In short, it will make your life easier. :) -MC On Thu, Sep 17, 2009 at 3:23 AM, Sias Mey wrote: > Hi, > > Im having a strange issue with a api triggered call transfer. > > There seems to be a long delay between when the transfer is triggered and > when it actually happens. > > 2009-09-17 11:36:26.995001 [NOTICE] switch_ivr.c:1350 Transfer > sofia/internal/1004 at 192.168.0.10 to xml > [incust-camp=lucidlive-call=78-conf=41 at default] > Error in my_thread_global_end(): 26 threads didn't exit > 2009-09-17 11:36:31.997191 [INFO] mod_dialplan_xml.c:315 Processing > 1004->incust-camp=lucidlive-call=7 > 8-conf=41 in context default > 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Execution start > 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Connecting to Ringback to add > call > 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Finished adding calls > 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Connecting to database > lucidlive to update call > 2009-09-17 11:36:32.37433 [INFO] regin.js:1 Finished updateing call > > I though it was my regin.js script causing the delay since it runs a couple > of database queries and other things, but the output above show that runs > fine. > > My question is about the delay between 11:36:26 -> 11:36:31. The call is > being transfered out of a fifo, but for those 5 seconds there is no MOH or > anything else. Just silence. > > The transfer is triggered via a xml rpc call. But since the delay is > between the switch_ivr and mod_dialplan_xml somewhere I doubt that that has > much to do with it. > > Any clues or other places I can go look?. > > Cheers, > Sias > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/fee5ee6e/attachment-0001.html From anthony.minessale at gmail.com Thu Sep 17 10:45:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Sep 2009 12:45:04 -0500 Subject: [Freeswitch-users] DB table sip_dialogs is always empty In-Reply-To: <4AB218C1.1010002@post.cz> References: <4AB218C1.1010002@post.cz> Message-ID: <191c3a030909171045sc7c999cl11d991642058308a@mail.gmail.com> that table is specific to the manage-presence option On Thu, Sep 17, 2009 at 6:08 AM, kokoska.rokoska wrote: > I'm sorry to resend this post, but even after few hours I can't see it > in the mailing-list... > Thanks. > > > Best regards, > > kokoska.rokoska > > > kokoska rokoska napsal(a): > > Hello, > > > > I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and based > > on logs FS connected to the dsns correctly. But during the calls I can't > > see any rows in sip_dialogs table. When I run "show channels" from > > console it works fine. > > All other tables are "populated and maintained" as I expect :-) > > > > I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB... > > > > Could you, please, point what I'm missing, or where I'm wrong? Thank you > > very much! > > > > > > Best regards, > > > > kokoska.rokoska > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/bd153d8d/attachment.html From email.list.subscriber at gmail.com Thu Sep 17 10:48:15 2009 From: email.list.subscriber at gmail.com (email lists) Date: Thu, 17 Sep 2009 13:48:15 -0400 Subject: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error) In-Reply-To: <191c3a030909170744t6bb456a5wbd53fe61a6efbe04@mail.gmail.com> References: <4EF4BF1E8F43894386584BE36354494A13E94D77@ZANEMS01.cc-ntd1.covad.com> <4ab23a57.151bf30a.1393.1f39@mx.google.com> <9859212C-CDCC-4F49-9055-5B309D79D331@freeswitch.org> <4ab247f8.171bf30a.41b9.1c84@mx.google.com> <191c3a030909170744t6bb456a5wbd53fe61a6efbe04@mail.gmail.com> Message-ID: <4ab27616.141bf30a.0af3.2571@mx.google.com> I'm running on a (2) Dual Core, 16G RAM machine. Tried 5 calls per second with 3 minute duration -- noticed same issues. I throttled down to 5 calls per second, with a 30 second duration and it seems quite a bit more stable now, though this doesn't meet our requirements. Looking at moving Freeswitch to a Quad-core 2.8ghz server now, and will re-test. Vladimir From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, September 17, 2009 10:44 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error) You should get a more powerful machine if you intend to sustain upwards of 3600 calls at 20cps We recommend a 64 bit multi-core machine running 64bit Centos5.3 On Thu, Sep 17, 2009 at 9:31 AM, email lists wrote: Yes. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, September 17, 2009 10:07 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error) Are you using SIPP? /b On Sep 17, 2009, at 8:33 AM, email lists wrote: > Now, after a few minutes of running calls at 20 calls per second with > a 3 min. duration, I start seeing an influx of "480 Temporarily > Unavailable" SIP error messages. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/19abda9b/attachment.html From msc at freeswitch.org Thu Sep 17 10:50:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 10:50:18 -0700 Subject: [Freeswitch-users] DB table sip_dialogs is always empty In-Reply-To: <4AB218C1.1010002@post.cz> References: <4AB218C1.1010002@post.cz> Message-ID: <87f2f3b90909171050p5bb2748fm2e2d06fa0ecc89e8@mail.gmail.com> Could you pastebin a minimal config that someone could put on their own system in order to try to simulate what you're doing? It might help if someone can try your setup on a different machine to see what happens. -MC On Thu, Sep 17, 2009 at 4:08 AM, kokoska.rokoska wrote: > I'm sorry to resend this post, but even after few hours I can't see it > in the mailing-list... > Thanks. > > > Best regards, > > kokoska.rokoska > > > kokoska rokoska napsal(a): > > Hello, > > > > I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and based > > on logs FS connected to the dsns correctly. But during the calls I can't > > see any rows in sip_dialogs table. When I run "show channels" from > > console it works fine. > > All other tables are "populated and maintained" as I expect :-) > > > > I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB... > > > > Could you, please, point what I'm missing, or where I'm wrong? Thank you > > very much! > > > > > > Best regards, > > > > kokoska.rokoska > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/1761cd71/attachment-0001.html From msc at freeswitch.org Thu Sep 17 10:52:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 10:52:29 -0700 Subject: [Freeswitch-users] FreeSWITCH and OpenBTS integration In-Reply-To: <69792d30f30205f33eb3585412d966f9.squirrel@correo.nodo50.org> References: <599dca04711d6c3180b63f36e220dd86.squirrel@correo.nodo50.org> <69792d30f30205f33eb3585412d966f9.squirrel@correo.nodo50.org> Message-ID: <87f2f3b90909171052i627cc2een7cb38fdad627eb6c@mail.gmail.com> On Thu, Sep 17, 2009 at 8:39 AM, Alberto Escudero wrote: > > Done! > http://wiki.freeswitch.org/wiki/OpenBTS > > AEP, Thanks for the update! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/bc96e3d8/attachment.html From msc at freeswitch.org Thu Sep 17 11:10:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 11:10:08 -0700 Subject: [Freeswitch-users] How to process s extension in FS In-Reply-To: References: Message-ID: <87f2f3b90909171110m46c3a930gbd3affb84c75218@mail.gmail.com> On Thu, Sep 17, 2009 at 4:25 AM, Ahmed Munir wrote: > > Hi, > > How can I process s extension in FS? Is there other way around of doing it? > Kindly advice me. > > Like Brian said, we need a little more information. Where is the call coming from? In some cases you will handle the incoming call in the public context. See conf/dialplan/public.xml and conf/dialplan/public/00_inbound_did.xml for an example of how to handle an incoming DID from a SIP provider. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/c377cfb8/attachment.html From msc at freeswitch.org Thu Sep 17 11:17:10 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 11:17:10 -0700 Subject: [Freeswitch-users] session record does not for very short calls In-Reply-To: References: <3CF5F946-4307-451E-8E6E-CBC597B5ABFD@gmail.com> <7d0bfd8c0909161705y5e870521g8a6181d9d7715904@mail.gmail.com> Message-ID: <87f2f3b90909171117g59f60ad2vb7858875e7d49986@mail.gmail.com> On Thu, Sep 17, 2009 at 6:21 AM, Brian West wrote: > I think if you remove data="RECORD_ANSWER_REQ=true"/> > > it'll do what you want. > > FYI http://wiki.freeswitch.org/wiki/Channel_Variables#RECORD_ANSWER_REQ Hopefully that will help fill in the details. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/211dabbc/attachment.html From msc at freeswitch.org Thu Sep 17 11:20:22 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 11:20:22 -0700 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> Message-ID: <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> On Sun, Sep 13, 2009 at 8:01 AM, mark morreny wrote: > Hello > > Has any tried setting up an ACD based on skillset? The current out-of-box > version of fifo does not seem to support acd based on agent skillset. Does > anyone have any experience in doing it with some external scripting using > lua or javascript? > > I am interested in hearing how others may have done it as I am trying to > implement one myself. > > thx, > > mark > > I was curious about this myself. Even if someone has built a non-free skills-based ACD using FS I'd like to know about it. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/f696ba81/attachment.html From gshfreesw at gmail.com Thu Sep 17 11:40:15 2009 From: gshfreesw at gmail.com (Shameem Shiek) Date: Thu, 17 Sep 2009 14:40:15 -0400 Subject: [Freeswitch-users] Event socket library bug on hangup? Message-ID: <5070fcbd0909171140v6d4755a7k52099acad58c28fc@mail.gmail.com> When a call is hangup, I get SERVER_DISCONNECTED Event over and over again instead of CHANNEL_HANGUP event. Has anyone else experienced this? I am using freeswitch 1.04. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/3a21a294/attachment.html From anthony.minessale at gmail.com Thu Sep 17 11:56:05 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Sep 2009 13:56:05 -0500 Subject: [Freeswitch-users] Event socket library bug on hangup? In-Reply-To: <5070fcbd0909171140v6d4755a7k52099acad58c28fc@mail.gmail.com> References: <5070fcbd0909171140v6d4755a7k52099acad58c28fc@mail.gmail.com> Message-ID: <191c3a030909171156m104b288u97fb8c4869a7f83c@mail.gmail.com> try sending the linger command linger\n\n On Thu, Sep 17, 2009 at 1:40 PM, Shameem Shiek wrote: > When a call is hangup, I get SERVER_DISCONNECTED Event over and over again > instead of CHANNEL_HANGUP event. Has anyone else experienced this? I am > using freeswitch 1.04. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/c2115de2/attachment.html From Christian.Jensen at Teligence.Net Thu Sep 17 11:56:33 2009 From: Christian.Jensen at Teligence.Net (Christian Jensen) Date: Thu, 17 Sep 2009 11:56:33 -0700 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> Message-ID: This would be a fantastic addition - my company is currently looking to Asterisk as a potential candidate for this if FS can't do it. I want FS to win of course :-) Christian Jensen Software Development Manager Back Office ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, September 17, 2009 11:20 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] skill-based ACD On Sun, Sep 13, 2009 at 8:01 AM, mark morreny wrote: Hello Has any tried setting up an ACD based on skillset? The current out-of-box version of fifo does not seem to support acd based on agent skillset. Does anyone have any experience in doing it with some external scripting using lua or javascript? I am interested in hearing how others may have done it as I am trying to implement one myself. thx, mark I was curious about this myself. Even if someone has built a non-free skills-based ACD using FS I'd like to know about it. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/d9fea56b/attachment-0001.html From anthony.minessale at gmail.com Thu Sep 17 11:57:38 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Sep 2009 13:57:38 -0500 Subject: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error) In-Reply-To: <4ab27616.141bf30a.0af3.2571@mx.google.com> References: <4EF4BF1E8F43894386584BE36354494A13E94D77@ZANEMS01.cc-ntd1.covad.com> <4ab23a57.151bf30a.1393.1f39@mx.google.com> <9859212C-CDCC-4F49-9055-5B309D79D331@freeswitch.org> <4ab247f8.171bf30a.41b9.1c84@mx.google.com> <191c3a030909170744t6bb456a5wbd53fe61a6efbe04@mail.gmail.com> <4ab27616.141bf30a.0af3.2571@mx.google.com> Message-ID: <191c3a030909171157x2dc595b6o6508a1f110d74036@mail.gmail.com> You never answered the question if you are using the 32 bit version of the OS If you are, make sure you do not, this has a huge impact on your performance. On Thu, Sep 17, 2009 at 12:48 PM, email lists < email.list.subscriber at gmail.com> wrote: > I'm running on a (2) Dual Core, 16G RAM machine. Tried 5 calls per > second with 3 minute duration -- noticed same issues. > > > > I throttled down to 5 calls per second, with a 30 second duration and it > seems quite a bit more stable now, though this doesn?t meet our > requirements. > > > > Looking at moving Freeswitch to a Quad-core 2.8ghz server now, and will > re-test. > > > > Vladimir > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Thursday, September 17, 2009 10:44 AM > > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max > Sessions (was 502 Bad Gateway: Destination out of order error) > > > > *You should get a more powerful machine if you intend to sustain upwards > of 3600 calls at 20cps* > > We recommend a 64 bit multi-core machine running 64bit Centos5.3 > > On Thu, Sep 17, 2009 at 9:31 AM, email lists < > email.list.subscriber at gmail.com> wrote: > > Yes. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian West > Sent: Thursday, September 17, 2009 10:07 AM > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max > Sessions (was 502 Bad Gateway: Destination out of order error) > > Are you using SIPP? > > /b > > On Sep 17, 2009, at 8:33 AM, email lists wrote: > > > Now, after a few minutes of running calls at 20 calls per second > with > > a 3 min. duration, I start seeing an influx of "480 Temporarily > > Unavailable" SIP error messages. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/379dd7f1/attachment.html From siniypin at gmail.com Thu Sep 17 11:58:22 2009 From: siniypin at gmail.com (RobertT) Date: Thu, 17 Sep 2009 22:58:22 +0400 Subject: [Freeswitch-users] mod_conference performance In-Reply-To: <2160023e0909161056p4338cc19rb982400ebf677c1d@mail.gmail.com> References: <2160023e0909161056p4338cc19rb982400ebf677c1d@mail.gmail.com> Message-ID: <2160023e0909171158i118f2296o8366192b954f7c1e@mail.gmail.com> Okay, I've performed some additional tests and this is what I've found: * codec* *max calls *speex (8kHz) 50 iLBC(8kHz) 50 PCMU(8kHz) 260(approx*) GSM(8kHz) 150(approx*) speex(16kHz) 50 G722(16kHz) 90(approx*) * - couldn't trace till the total load of CPU 'cause RDP was timedout due to channel load. Calculated by trend. Still I think Linux tests are necessary. Cheers, Robert. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/53d0d61d/attachment.html From gshfreesw at gmail.com Thu Sep 17 12:08:52 2009 From: gshfreesw at gmail.com (Shameem Shiek) Date: Thu, 17 Sep 2009 15:08:52 -0400 Subject: [Freeswitch-users] Event socket library bug on hangup? In-Reply-To: <191c3a030909171156m104b288u97fb8c4869a7f83c@mail.gmail.com> References: <5070fcbd0909171140v6d4755a7k52099acad58c28fc@mail.gmail.com> <191c3a030909171156m104b288u97fb8c4869a7f83c@mail.gmail.com> Message-ID: <5070fcbd0909171208g593e79f0i8ea9832608a1d21b@mail.gmail.com> What does "linger" do? I do not see it documented anywhere. Why would I get a SERVER_DISCONNECTED instead of CHANNEL_HANGUP ? By the way the BUG does not happen every time and normally I *DO* see the CHANNEL_HANGUP come through in most cases. Thanks for the help. On Thu, Sep 17, 2009 at 2:56 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > try sending the linger command > > linger\n\n > > > > > On Thu, Sep 17, 2009 at 1:40 PM, Shameem Shiek wrote: > >> When a call is hangup, I get SERVER_DISCONNECTED Event over and over again >> instead of CHANNEL_HANGUP event. Has anyone else experienced this? I am >> using freeswitch 1.04. >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/9a88a2ce/attachment.html From R.Kloosterman at mtel.nl Thu Sep 17 12:13:05 2009 From: R.Kloosterman at mtel.nl (Remko Kloosterman) Date: Thu, 17 Sep 2009 21:13:05 +0200 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com><87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail .com> Message-ID: <11372C8B9E603F4FACDE6AB18256DEC695A9EC@srvmtel.office.mtel.nl> I have been working on several voice projects in the past with ACD features, mostly based on TDM technology. It's all commercial stuff, but I have the experience and I am willing to share that. If anyone wishes to start such a development I'm sure I can dig up a functional model and help with the design. Regards, Remko ________________________________ Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Christian Jensen Verzonden: donderdag 17 september 2009 20:57 Aan: freeswitch-users at lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] skill-based ACD This would be a fantastic addition - my company is currently looking to Asterisk as a potential candidate for this if FS can't do it. I want FS to win of course :-) Christian Jensen Software Development Manager Back Office ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, September 17, 2009 11:20 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] skill-based ACD On Sun, Sep 13, 2009 at 8:01 AM, mark morreny wrote: Hello Has any tried setting up an ACD based on skillset? The current out-of-box version of fifo does not seem to support acd based on agent skillset. Does anyone have any experience in doing it with some external scripting using lua or javascript? I am interested in hearing how others may have done it as I am trying to implement one myself. thx, mark I was curious about this myself. Even if someone has built a non-free skills-based ACD using FS I'd like to know about it. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/51c4fc52/attachment-0001.html From brian at freeswitch.org Thu Sep 17 12:53:28 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 17 Sep 2009 14:53:28 -0500 Subject: [Freeswitch-users] mod_conference performance In-Reply-To: <2160023e0909171158i118f2296o8366192b954f7c1e@mail.gmail.com> References: <2160023e0909161056p4338cc19rb982400ebf677c1d@mail.gmail.com> <2160023e0909171158i118f2296o8366192b954f7c1e@mail.gmail.com> Message-ID: <083EC95A-E1E4-4573-BCD6-C6C2CE68CABA@freeswitch.org> also was this 260 people in a single conference or multiple smaller conferences? /b On Sep 17, 2009, at 1:58 PM, RobertT wrote: > Okay, I've performed some additional tests and this is what I've > found: > > codec max calls > speex (8kHz) 50 > iLBC(8kHz) 50 > PCMU(8kHz) 260(approx*) > GSM(8kHz) 150(approx*) > speex(16kHz) 50 > G722(16kHz) 90(approx*) > > * - couldn't trace till the total load of CPU 'cause RDP was > timedout due to channel load. Calculated by trend. > > Still I think Linux tests are necessary. > > Cheers, Robert. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/96735c23/attachment.html From email.list.subscriber at gmail.com Thu Sep 17 13:00:59 2009 From: email.list.subscriber at gmail.com (email lists) Date: Thu, 17 Sep 2009 16:00:59 -0400 Subject: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error) In-Reply-To: <191c3a030909171157x2dc595b6o6508a1f110d74036@mail.gmail.com> References: <4EF4BF1E8F43894386584BE36354494A13E94D77@ZANEMS01.cc-ntd1.covad.com> <4ab23a57.151bf30a.1393.1f39@mx.google.com> <9859212C-CDCC-4F49-9055-5B309D79D331@freeswitch.org> <4ab247f8.171bf30a.41b9.1c84@mx.google.com> <191c3a030909170744t6bb456a5wbd53fe61a6efbe04@mail.gmail.com> <4ab27616.141bf30a.0af3.2571@mx.google.com> <191c3a030909171157x2dc595b6o6508a1f110d74036@mail.gmail.com> Message-ID: <4ab29532.151bf30a.1393.2d71@mx.google.com> Solaris OS is at 64 bit. Freeswitch was compiled as 32 bit however. Our sysadmin will attempt to compile it at 64 bit. In the meantime, I have moved Freeswitch over to the new server and am in the process of testing. Vladimir From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, September 17, 2009 2:58 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error) You never answered the question if you are using the 32 bit version of the OS If you are, make sure you do not, this has a huge impact on your performance. On Thu, Sep 17, 2009 at 12:48 PM, email lists wrote: I'm running on a (2) Dual Core, 16G RAM machine. Tried 5 calls per second with 3 minute duration -- noticed same issues. I throttled down to 5 calls per second, with a 30 second duration and it seems quite a bit more stable now, though this doesn't meet our requirements. Looking at moving Freeswitch to a Quad-core 2.8ghz server now, and will re-test. Vladimir From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Thursday, September 17, 2009 10:44 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error) You should get a more powerful machine if you intend to sustain upwards of 3600 calls at 20cps We recommend a 64 bit multi-core machine running 64bit Centos5.3 On Thu, Sep 17, 2009 at 9:31 AM, email lists wrote: Yes. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Thursday, September 17, 2009 10:07 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] 480 Temporarily Unavailable/503 Max Sessions (was 502 Bad Gateway: Destination out of order error) Are you using SIPP? /b On Sep 17, 2009, at 8:33 AM, email lists wrote: > Now, after a few minutes of running calls at 20 calls per second with > a 3 min. duration, I start seeing an influx of "480 Temporarily > Unavailable" SIP error messages. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use rs 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/8ba306a7/attachment-0001.html From kadantsev.d at gmail.com Thu Sep 17 13:01:51 2009 From: kadantsev.d at gmail.com (Dmitry Kadantsev) Date: Thu, 17 Sep 2009 22:01:51 +0200 Subject: [Freeswitch-users] mod_conference performance In-Reply-To: <083EC95A-E1E4-4573-BCD6-C6C2CE68CABA@freeswitch.org> References: <2160023e0909161056p4338cc19rb982400ebf677c1d@mail.gmail.com> <2160023e0909171158i118f2296o8366192b954f7c1e@mail.gmail.com> <083EC95A-E1E4-4573-BCD6-C6C2CE68CABA@freeswitch.org> Message-ID: <681a20520909171301k5e502b8ao3dc4aab064460f7@mail.gmail.com> Hi Brian, Robert has left to Germany. He will be again online on next Monday. -- Best regards, Dmitry Kadantsev On Thu, Sep 17, 2009 at 9:53 PM, Brian West wrote: > also was this 260 people in a single conference or multiple smaller > conferences? > /b > > On Sep 17, 2009, at 1:58 PM, RobertT wrote: > > Okay, I've performed some additional tests and this is what I've found: > * > codec* *max calls > *speex (8kHz) 50 > iLBC(8kHz) 50 > PCMU(8kHz) 260(approx*) > GSM(8kHz) 150(approx*) > speex(16kHz) 50 > G722(16kHz) 90(approx*) > > * - couldn't trace till the total load of CPU 'cause RDP was timedout due > to channel load. Calculated by trend. > > Still I think Linux tests are necessary. > > Cheers, Robert. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/44853d94/attachment.html From pawel at voiceworks.pl Thu Sep 17 13:04:57 2009 From: pawel at voiceworks.pl (=?UTF-8?B?UGF3ZcWCIFBpZXLFm2Npb25law==?=) Date: Thu, 17 Sep 2009 22:04:57 +0200 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> Message-ID: <4AB29669.7050209@voiceworks.pl> Michael Collins pisze: > > > On Sun, Sep 13, 2009 at 8:01 AM, mark morreny > wrote: > > Hello > > Has any tried setting up an ACD based on skillset? The current > out-of-box version of fifo does not seem to support acd based on > agent skillset. Does anyone have any experience in doing it with > some external scripting using lua or javascript? > > I am interested in hearing how others may have done it as I am > trying to implement one myself. > > thx, > > mark > > > I was curious about this myself. Even if someone has built a non-free > skills-based ACD using FS I'd like to know about it. > -MC > What is a skills-based ACD ? My FS based ACD allows agents to log-in to multiple queues at once and have different priority setting in each of the queues - does this count as skills-based ? It does not allow to limit the number of calls an agent can answer from a queue daily nor does it allow to set distribution of calls to an agent from his queues (other the by priority). Pawel, From pjintheusa at gmail.com Thu Sep 17 13:42:00 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 17 Sep 2009 16:42:00 -0400 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <4AB29669.7050209@voiceworks.pl> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> <4AB29669.7050209@voiceworks.pl> Message-ID: <367751820909171342r183ab9ecv56d67854a62c6d2e@mail.gmail.com> >> My FS based ACD allows agents to log-in to multiple queues at once What is the difference between those queues? Does each q require that agents have a different skill? ie Agent A has skills x,y,z and queue 1,2,3 have calls about x,y,z - there agent A logins to queue 1,2,3. 2009/9/17 Pawe? Pier?cionek > Michael Collins pisze: > > > > > > On Sun, Sep 13, 2009 at 8:01 AM, mark morreny > > wrote: > > > > Hello > > > > Has any tried setting up an ACD based on skillset? The current > > out-of-box version of fifo does not seem to support acd based on > > agent skillset. Does anyone have any experience in doing it with > > some external scripting using lua or javascript? > > > > I am interested in hearing how others may have done it as I am > > trying to implement one myself. > > > > thx, > > > > mark > > > > > > I was curious about this myself. Even if someone has built a non-free > > skills-based ACD using FS I'd like to know about it. > > -MC > > > What is a skills-based ACD ? > My FS based ACD allows agents to log-in to multiple queues at once and > have different priority setting in each of the queues - does this count > as skills-based ? > It does not allow to limit the number of calls an agent can answer from > a queue daily nor does it allow to set distribution of calls to an agent > from his queues (other the by priority). > > Pawel, > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/37c1b3ca/attachment.html From gshfreesw at gmail.com Thu Sep 17 13:48:12 2009 From: gshfreesw at gmail.com (Shameem Shiek) Date: Thu, 17 Sep 2009 16:48:12 -0400 Subject: [Freeswitch-users] Event socket library bug on hangup? In-Reply-To: <5070fcbd0909171208g593e79f0i8ea9832608a1d21b@mail.gmail.com> References: <5070fcbd0909171140v6d4755a7k52099acad58c28fc@mail.gmail.com> <191c3a030909171156m104b288u97fb8c4869a7f83c@mail.gmail.com> <5070fcbd0909171208g593e79f0i8ea9832608a1d21b@mail.gmail.com> Message-ID: <5070fcbd0909171348t5ccf0a55r7d321cc75e1b05a0@mail.gmail.com> I found the linger command on a old freeswitch user's email thread. Updated the Event socket outbound wiki with the command. On Thu, Sep 17, 2009 at 3:08 PM, Shameem Shiek wrote: > What does "linger" do? I do not see it documented anywhere. > > Why would I get a SERVER_DISCONNECTED instead of CHANNEL_HANGUP ? By the > way the BUG does not happen every time and normally I *DO* see the > CHANNEL_HANGUP come through in most cases. > > Thanks for the help. > > > On Thu, Sep 17, 2009 at 2:56 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> try sending the linger command >> >> linger\n\n >> >> >> >> >> On Thu, Sep 17, 2009 at 1:40 PM, Shameem Shiek wrote: >> >>> When a call is hangup, I get SERVER_DISCONNECTED Event over and over >>> again instead of CHANNEL_HANGUP event. Has anyone else experienced this? I >>> am using freeswitch 1.04. >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/98d41e42/attachment.html From pjintheusa at gmail.com Thu Sep 17 13:50:38 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 17 Sep 2009 16:50:38 -0400 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <367751820909171342r183ab9ecv56d67854a62c6d2e@mail.gmail.com> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> <4AB29669.7050209@voiceworks.pl> <367751820909171342r183ab9ecv56d67854a62c6d2e@mail.gmail.com> Message-ID: <367751820909171350vf8cd02dk6650bf4660af12d0@mail.gmail.com> see here: *http://en.wikipedia.org/wiki/Skills_based_routing* On Thu, Sep 17, 2009 at 4:42 PM, Phillip Jones wrote: > >> My FS based ACD allows agents to log-in to multiple queues at once > > What is the difference between those queues? Does each q require that > agents have a different skill? ie Agent A has skills x,y,z and queue 1,2,3 > have calls about x,y,z - there agent A logins to queue 1,2,3. > > 2009/9/17 Pawe? Pier?cionek > > Michael Collins pisze: >> > >> > >> > On Sun, Sep 13, 2009 at 8:01 AM, mark morreny > > > wrote: >> > >> > Hello >> > >> > Has any tried setting up an ACD based on skillset? The current >> > out-of-box version of fifo does not seem to support acd based on >> > agent skillset. Does anyone have any experience in doing it with >> > some external scripting using lua or javascript? >> > >> > I am interested in hearing how others may have done it as I am >> > trying to implement one myself. >> > >> > thx, >> > >> > mark >> > >> > >> > I was curious about this myself. Even if someone has built a non-free >> > skills-based ACD using FS I'd like to know about it. >> > -MC >> > >> What is a skills-based ACD ? >> My FS based ACD allows agents to log-in to multiple queues at once and >> have different priority setting in each of the queues - does this count >> as skills-based ? >> It does not allow to limit the number of calls an agent can answer from >> a queue daily nor does it allow to set distribution of calls to an agent >> from his queues (other the by priority). >> >> Pawel, >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/9bf6c849/attachment-0001.html From msc at freeswitch.org Thu Sep 17 13:51:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 13:51:08 -0700 Subject: [Freeswitch-users] fax detection In-Reply-To: <65d96fc80909150845i5b287bc2w5a04537c836b1b48@mail.gmail.com> References: <65d96fc80909150845i5b287bc2w5a04537c836b1b48@mail.gmail.com> Message-ID: <87f2f3b90909171351y401b2782offc5522b9fc0681c@mail.gmail.com> On Tue, Sep 15, 2009 at 8:45 AM, Tihomir Culjaga wrote: > Hi, > > is there any way to route fax calls according to the call capability? > > I mean .. if the fax call supports T.38 i'd like to route it to a T.38 > capable gateway. All other fax calls (meaning inband) should be handled by > FS/SpanDSP. > Of course, I know that every fax call starts as a voice call and upon fax > tone detection additional capabilities are being negotiated(T.38 or G711). > Can it be done in early media, before the call is even answered? > > I don't claim to be an expert in all this, especially T.38, but if I understand correctly, in both cases the call needs to be answered first. I'm pretty sure that the sending fax machine won't start emitting the 1100Hz tone until the receiving end answers. Also, with T.38 doesn't the call have to come up and then T.38 gets negotiated? (I don't know, I have only read about it.) -MC > > > So, here the goal is to have a T.38 capable GW handling T.38 calls while > SpanDSP handling T.30... > > > Any chance to do that with FS? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/7795c128/attachment.html From msc at freeswitch.org Thu Sep 17 13:54:33 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 13:54:33 -0700 Subject: [Freeswitch-users] Event socket library bug on hangup? In-Reply-To: <5070fcbd0909171348t5ccf0a55r7d321cc75e1b05a0@mail.gmail.com> References: <5070fcbd0909171140v6d4755a7k52099acad58c28fc@mail.gmail.com> <191c3a030909171156m104b288u97fb8c4869a7f83c@mail.gmail.com> <5070fcbd0909171208g593e79f0i8ea9832608a1d21b@mail.gmail.com> <5070fcbd0909171348t5ccf0a55r7d321cc75e1b05a0@mail.gmail.com> Message-ID: <87f2f3b90909171354p72dceb03vecfc63722c06f819@mail.gmail.com> On Thu, Sep 17, 2009 at 1:48 PM, Shameem Shiek wrote: > I found the linger command on a old freeswitch user's email thread. Updated > the Event socket outbound wiki with the command. > > > http://wiki.freeswitch.org/wiki/Event_socket_outbound#Events Look for "linger" in the events section of that page... -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/2712cb41/attachment.html From msc at freeswitch.org Thu Sep 17 13:56:33 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 13:56:33 -0700 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <4AB29669.7050209@voiceworks.pl> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> <4AB29669.7050209@voiceworks.pl> Message-ID: <87f2f3b90909171356q66773734h3043ea4fdd8f647@mail.gmail.com> 2009/9/17 Pawe? Pier?cionek > Michael Collins pisze: > > > > > > On Sun, Sep 13, 2009 at 8:01 AM, mark morreny > > wrote: > > > > Hello > > > > Has any tried setting up an ACD based on skillset? The current > > out-of-box version of fifo does not seem to support acd based on > > agent skillset. Does anyone have any experience in doing it with > > some external scripting using lua or javascript? > > > > I am interested in hearing how others may have done it as I am > > trying to implement one myself. > > > > thx, > > > > mark > > > > > > I was curious about this myself. Even if someone has built a non-free > > skills-based ACD using FS I'd like to know about it. > > -MC > > > What is a skills-based ACD ? > My FS based ACD allows agents to log-in to multiple queues at once and > have different priority setting in each of the queues - does this count > as skills-based ? > It does not allow to limit the number of calls an agent can answer from > a queue daily nor does it allow to set distribution of calls to an agent > from his queues (other the by priority). > > Pawel, > I would describe this scenario as "poor man's skills-based routing" - it's a bit of a hack but if it works then great. It isn't truly skills-based routing but it emulates some of the functionality of skills-based routing. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/0a899a80/attachment.html From shiyanov at gmail.com Thu Sep 17 13:57:26 2009 From: shiyanov at gmail.com (Artem Shiyanov) Date: Fri, 18 Sep 2009 00:57:26 +0400 Subject: [Freeswitch-users] "barge in" implementation with mod_socket and eavesdrop In-Reply-To: <191c3a030909150808q282d2ee7y3cf9ab2e4c8d77ae@mail.gmail.com> References: <191c3a030909150808q282d2ee7y3cf9ab2e4c8d77ae@mail.gmail.com> Message-ID: Anthony, thank you much, "three_way" is a powerful app! I've added small description in wiki, if you don't mind. http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_three_way Meanwhile, another question has been evolved: In words of the scenario below, now if A hangs up the call then all other (B, C) channels also being hanged up automatically by FreeSwitch. Is there any way to save the call B <--> C if A has hanged up the phone? Again, the whole scenario is the same as it is described in my first message. Artem On Tue, Sep 15, 2009 at 7:08 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > yes call the app as "three_way" like i said in the other thread. > > > On Tue, Sep 15, 2009 at 9:22 AM, Artem Shiyanov wrote: > >> Hello! >> >> I'm trying to implement "barge in" functionality (see >> http://www.yourdictionary.com/telecom/barge-in) with "eavesdrop" but >> still with no success. >> >> The situation is: >> - Person A calls to the extension: >> >> >> >> >> >> >> >> - I bridge him with person B with help of mod_socket: >> SendMsg >> call-command: execute >> execute-app-name: bridge >> execute-app-arg: >> >> - A and B talks >> >> - Person C decides to barge in the call A<-->B (to become a third >> participator in the call) >> a) I send (mod_socket): >> SendMsg >> call-command: execute >> execute-app-name: eavesdrop >> execute-app-arg: >> >> b) Then, as the spec says ( >> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop) >> I sent DTMF "3" with >> api uuid_send_dtmf 3 >> but it doesn't work. I mean that A can hear B and vice verse, but both A >> and B can't hear C. C also doesn't hear neither A nor B. >> >> If I press "3" on the C's softphone (latest X-Lite) then, really, C >> becomes a full-capabilities participator of the call. >> Instead of "uuid_send_dtmf" I tried: >> 1) >> sendevent DTMF >> Unique-ID: >> DTMF-Digit: 3 >> DTMF-Duration: 2000 >> >> 2) first make queue_dtmf for the , and then eavesdrop >> >> 3) >> SendMsg >> call-command: execute >> execute-app-name: gentones >> execute-app-arg: 3 >> >> 4) >> SendMsg >> call-command: execute >> execute-app-name: send_dtmf >> execute-app-arg: 3 >> >> And none of these methods leads to the "barged in" call. >> >> Anyone knows how to press "3" programmatically on behalf of the given >> channel with mod_socket?! >> >> >> Artem >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/cf59f4e1/attachment.html From msc at freeswitch.org Thu Sep 17 13:57:40 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 13:57:40 -0700 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <11372C8B9E603F4FACDE6AB18256DEC695A9EC@srvmtel.office.mtel.nl> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <11372C8B9E603F4FACDE6AB18256DEC695A9EC@srvmtel.office.mtel.nl> Message-ID: <87f2f3b90909171357y40fe0db4kef1d59fb86e9790c@mail.gmail.com> On Thu, Sep 17, 2009 at 12:13 PM, Remko Kloosterman wrote: > I have been working on several voice projects in the past with ACD > features, mostly based on TDM technology. It's all commercial stuff, but I > have the experience and I am willing to share that. If anyone wishes to > start such a development I'm sure I can dig up a functional model and help > with the design. > > I would like to see the functional model. That sounds interesting. We could take it from there. Perhaps the FS community will have a few members willing to help out. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/8557bfd8/attachment-0001.html From tayeb.meftah at gmail.com Thu Sep 17 15:01:00 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 17 Sep 2009 22:01:00 +0000 Subject: [Freeswitch-users] IAX Origination error Message-ID: <4AB2B19C.2040305@gmail.com> hello, i have a problem while originating a call using mod_iax: originate IAX/guest at pbx.digium.com/s 1000 freeswitch will crach and exit automatikaly anthm updated the mod_iax but same problem see the trace in the Pastebin (http://pastebin:freeswitch at pastebin.freeswitch.org/1040) this is the asterisk definition of this extension: exten => 500,n,Dial(IAX2/guest at pbx.digium.com/s at default) ; Call the Asterisk demo any help? thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4435 (20090917) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __________ Information from ESET NOD32 Antivirus, version of virus signature database 4435 (20090917) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com __________ Information from ESET NOD32 Antivirus, version of virus signature database 4435 (20090917) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/ce50d549/attachment.html From msc at freeswitch.org Thu Sep 17 14:05:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 14:05:16 -0700 Subject: [Freeswitch-users] Please Help: TDM Hardware For Testing Message-ID: <87f2f3b90909171405r3d496f1dq2a171ce0a9859ece@mail.gmail.com> Hello all! We could use your help with something. We've had several volunteers willing to assist with testing and debugging various scenarios with TDM circuits. However, there is a need to get TDM hardware into the hands of those volunteers. If you have old TDM hardware, be it analog or digital, that is sitting around collecting dust and you'd like to donate it (or loan it) to the cause then please email me off list. Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/0a5f620d/attachment.html From sramsey at sipinterchange.com Thu Sep 17 14:15:23 2009 From: sramsey at sipinterchange.com (Shelby Ramsey) Date: Thu, 17 Sep 2009 16:15:23 -0500 Subject: [Freeswitch-users] [Freeswitch-dev] Please Help: TDM Hardware For Testing In-Reply-To: <87f2f3b90909171405r3d496f1dq2a171ce0a9859ece@mail.gmail.com> References: <87f2f3b90909171405r3d496f1dq2a171ce0a9859ece@mail.gmail.com> Message-ID: <4AB2A6EB.8080404@sipinterchange.com> MC, Just sent you the message offline ... SDR Michael Collins wrote: > Hello all! > > We could use your help with something. We've had several volunteers > willing to assist with testing and debugging various scenarios with > TDM circuits. However, there is a need to get TDM hardware into the > hands of those volunteers. If you have old TDM hardware, be it analog > or digital, that is sitting around collecting dust and you'd like to > donate it (or loan it) to the cause then please email me off list. > > Thanks! > -MC > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > From tayeb.meftah at gmail.com Thu Sep 17 15:14:56 2009 From: tayeb.meftah at gmail.com (Meftah Tayeb) Date: Thu, 17 Sep 2009 22:14:56 +0000 Subject: [Freeswitch-users] IAX Origination error In-Reply-To: <4AB2B19C.2040305@gmail.com> References: <4AB2B19C.2040305@gmail.com> Message-ID: <4AB2B4E0.1090406@gmail.com> Meftah Tayeb a ?crit : > hello, > i have a problem while originating a call using mod_iax: > originate IAX/guest at pbx.digium.com/s 1000 > freeswitch will crach and exit automatikaly > anthm updated the mod_iax but same problem > see the trace in the > Pastebin > (http://pastebin:freeswitch at pastebin.freeswitch.org/1040) > this is the asterisk definition of this extension: > exten => 500,n,Dial(IAX2/guest at pbx.digium.com/s at default) ; Call the > Asterisk demo > any help? > thanks > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4435 (20090917) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4435 (20090917) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4435 (20090917) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com > > > __________ Information from ESET NOD32 Antivirus, version of virus > signature database 4435 (20090917) __________ > > The message was checked by ESET NOD32 Antivirus. > > http://www.eset.com hello, sory for the ERROR in the URL the pb URL is: http://pastebin.freeswitch.org/10408 thanks __________ Information from ESET NOD32 Antivirus, version of virus signature database 4435 (20090917) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/8b0eb76b/attachment.html From msc at freeswitch.org Thu Sep 17 14:17:17 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 14:17:17 -0700 Subject: [Freeswitch-users] IAX Origination error In-Reply-To: <4AB2B19C.2040305@gmail.com> References: <4AB2B19C.2040305@gmail.com> Message-ID: <87f2f3b90909171417r5ff87044ree1637fd38fd6fca@mail.gmail.com> On Thu, Sep 17, 2009 at 3:01 PM, Meftah Tayeb wrote: > hello, > i have a problem while originating a call using mod_iax: > originate IAX/guest at pbx.digium.com/s 1000 > freeswitch will crach and exit automatikaly > anthm updated the mod_iax but same problem > see the trace in the > Pastebin > (http://pastebin:freeswitch at pastebin.freeswitch.org/1040) > FYI, Meftah reports that the correct pastebin is: http://pastebin.freeswitch.org/10408 -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/7feb06a7/attachment.html From aep.lists at it46.se Thu Sep 17 14:20:14 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Thu, 17 Sep 2009 23:20:14 +0200 Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! Message-ID: We are trying to create a callback application in Javascript. We get the callerid from the unanswered call and after destroying the session, we initiate a callback to the user to conenct it to a local extension in the dialplan. Although we have tried to destroy the first session, or even invoke a second script using apiExecute("jsrun",dialer.js"), tried session.hangup() or exit()... the first session does not seem to close properly until the whole chain of scripts are completed. Here is a piece of code that shows the concept (yes!, the sleep function is far from ideal. CPU loves it! ) function sleep(milliseconds) { var start = new Date().getTime(); for (var i = 0; i < 1e7; i++) { if ((new Date().getTime() - start) > milliseconds){ break; } } } if (session.ready()) { //We catch the caller_id caller_id_num = session.caller_id_num; console_log("Now we got your Caller ID\n"); //How long we want to wait to trigger a call back session.execute("sleep",5000); console_log("We have waited a while... time to create the callback\n"); //apiExecute("jsrun", "callback.js"); } //Destroy the session... session.destroy(); session=undefined; sleep(10000); //Preparing callback session2 = new Session('{ignore_early_media=true}celliax/interface1/600464646'); session2.setAutoHangup(false); session2.answer(); exit(); ++ Wisdom thoughts? -- Stopping junk mailers is good for the environment From mgg at giagnocavo.net Thu Sep 17 14:31:41 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 17 Sep 2009 17:31:41 -0400 Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! In-Reply-To: References: Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DCBE4F41@mse17be1.mse17.exchange.ms> So, what happens is that when you are executing an app, the state is CS_EXECUTE. Even if the session is hungup, the state machine doesn't go through all the hangup code until your app executes. The easiest workaround is probably to start a background api (bgapi?) call to a script. This will happen on another thread, then allow your current thread to execute and the hangup code will execute. This should work just fine, I think. (You can stop reading here.) But wait, there's even more fun! anthm recently checked in a change a couple days that lets you work around this. Don't call destroy, call hangup on the session, on that session's thread. This will perform a hangup, then progress the state machine. Then the session will truly be hungup. Maybe you need update your freeswitch code, if this is not happening for you. If you updated and hangup still isn't hanging up, you might want to ask specifically about that. Or, you may need to call switch_core_session_hangup_state directly -- just hangup alone might not do the trick. This is a C function, and not exposed to languages by default - you can either patch javascript plugin to expose this safely (and I have no idea what this means for the javascript runtime), or use a more capable plugin like mod_managed which _does_ expose all the C functions, and lets you call in and out of them as you please. And now, someone who knows what they're talking about will chime in and point out what I got wrong. Thanks, -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Alberto Escudero Sent: Thursday, September 17, 2009 3:20 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! We are trying to create a callback application in Javascript. We get the callerid from the unanswered call and after destroying the session, we initiate a callback to the user to conenct it to a local extension in the dialplan. Although we have tried to destroy the first session, or even invoke a second script using apiExecute("jsrun",dialer.js"), tried session.hangup() or exit()... the first session does not seem to close properly until the whole chain of scripts are completed. Here is a piece of code that shows the concept (yes!, the sleep function is far from ideal. CPU loves it! ) function sleep(milliseconds) { var start = new Date().getTime(); for (var i = 0; i < 1e7; i++) { if ((new Date().getTime() - start) > milliseconds){ break; } } } if (session.ready()) { //We catch the caller_id caller_id_num = session.caller_id_num; console_log("Now we got your Caller ID\n"); //How long we want to wait to trigger a call back session.execute("sleep",5000); console_log("We have waited a while... time to create the callback\n"); //apiExecute("jsrun", "callback.js"); } //Destroy the session... session.destroy(); session=undefined; sleep(10000); //Preparing callback session2 = new Session('{ignore_early_media=true}celliax/interface1/600464646'); session2.setAutoHangup(false); session2.answer(); exit(); ++ Wisdom thoughts? -- Stopping junk mailers is good for the environment _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From pjintheusa at gmail.com Thu Sep 17 14:37:47 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 17 Sep 2009 17:37:47 -0400 Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! In-Reply-To: References: Message-ID: <367751820909171437r6530828ayee5d8cc392c2c7fd@mail.gmail.com> session.dispose(); ??? On Thu, Sep 17, 2009 at 5:20 PM, Alberto Escudero wrote: > We are trying to create a callback application in Javascript. We get the > callerid from the unanswered call and after destroying the session, we > initiate a callback to the user to conenct it to a local extension in the > dialplan. > > Although we have tried to destroy the first session, or even invoke a > second script using apiExecute("jsrun",dialer.js"), tried session.hangup() > or exit()... the first session does not seem to close properly until the > whole chain of scripts are completed. > > Here is a piece of code that shows the concept (yes!, the sleep function > is far from ideal. CPU loves it! ) > > function sleep(milliseconds) { > var start = new Date().getTime(); > for (var i = 0; i < 1e7; i++) { > if ((new Date().getTime() - start) > milliseconds){ > break; > } > } > } > > if (session.ready()) { > //We catch the caller_id > caller_id_num = session.caller_id_num; > > console_log("Now we got your Caller ID\n"); > > //How long we want to wait to trigger a call back > session.execute("sleep",5000); > > console_log("We have waited a while... time to create the > callback\n"); > > //apiExecute("jsrun", "callback.js"); > } > > //Destroy the session... > session.destroy(); > session=undefined; > > sleep(10000); > > //Preparing callback > session2 = new > Session('{ignore_early_media=true}celliax/interface1/600464646'); > session2.setAutoHangup(false); > session2.answer(); > exit(); > > ++ > Wisdom thoughts? > > -- > Stopping junk mailers is good for the environment > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/dd92e4f2/attachment-0001.html From anthony.minessale at gmail.com Thu Sep 17 15:04:03 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Sep 2009 17:04:03 -0500 Subject: [Freeswitch-users] "barge in" implementation with mod_socket and eavesdrop In-Reply-To: References: <191c3a030909150808q282d2ee7y3cf9ab2e4c8d77ae@mail.gmail.com> Message-ID: <191c3a030909171504u388783r9b6bef32a30680db@mail.gmail.com> to do that you would have to transfer all the parties into a conference On Thu, Sep 17, 2009 at 3:57 PM, Artem Shiyanov wrote: > Anthony, > thank you much, "three_way" is a powerful app! > I've added small description in wiki, if you don't mind. > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_three_way > > Meanwhile, another question has been evolved: > In words of the scenario below, now if A hangs up the call then all other > (B, C) channels also being hanged up automatically by FreeSwitch. Is there > any way to save the call B <--> C if A has hanged up the phone? Again, the > whole scenario is the same as it is described in my first message. > > > Artem > > > > On Tue, Sep 15, 2009 at 7:08 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> yes call the app as "three_way" like i said in the other thread. >> >> >> On Tue, Sep 15, 2009 at 9:22 AM, Artem Shiyanov wrote: >> >>> Hello! >>> >>> I'm trying to implement "barge in" functionality (see >>> http://www.yourdictionary.com/telecom/barge-in) with "eavesdrop" but >>> still with no success. >>> >>> The situation is: >>> - Person A calls to the extension: >>> >>> >>> >>> >>> >>> >>> >>> - I bridge him with person B with help of mod_socket: >>> SendMsg >>> call-command: execute >>> execute-app-name: bridge >>> execute-app-arg: >>> >>> - A and B talks >>> >>> - Person C decides to barge in the call A<-->B (to become a third >>> participator in the call) >>> a) I send (mod_socket): >>> SendMsg >>> call-command: execute >>> execute-app-name: eavesdrop >>> execute-app-arg: >>> >>> b) Then, as the spec says ( >>> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_eavesdrop) >>> I sent DTMF "3" with >>> api uuid_send_dtmf 3 >>> but it doesn't work. I mean that A can hear B and vice verse, but both A >>> and B can't hear C. C also doesn't hear neither A nor B. >>> >>> If I press "3" on the C's softphone (latest X-Lite) then, really, C >>> becomes a full-capabilities participator of the call. >>> Instead of "uuid_send_dtmf" I tried: >>> 1) >>> sendevent DTMF >>> Unique-ID: >>> DTMF-Digit: 3 >>> DTMF-Duration: 2000 >>> >>> 2) first make queue_dtmf for the , and then eavesdrop >>> >>> 3) >>> SendMsg >>> call-command: execute >>> execute-app-name: gentones >>> execute-app-arg: 3 >>> >>> 4) >>> SendMsg >>> call-command: execute >>> execute-app-name: send_dtmf >>> execute-app-arg: 3 >>> >>> And none of these methods leads to the "barged in" call. >>> >>> Anyone knows how to press "3" programmatically on behalf of the given >>> channel with mod_socket?! >>> >>> >>> Artem >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/f816769d/attachment.html From mgg at giagnocavo.net Thu Sep 17 15:08:48 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 17 Sep 2009 18:08:48 -0400 Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! In-Reply-To: <367751820909171437r6530828ayee5d8cc392c2c7fd@mail.gmail.com> References: <367751820909171437r6530828ayee5d8cc392c2c7fd@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DCBE4F5D@mse17be1.mse17.exchange.ms> Dispose is a .NET only thing. But I think you are right - with anthm's changes, any way you kill your session, if you're on the right thread, it should really hangup. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Phillip Jones Sent: Thursday, September 17, 2009 3:38 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! session.dispose(); ??? On Thu, Sep 17, 2009 at 5:20 PM, Alberto Escudero > wrote: We are trying to create a callback application in Javascript. We get the callerid from the unanswered call and after destroying the session, we initiate a callback to the user to conenct it to a local extension in the dialplan. Although we have tried to destroy the first session, or even invoke a second script using apiExecute("jsrun",dialer.js"), tried session.hangup() or exit()... the first session does not seem to close properly until the whole chain of scripts are completed. Here is a piece of code that shows the concept (yes!, the sleep function is far from ideal. CPU loves it! ) function sleep(milliseconds) { var start = new Date().getTime(); for (var i = 0; i < 1e7; i++) { if ((new Date().getTime() - start) > milliseconds){ break; } } } if (session.ready()) { //We catch the caller_id caller_id_num = session.caller_id_num; console_log("Now we got your Caller ID\n"); //How long we want to wait to trigger a call back session.execute("sleep",5000); console_log("We have waited a while... time to create the callback\n"); //apiExecute("jsrun", "callback.js"); } //Destroy the session... session.destroy(); session=undefined; sleep(10000); //Preparing callback session2 = new Session('{ignore_early_media=true}celliax/interface1/600464646'); session2.setAutoHangup(false); session2.answer(); exit(); ++ Wisdom thoughts? -- Stopping junk mailers is good for the environment _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/10af8c36/attachment.html From anthony.minessale at gmail.com Thu Sep 17 15:17:04 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Sep 2009 17:17:04 -0500 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> Message-ID: <191c3a030909171517u239394bas1d5ab96914aff902@mail.gmail.com> I can tell you from years of painful experience, don't use asterisk for queues. see http://www.freeswitch.org/node/117 You don't have to use FS, but please don't let the asterisk siren lure you to the rocks. mod_fifo is like a tool with basic functions you can exploit however you wish, it does not try to do high level features because those are best left in external logic. mod_fifo has priorities which means each individual fifo is really an array of 10 fifos when you set the priority you are choosing which index in the array to insert the caller. when an agent belongs to a queue he drills down the array from 0-9 so you could for instance put everyone in 5 by default and put more important people in 0 so they always go to the front when you assign an agent to take calls off hook you can set a fifo_pop_order variable that tells you which array indexes to service and in what order. so if you pretend slot 1 is for general problems and slot 2 is for hard problems you can put one agent in 1,2 and a more stupid agent in just 1 *shrug* On Thu, Sep 17, 2009 at 1:56 PM, Christian Jensen < Christian.Jensen at teligence.net> wrote: > This would be a fantastic addition ? my company is currently looking to > Asterisk as a potential candidate for this if FS can?t do it. > > > > I want FS to win of course J > > > > *Christian Jensen* > Software Development Manager > > Back Office > ------------------------------ > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Thursday, September 17, 2009 11:20 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] skill-based ACD > > > > > > On Sun, Sep 13, 2009 at 8:01 AM, mark morreny > wrote: > > Hello > > > > Has any tried setting up an ACD based on skillset? The current out-of-box > version of fifo does not seem to support acd based on agent skillset. Does > anyone have any experience in doing it with some external scripting using > lua or javascript? > > > > I am interested in hearing how others may have done it as I am trying to > implement one myself. > > > > thx, > > > > mark > > > > > I was curious about this myself. Even if someone has built a non-free > skills-based ACD using FS I'd like to know about it. > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/23e2ce39/attachment-0001.html From tculjaga at gmail.com Thu Sep 17 15:25:09 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 18 Sep 2009 00:25:09 +0200 Subject: [Freeswitch-users] fax detection In-Reply-To: <87f2f3b90909171351y401b2782offc5522b9fc0681c@mail.gmail.com> References: <65d96fc80909150845i5b287bc2w5a04537c836b1b48@mail.gmail.com> <87f2f3b90909171351y401b2782offc5522b9fc0681c@mail.gmail.com> Message-ID: <65d96fc80909171525r72cbc381u6a4565269fe02512@mail.gmail.com> Hi Michael, thanks for your response. i think it will be enough to check the call capability... we always know the call is fax. We just need to apply the correct protocol :P let's suppose you have 2 incoming calls: 1. SDP containing G711, gsm, T.38 caps 2. SDP containing G711, gsm caps the caps will be known on within INVITE message and FS can act accordingly. - If there is no T.38 support within SDP, start fax application (SpanDSP). - If there is T.38 suport within the SDP, route the call to some predefined gateway meant for T.38 fax receiving. In both cases, the fax should be received :P. So, any chance to route the call according to T.38 caps within SDP message? Now, for sending faxes there is some challenge.... we still have 2 calls: we start sending a fax inband (SpanDSP) as this is the only thing we know... and: - if we receive 200 OK with SDP containing G711, anyCaps => continue with InBand Fax - if we receive 200 OK with DSP containing anyCompressedCaps, T.38 => drop the call without sending ACK to 200 OK ... and move the fax to be sent into a different directory. well, as i said, while receiving faxes will work 100%, sending is tricky... but it might work. what do you guys think? T. On Thu, Sep 17, 2009 at 10:51 PM, Michael Collins wrote: > > > On Tue, Sep 15, 2009 at 8:45 AM, Tihomir Culjaga wrote: > >> Hi, >> >> is there any way to route fax calls according to the call capability? >> >> I mean .. if the fax call supports T.38 i'd like to route it to a T.38 >> capable gateway. All other fax calls (meaning inband) should be handled by >> FS/SpanDSP. >> Of course, I know that every fax call starts as a voice call and upon fax >> tone detection additional capabilities are being negotiated(T.38 or G711). >> Can it be done in early media, before the call is even answered? >> >> I don't claim to be an expert in all this, especially T.38, but if I > understand correctly, in both cases the call needs to be answered first. I'm > pretty sure that the sending fax machine won't start emitting the 1100Hz > tone until the receiving end answers. Also, with T.38 doesn't the call have > to come up and then T.38 gets negotiated? (I don't know, I have only read > about it.) > -MC > >> >> >> So, here the goal is to have a T.38 capable GW handling T.38 calls while >> SpanDSP handling T.30... >> >> >> Any chance to do that with FS? >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/87ede743/attachment.html From anthony.minessale at gmail.com Thu Sep 17 15:29:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 17 Sep 2009 17:29:19 -0500 Subject: [Freeswitch-users] Event socket library bug on hangup? In-Reply-To: <5070fcbd0909171208g593e79f0i8ea9832608a1d21b@mail.gmail.com> References: <5070fcbd0909171140v6d4755a7k52099acad58c28fc@mail.gmail.com> <191c3a030909171156m104b288u97fb8c4869a7f83c@mail.gmail.com> <5070fcbd0909171208g593e79f0i8ea9832608a1d21b@mail.gmail.com> Message-ID: <191c3a030909171529k5cfaf2e7ub16d1820d4d3d181@mail.gmail.com> it's a race between getting the event or getting the disconnect from the socket where if you need to get the channel hangup event instead of just getting the socket disconnect you can send linger which makes the socket connection wait for the hangup event to disconnect. On Thu, Sep 17, 2009 at 2:08 PM, Shameem Shiek wrote: > What does "linger" do? I do not see it documented anywhere. > > Why would I get a SERVER_DISCONNECTED instead of CHANNEL_HANGUP ? By the > way the BUG does not happen every time and normally I *DO* see the > CHANNEL_HANGUP come through in most cases. > > Thanks for the help. > > > On Thu, Sep 17, 2009 at 2:56 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> try sending the linger command >> >> linger\n\n >> >> >> >> >> On Thu, Sep 17, 2009 at 1:40 PM, Shameem Shiek wrote: >> >>> When a call is hangup, I get SERVER_DISCONNECTED Event over and over >>> again instead of CHANNEL_HANGUP event. Has anyone else experienced this? I >>> am using freeswitch 1.04. >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/55368cff/attachment.html From jmesquita at freeswitch.org Thu Sep 17 15:40:28 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Thu, 17 Sep 2009 19:40:28 -0300 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <191c3a030909171517u239394bas1d5ab96914aff902@mail.gmail.com> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> <191c3a030909171517u239394bas1d5ab96914aff902@mail.gmail.com> Message-ID: I would be very interested in getting my poor programming skills into getting some decent real skill based routing working and shut those Avaya bastards up. Functional model? Get it to me and I will try to make it happen as time lets me. jmesquita On Thu, Sep 17, 2009 at 7:17 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I can tell you from years of painful experience, don't use asterisk for > queues. > see http://www.freeswitch.org/node/117 > > You don't have to use FS, but please don't let the asterisk siren lure you > to the rocks. > > mod_fifo is like a tool with basic functions you can exploit however you > wish, it does not try to do high level > features because those are best left in external logic. > > > mod_fifo has priorities which means each individual fifo is really an array > of 10 fifos > when you set the priority you are choosing which index in the array to > insert the caller. > when an agent belongs to a queue he drills down the array from 0-9 so you > could for instance put everyone in 5 by default and put more > important people in 0 so they always go to the front > > when you assign an agent to take calls off hook you can set a > fifo_pop_order variable that tells you which array indexes to service and in > what order. > so if you pretend slot 1 is for general problems and slot 2 is for hard > problems you can put one agent in 1,2 and a more stupid agent in just 1 > > *shrug* > > > > On Thu, Sep 17, 2009 at 1:56 PM, Christian Jensen < > Christian.Jensen at teligence.net> wrote: > >> This would be a fantastic addition ? my company is currently looking to >> Asterisk as a potential candidate for this if FS can?t do it. >> >> >> >> I want FS to win of course J >> >> >> >> *Christian Jensen* >> Software Development Manager >> >> Back Office >> ------------------------------ >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael >> Collins >> *Sent:* Thursday, September 17, 2009 11:20 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] skill-based ACD >> >> >> >> >> >> On Sun, Sep 13, 2009 at 8:01 AM, mark morreny >> wrote: >> >> Hello >> >> >> >> Has any tried setting up an ACD based on skillset? The current out-of-box >> version of fifo does not seem to support acd based on agent skillset. Does >> anyone have any experience in doing it with some external scripting using >> lua or javascript? >> >> >> >> I am interested in hearing how others may have done it as I am trying to >> implement one myself. >> >> >> >> thx, >> >> >> >> mark >> >> >> >> >> I was curious about this myself. Even if someone has built a non-free >> skills-based ACD using FS I'd like to know about it. >> -MC >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/1d29dcfa/attachment-0001.html From pjintheusa at gmail.com Thu Sep 17 16:13:36 2009 From: pjintheusa at gmail.com (Phillip Jones) Date: Thu, 17 Sep 2009 16:13:36 -0700 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <191c3a030909171517u239394bas1d5ab96914aff902@mail.gmail.com> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> <191c3a030909171517u239394bas1d5ab96914aff902@mail.gmail.com> Message-ID: <367751820909171613u752aa972l981dd33046952c07@mail.gmail.com> I would be interested in this too.... Concerning mod_fifo - can you restrict an agent to a slot. So lets say DNIS A is for product A and DNIS B product B - some agents know both - some know just 1 product - would that be possible? On Thu, Sep 17, 2009 at 3:17 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I can tell you from years of painful experience, don't use asterisk for > queues. > see http://www.freeswitch.org/node/117 > > You don't have to use FS, but please don't let the asterisk siren lure you > to the rocks. > > mod_fifo is like a tool with basic functions you can exploit however you > wish, it does not try to do high level > features because those are best left in external logic. > > > mod_fifo has priorities which means each individual fifo is really an array > of 10 fifos > when you set the priority you are choosing which index in the array to > insert the caller. > when an agent belongs to a queue he drills down the array from 0-9 so you > could for instance put everyone in 5 by default and put more > important people in 0 so they always go to the front > > when you assign an agent to take calls off hook you can set a > fifo_pop_order variable that tells you which array indexes to service and in > what order. > so if you pretend slot 1 is for general problems and slot 2 is for hard > problems you can put one agent in 1,2 and a more stupid agent in just 1 > > *shrug* > > > > On Thu, Sep 17, 2009 at 1:56 PM, Christian Jensen < > Christian.Jensen at teligence.net> wrote: > >> This would be a fantastic addition ? my company is currently looking to >> Asterisk as a potential candidate for this if FS can?t do it. >> >> >> >> I want FS to win of course J >> >> >> >> *Christian Jensen* >> Software Development Manager >> >> Back Office >> ------------------------------ >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael >> Collins >> *Sent:* Thursday, September 17, 2009 11:20 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] skill-based ACD >> >> >> >> >> >> On Sun, Sep 13, 2009 at 8:01 AM, mark morreny >> wrote: >> >> Hello >> >> >> >> Has any tried setting up an ACD based on skillset? The current out-of-box >> version of fifo does not seem to support acd based on agent skillset. Does >> anyone have any experience in doing it with some external scripting using >> lua or javascript? >> >> >> >> I am interested in hearing how others may have done it as I am trying to >> implement one myself. >> >> >> >> thx, >> >> >> >> mark >> >> >> >> >> I was curious about this myself. Even if someone has built a non-free >> skills-based ACD using FS I'd like to know about it. >> -MC >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/b6811f77/attachment.html From msc at freeswitch.org Thu Sep 17 16:22:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 16:22:14 -0700 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <367751820909171613u752aa972l981dd33046952c07@mail.gmail.com> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> <191c3a030909171517u239394bas1d5ab96914aff902@mail.gmail.com> <367751820909171613u752aa972l981dd33046952c07@mail.gmail.com> Message-ID: <87f2f3b90909171622m2515b62dpaa27ce77e6cfe7f7@mail.gmail.com> On Thu, Sep 17, 2009 at 4:13 PM, Phillip Jones wrote: > I would be interested in this too.... > > Concerning mod_fifo - can you restrict an agent to a slot. So lets say DNIS > A is for product A and DNIS B product B - some agents know both - some know > just 1 product - would that be possible? > Per Tony's post: when you assign an agent to take calls off hook you can set a fifo_pop_order variable that tells you which array indexes to service and in what order. so if you pretend slot 1 is for general problems and slot 2 is for hard problems you can put one agent in 1,2 and a more stupid agent in just 1 So it's a matter of setting up the routing on the calls and the priorities that the agents are allowed to answer. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/e1eb9016/attachment.html From gmaruzz at celliax.org Thu Sep 17 16:27:15 2009 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Fri, 18 Sep 2009 01:27:15 +0200 Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DCBE4F5D@mse17be1.mse17.exchange.ms> References: <367751820909171437r6530828ayee5d8cc392c2c7fd@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DCBE4F5D@mse17be1.mse17.exchange.ms> Message-ID: <7b197bef0909171627p3c4bc8a2x4068c239eebd0f52@mail.gmail.com> On Fri, Sep 18, 2009 at 12:08 AM, Michael Giagnocavo wrote: > Dispose is a .NET only thing. But I think you are right ? with anthm?s > changes, any way you kill your session, if you?re on the right thread, it > should really hangup. > Problem is, we are trying to *not answer* the incoming call, get the callid from the ring, destroy the session, create another session (on the same, monoline interface), and make an outbound call. Javascript (last svn) give us a 2009-09-18 01:18:49.291721 [ERR] inline:1 Session is not answered! if we try to session.hangup() a session that was not answered (by the way, it makes sense). -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From mcampbellsmith at gmail.com Thu Sep 17 17:15:06 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Fri, 18 Sep 2009 10:15:06 +1000 Subject: [Freeswitch-users] Simple call waiting question In-Reply-To: References: <33c87fa30909170558v4b032da2w6f8202b187d3bcf6@mail.gmail.com> Message-ID: <33c87fa30909171715v439a14d1x2f6ff8675128b0f5@mail.gmail.com> Thanks Brian, (I bought a dud phone.. and its a new DECT! - crazy) I am using the 5900 and 5901 for parking/unparking. That functionality works fine and I can park/unpark the B leg as I wish. The problem is that if I park the B-leg, the A-leg then gets a busy signal. If the A leg is then hung up, a user-busy signal is sent to the C-leg, so the call goes to voicemail. What I want to happen is park B and answer C directly. Is this possible? On Thu, Sep 17, 2009 at 11:21 PM, Brian West wrote: > Personally I would throw the phone in the trash. ?:P > > In the default dialplan look at 5900 for park and 5901 for unpark. > > /b > > > On Sep 17, 2009, at 7:58 AM, Mark Campbell-Smith wrote: > >> I am trying to create a simple call waiting dialplan as my phone does >> not have Recall button. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mgg at giagnocavo.net Thu Sep 17 17:53:55 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 17 Sep 2009 20:53:55 -0400 Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! In-Reply-To: <7b197bef0909171627p3c4bc8a2x4068c239eebd0f52@mail.gmail.com> References: <367751820909171437r6530828ayee5d8cc392c2c7fd@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DCBE4F5D@mse17be1.mse17.exchange.ms> <7b197bef0909171627p3c4bc8a2x4068c239eebd0f52@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DCBE4FC6@mse17be1.mse17.exchange.ms> Oh, weird. Seems to work in other languages. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Thursday, September 17, 2009 5:27 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! On Fri, Sep 18, 2009 at 12:08 AM, Michael Giagnocavo wrote: > Dispose is a .NET only thing. But I think you are right - with anthm's > changes, any way you kill your session, if you're on the right thread, it > should really hangup. > Problem is, we are trying to *not answer* the incoming call, get the callid from the ring, destroy the session, create another session (on the same, monoline interface), and make an outbound call. Javascript (last svn) give us a 2009-09-18 01:18:49.291721 [ERR] inline:1 Session is not answered! if we try to session.hangup() a session that was not answered (by the way, it makes sense). -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jason at jasonjgw.net Thu Sep 17 19:07:26 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 18 Sep 2009 12:07:26 +1000 Subject: [Freeswitch-users] Build problems with Shoutcast module under Debian Message-ID: <20090918020726.GA3242@jdc.jasonjgw.net> While trying to build FreeSWITCH rev. 14913, compilation failed with the following. the operating system is Debian Sid. Ogg development files are installed, but libogg.la does not exist anywhere. I'm still using libtool 1.5.26, because the build problems with FreeSWITCH and libtool 2 under Debian haven't been resolved. As soon as someone takes over the Debian packaging I'll gladly help out with testing and fixes - I'm far too busy at the moment to work on it intensively. ranlib .libs/libshout.a rm -fr .libs/libshout.lax creating libshout.la /bin/sed: can't read /usr/lib/libogg.la: No such file or directory libtool: link: `/usr/lib/libogg.la' is not a valid libtool archive make[10]: *** [libshout.la] Error 1 From jason at jasonjgw.net Thu Sep 17 19:55:35 2009 From: jason at jasonjgw.net (Jason White) Date: Fri, 18 Sep 2009 12:55:35 +1000 Subject: [Freeswitch-users] Build problems with Shoutcast module under Debian In-Reply-To: <20090918020726.GA3242@jdc.jasonjgw.net> References: <20090918020726.GA3242@jdc.jasonjgw.net> Message-ID: <20090918025535.GA19122@jdc.jasonjgw.net> It turns out that Debian recently removed the libogg.la file, deliberately, from the package. From pete at privateconnect.com Thu Sep 17 21:21:42 2009 From: pete at privateconnect.com (Pete Mueller) Date: Thu, 17 Sep 2009 21:21:42 -0700 Subject: [Freeswitch-users] skill-based ACD Message-ID: <20090917212142.2ad02225396a31c9de30536f2e338977.9c57d3c8f2.wbe@email04.secureserver.net> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/23cdf3dd/attachment.html From listas at askterisk.com Thu Sep 17 19:24:05 2009 From: listas at askterisk.com (Marcelo Sosa - LST) Date: Thu, 17 Sep 2009 23:24:05 -0300 Subject: [Freeswitch-users] mod_lcr and indexes Message-ID: Hello all, This is my first message on the list, i?m pretty new to FS. I was playing a bit with mod_lcr and found that the sql query for fetching the lowest rate can be changed to a better use of indexes, at least on mysql. Anyone can do some test using other DBs? The change i've made was simple, the original query was something about "... AND digits IN (12345, 1234, 123, 12, 1) ..." and using EXPLAIN i saw that it was using carrier_id as key for the biggest table and not digits. I've changed the code so the query is " AND (digits='12345' OR digits='1234' OR digits='123' OR digits='12' OR digits='1') " and mysql uses the index from the digits row, reducing the returned resultset of the subquery from all the digits from a carrier to the number of "OR" in the query (in my case, from 19850+ to 14). Anyone think that this may be a nice change? or it is just a bad use of indexes by mysql? Regards, Marcelo Sosa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/7a8e601c/attachment.html From msc at freeswitch.org Thu Sep 17 23:44:53 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 17 Sep 2009 23:44:53 -0700 Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DCBE4FC6@mse17be1.mse17.exchange.ms> References: <367751820909171437r6530828ayee5d8cc392c2c7fd@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DCBE4F5D@mse17be1.mse17.exchange.ms> <7b197bef0909171627p3c4bc8a2x4068c239eebd0f52@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DCBE4FC6@mse17be1.mse17.exchange.ms> Message-ID: <87f2f3b90909172344y7b762235h657806dbc26ba7b6@mail.gmail.com> On Thu, Sep 17, 2009 at 5:53 PM, Michael Giagnocavo wrote: > Oh, weird. Seems to work in other languages. > > Yet another reason to use Lua instead of JS. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090917/7b407511/attachment.html From shahzad at vopium.com Fri Sep 18 00:18:42 2009 From: shahzad at vopium.com (Muhammad Shahzad) Date: Fri, 18 Sep 2009 13:18:42 +0600 Subject: [Freeswitch-users] FreeSWITCH Documentation Message-ID: Hi, I have observed that one of the major hurdle while writing patches and / or bug fixes is lack of doxygen documentation for FS source code. For example it took me 5+ days to understand mod_dingaling code and its hooks into FS source code to write up soft reload patch, while it could have taken less then 3 days to do so if source code documentation was available. So, since right now i have some human resources including myself available, I would like to document all source code (or at least core FS code i.e. everything that has "switch_" prefix) using doxygen. I know its a huge task and will take a while to complete but at least lets get it started. If anyone else wants to participate as well in this task, then we can team up to complete it quickly. Let me know if you guys are interested. Thank you. -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/322eb73b/attachment.html From mgg at giagnocavo.net Fri Sep 18 00:18:38 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 18 Sep 2009 03:18:38 -0400 Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! In-Reply-To: <87f2f3b90909172344y7b762235h657806dbc26ba7b6@mail.gmail.com> References: <367751820909171437r6530828ayee5d8cc392c2c7fd@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DCBE4F5D@mse17be1.mse17.exchange.ms> <7b197bef0909171627p3c4bc8a2x4068c239eebd0f52@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DCBE4FC6@mse17be1.mse17.exchange.ms> <87f2f3b90909172344y7b762235h657806dbc26ba7b6@mail.gmail.com> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DCBE4FFD@mse17be1.mse17.exchange.ms> >Yet another reason to use Lua F# instead of JS. :) Fixed that for ya From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Friday, September 18, 2009 12:45 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! On Thu, Sep 17, 2009 at 5:53 PM, Michael Giagnocavo > wrote: Oh, weird. Seems to work in other languages. Yet another reason to use Lua instead of JS. :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/fecac47d/attachment-0001.html From mustafa.pk at gmail.com Fri Sep 18 00:42:27 2009 From: mustafa.pk at gmail.com (Ghulam Mustafa) Date: Fri, 18 Sep 2009 13:42:27 +0600 Subject: [Freeswitch-users] FreeSWITCH Documentation In-Reply-To: References: Message-ID: <4AB339E3.4040705@gmail.com> that's what i was thinking about, but you have taken it a mile ahead :) i would appreciate if you could help people like me to understand the *switch* API without spending sooo much time. would love examples along with the documentation. :| -mustafa -- Ghulam Mustafa cell: +92 333.611.7681 sip: cyrenity at ekiga.net mail: mustafa.pk at gmail.com web: cyrenity.wordpress.com Muhammad Shahzad wrote: > Hi, > > I have observed that one of the major hurdle while writing patches and > / or bug fixes is lack of doxygen documentation for FS source code. > For example it took me 5+ days to understand mod_dingaling code and > its hooks into FS source code to write up soft reload patch, while it > could have taken less then 3 days to do so if source code > documentation was available. > > So, since right now i have some human resources including myself > available, I would like to document all source code (or at least core > FS code i.e. everything that has "switch_" prefix) using doxygen. I > know its a huge task and will take a while to complete but at least > lets get it started. > > If anyone else wants to participate as well in this task, then we can > team up to complete it quickly. > > Let me know if you guys are interested. > > Thank you. > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Fri Sep 18 00:53:26 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Fri, 18 Sep 2009 09:53:26 +0200 Subject: [Freeswitch-users] Hangup: Always the same Q.850 cause code Message-ID: <4AB33C76.5050702@gmx.net> Hello , I try to hangup aa call with a certain cause code. If the user dials a number which we currently do not serve we send which gives a "SIP/2.0 480 Temporarily Unavailable." Message , which is fine. For the target number being busy or having another state, we use this. which gives a "SIP/2.0 486 Busy Here."" Message , which is fine in case of Busy. However in both cases the SIP mssage has the following cause code: Reason: Q.850;cause=16;text="NORMAL_CLEARING". which can lead to problems when forwarding to a PSTN Gateway. How can we achieve, that the cause code is in sync with the deiivered message? We are on Trunk 14741M. Best regards Peter From sias at cpdata.co.za Fri Sep 18 01:41:19 2009 From: sias at cpdata.co.za (Sias Mey) Date: Fri, 18 Sep 2009 10:41:19 +0200 Subject: [Freeswitch-users] Delay when transferring call In-Reply-To: <87f2f3b90909171040n1844d4ccu44d070a785cc0d6a@mail.gmail.com> References: <20090917102350.GA15903@sias-laptop.cpdata.co.za> <87f2f3b90909171040n1844d4ccu44d070a785cc0d6a@mail.gmail.com> Message-ID: <20090918084119.GA32017@sias-laptop.cpdata.co.za> Indeed the debug info did shed some light. It seems the normal output was hiding a secret from me ;-) Even though the transfer had apparently already been triggered, it seems the problem was that the js wasent exiting till the nigling Error in my_thread_global_end() line. 2009-09-18 10:33:49.190769 [DEBUG] switch_odbc.c:210 Connected to [lucidt] 2009-09-18 10:33:49.193641 [INFO] regin.js:1 Starting query run now!!! 2009-09-18 10:33:49.227567 [DEBUG] mod_conference.c:2407 Setup timer success interval: 20 samples: 160 2009-09-18 10:33:49.252869 [INFO] regin.js:1 Finished query run now!!! 2009-09-18 10:33:49.608559 [DEBUG] switch_core_io.c:649 sofia/internal/sip:1004 at 192.168.250.97:5060 receive message [TRANSCODING_NECESSARY] 2009-09-18 10:33:51.909401 [DEBUG] mod_local_stream.c:346 Opening Stream [moh/8000] 8000hz >>>>>> Error in my_thread_global_end(): 1 threads didn't exit <<<<<< 2009-09-18 10:33:54.255485 [DEBUG] switch_odbc.c:119 Disconnected 0 from [lucidt] 2009-09-18 10:33:54.255485 [INFO] regin.js:1 Finished updateing call I know at least one other poster has asked about that before. It seems to be an error related to accesing mysql InnoDB via odbc. Something about not decreasing the thread count before closing the connection. Unfortunately the only information I can get on fixing the bug says replace you PHP mysql.dll with and older version and restart IIS??? .... Not so usefull in my case of running in linux and not using PHP. I have tried adding a db.close to all my scripts but that doesent seem to help either. So, if any of the rest of you have come across this and know how to fix in if im not using php and IIS, some help would be greatly appreciated. I think in the mean time I will try to recompile my mysql odbc lib (I installed from the package manager initially). Im not going to post a bug to freeswitch since this seems to be a mysql odbc related issue, but thankyou for the help in tracking it down to that. Cheers, Sias On Thu, Sep 17, 2009 at 10:40:21AM -0700, Michael Collins wrote: > Turn on debug (press F8) level logging and capture the output, put in > [1]pastebin.freeswitch.org. Hopefully the debug output will shed some > light on where the delay is occurring. > Also, see this page for some tips on how to collect information for > debugging purposes: > [2]http://wiki.freeswitch.org/wiki/Reporting_Bugs > It will give you handy tips on collecting information, posting to > pastebin, asking community for help, etc. etc. In short, it will make > your life easier. :) > -MC > > On Thu, Sep 17, 2009 at 3:23 AM, Sias Mey <[3]sias at cpdata.co.za> wrote: > > Hi, > Im having a strange issue with a api triggered call transfer. > There seems to be a long delay between when the transfer is > triggered and when it actually happens. > 2009-09-17 11:36:26.995001 [NOTICE] switch_ivr.c:1350 Transfer > sofia/internal/[4]1004 at 192.168.0.10 to xml > [incust-camp=lucidlive-call=78-conf=41 at default] > Error in my_thread_global_end(): 26 threads didn't exit > 2009-09-17 11:36:31.997191 [INFO] mod_dialplan_xml.c:315 Processing > 1004->incust-camp=lucidlive-call=7 > 8-conf=41 in context default > 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Execution start > 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Connecting to Ringback > to add call > 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Finished adding calls > 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Connecting to database > lucidlive to update call > 2009-09-17 11:36:32.37433 [INFO] regin.js:1 Finished updateing call > I though it was my regin.js script causing the delay since it runs a > couple of database queries and other things, but the output above > show that runs fine. > My question is about the delay between 11:36:26 -> 11:36:31. The > call is being transfered out of a fifo, but for those 5 seconds > there is no MOH or anything else. Just silence. > The transfer is triggered via a xml rpc call. But since the delay is > between the switch_ivr and mod_dialplan_xml somewhere I doubt that > that has much to do with it. > Any clues or other places I can go look?. > Cheers, > Sias > _______________________________________________ > FreeSWITCH-users mailing list > [5]FreeSWITCH-users at lists.freeswitch.org > [6]http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:[7]http://lists.freeswitch.org/mailman/options/freeswitc > h-users > [8]http://www.freeswitch.org > > References > > 1. http://pastebin.freeswitch.org/ > 2. http://wiki.freeswitch.org/wiki/Reporting_Bugs > 3. mailto:sias at cpdata.co.za > 4. mailto:1004 at 192.168.0.10 > 5. mailto:FreeSWITCH-users at lists.freeswitch.org > 6. http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > 7. http://lists.freeswitch.org/mailman/options/freeswitch-users > 8. http://www.freeswitch.org/ > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From aep.lists at it46.se Fri Sep 18 03:03:12 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Fri, 18 Sep 2009 12:03:12 +0200 Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DCBE4F41@mse17be1.mse17.exchange.ms> References: <6E8D2069C08AA84A83D336E996AE4C6702DCBE4F41@mse17be1.mse17.exchange.ms> Message-ID: <85b1dd14700c0857d483d74807a5c6d2.squirrel@correo.nodo50.org> Thanks for all the tips. I tried to run apiExecute(bgapi, jsrun originate) and still Javascript?? does not let the thread go. No matter the combination of session.hangup(), exit, apiExecute with or without bgapi, the state remains in CS_EXECUTE. So at the end i am triggering an event that i can later use to execute a originate callback. It is nicer with ESL but i still think that will be nice to have a real way to expunge a second Javascript and let the first one die. The GSM channel/modem needs to be free-free (as I am a serial port-free) to handle the outgoing call. The callback script worked perfect with SIP because it does not care how many sessions are running in parallel. It can always place a call back event the channel is not properly close. /aep -- Stopping junk mailers is good for the environment > So, what happens is that when you are executing an app, the state is > CS_EXECUTE. Even if the session is hungup, the state machine doesn't go > through all the hangup code until your app executes. > > The easiest workaround is probably to start a background api (bgapi?) call > to a script. This will happen on another thread, then allow your current > thread to execute and the hangup code will execute. This should work just > fine, I think. (You can stop reading here.) > > But wait, there's even more fun! anthm recently checked in a change a > couple days that lets you work around this. Don't call destroy, call > hangup on the session, on that session's thread. This will perform a > hangup, then progress the state machine. Then the session will truly be > hungup. Maybe you need update your freeswitch code, if this is not > happening for you. > > If you updated and hangup still isn't hanging up, you might want to ask > specifically about that. Or, you may need to call > switch_core_session_hangup_state directly -- just hangup alone might not > do the trick. This is a C function, and not exposed to languages by > default - you can either patch javascript plugin to expose this safely > (and I have no idea what this means for the javascript runtime), or use a > more capable plugin like mod_managed which _does_ expose all the C > functions, and lets you call in and out of them as you please. > > And now, someone who knows what they're talking about will chime in and > point out what I got wrong. > > Thanks, > -Michael > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Alberto Escudero > Sent: Thursday, September 17, 2009 3:20 PM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does > not free the channel! > > We are trying to create a callback application in Javascript. We get the > callerid from the unanswered call and after destroying the session, we > initiate a callback to the user to conenct it to a local extension in the > dialplan. > > Although we have tried to destroy the first session, or even invoke a > second script using apiExecute("jsrun",dialer.js"), tried session.hangup() > or exit()... the first session does not seem to close properly until the > whole chain of scripts are completed. > > Here is a piece of code that shows the concept (yes!, the sleep function > is far from ideal. CPU loves it! ) > > function sleep(milliseconds) { > var start = new Date().getTime(); > for (var i = 0; i < 1e7; i++) { > if ((new Date().getTime() - start) > milliseconds){ > break; > } > } > } > > if (session.ready()) { > //We catch the caller_id > caller_id_num = session.caller_id_num; > > console_log("Now we got your Caller ID\n"); > > //How long we want to wait to trigger a call back > session.execute("sleep",5000); > > console_log("We have waited a while... time to create the > callback\n"); > > //apiExecute("jsrun", "callback.js"); > } > > //Destroy the session... > session.destroy(); > session=undefined; > > sleep(10000); > > //Preparing callback > session2 = new > Session('{ignore_early_media=true}celliax/interface1/600464646'); > session2.setAutoHangup(false); > session2.answer(); > exit(); > > ++ > Wisdom thoughts? > > -- > Stopping junk mailers is good for the environment > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From pankajanand18 at gmail.com Fri Sep 18 03:13:20 2009 From: pankajanand18 at gmail.com (pankaj anand) Date: Fri, 18 Sep 2009 15:43:20 +0530 Subject: [Freeswitch-users] Not able to make call using external profile Message-ID: <809ad7ab0909180313n5538c1c0ve1b8070c0d1da2cc@mail.gmail.com> hi folks, I m not able to make SIP calls using external profile. i have added the following lines to the $installdir/conf/dialplan/public.xml I m able to connect using 1000 and 1001 from public Internet. I am able to make an echo call. *when i type :* $: sofia status profile external reg It shows the list of the connected clients and their information. but when I m trying to make a call from one user to other user, it generates the following error 2009-09-18 15:41:45.675054 [NOTICE] switch_channel.c:602 New Channel sofia/external/1001 at 192.168.1.50 [fcb6c23e-bdcd-41dd-b73e-df07b71252be] 2009-09-18 15:41:45.677063 [INFO] mod_dialplan_xml.c:315 Processing 1000->1000 in context public 2009-09-18 15:41:45.679071 [NOTICE] switch_channel.c:602 New Channel sofia/external/1000 at 192.168.1.50 [1a537865-be53-42ce-b8f5-cc183f4f1306] 2009-09-18 15:41:45.688161 [ERR] sofia_reg.c:1568 No Matching gateway found 2009-09-18 15:41:45.688161 [NOTICE] sofia_reg.c:1588 Hangup sofia/external/ 1000 at 192.168.1.50 [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] 2009-09-18 15:41:45.688161 [INFO] mod_dptools.c:2093 Originate Failed. Cause: MANDATORY_IE_MISSING 2009-09-18 15:41:45.689090 [NOTICE] mod_dptools.c:2125 Hangup sofia/external/1001 at 192.168.1.50 [CS_EXECUTE] [MANDATORY_IE_MISSING] 2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/external/1001 at 192.168.1.50) Ended 2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1001 at 192.168.1.50 [CS_DESTROY] 2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/external/1000 at 192.168.1.50) Ended 2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/1000 at 192.168.1.50 [CS_DESTROY] with regards Pankaj anand -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/cbfbbd09/attachment.html From brian at freeswitch.org Fri Sep 18 06:53:54 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Sep 2009 08:53:54 -0500 Subject: [Freeswitch-users] Not able to make call using external profile In-Reply-To: <809ad7ab0909180313n5538c1c0ve1b8070c0d1da2cc@mail.gmail.com> References: <809ad7ab0909180313n5538c1c0ve1b8070c0d1da2cc@mail.gmail.com> Message-ID: <08D904F9-E064-4BE9-88C1-0D9DB73BC466@freeswitch.org> The far end is sending you a challenge and we can't answer it because we haven't been told what gateway to use. /b On Sep 18, 2009, at 5:13 AM, pankaj anand wrote: > hi folks, > I m not able to make SIP calls using external profile. > From rupa at rupa.com Fri Sep 18 06:57:11 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 18 Sep 2009 08:57:11 -0500 Subject: [Freeswitch-users] Build problems with Shoutcast module under Debian In-Reply-To: <20090918025535.GA19122@jdc.jasonjgw.net> References: <20090918020726.GA3242@jdc.jasonjgw.net> <20090918025535.GA19122@jdc.jasonjgw.net> Message-ID: did you try to rerun configure? I think that'll fix it for you. On Thu, Sep 17, 2009 at 9:55 PM, Jason White wrote: > It turns out that Debian recently removed the libogg.la file, deliberately, > from the package. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From aep.lists at it46.se Fri Sep 18 07:00:06 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Fri, 18 Sep 2009 16:00:06 +0200 Subject: [Freeswitch-users] Not able to make call using external profile In-Reply-To: <809ad7ab0909180313n5538c1c0ve1b8070c0d1da2cc@mail.gmail.com> References: <809ad7ab0909180313n5538c1c0ve1b8070c0d1da2cc@mail.gmail.com> Message-ID: <14164e5b2c45e6b563bf2045223df317.squirrel@correo.nodo50.org> Have you tried with instead? /aep -- Stopping junk mailers is good for the environment > hi folks, I m not able to make SIP calls using external profile. > > i have added the following lines to the > $installdir/conf/dialplan/public.xml > > > > > > > > > > > > > > > I m able to connect using 1000 and 1001 from public Internet. I am able > to > make an echo call. > > *when i type :* > > $: sofia status profile external reg > > It shows the list of the connected clients and their information. > > but when I m trying to make a call from one user to other user, it > generates > the following error > > > 2009-09-18 15:41:45.675054 [NOTICE] switch_channel.c:602 New Channel > sofia/external/1001 at 192.168.1.50 [fcb6c23e-bdcd-41dd-b73e-df07b71252be] > 2009-09-18 15:41:45.677063 [INFO] mod_dialplan_xml.c:315 Processing > 1000->1000 in context public > 2009-09-18 15:41:45.679071 [NOTICE] switch_channel.c:602 New Channel > sofia/external/1000 at 192.168.1.50 [1a537865-be53-42ce-b8f5-cc183f4f1306] > 2009-09-18 15:41:45.688161 [ERR] sofia_reg.c:1568 No Matching gateway > found > 2009-09-18 15:41:45.688161 [NOTICE] sofia_reg.c:1588 Hangup > sofia/external/ > 1000 at 192.168.1.50 [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] > 2009-09-18 15:41:45.688161 [INFO] mod_dptools.c:2093 Originate Failed. > Cause: MANDATORY_IE_MISSING > 2009-09-18 15:41:45.689090 [NOTICE] mod_dptools.c:2125 Hangup > sofia/external/1001 at 192.168.1.50 [CS_EXECUTE] [MANDATORY_IE_MISSING] > 2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1086 Session 1 > (sofia/external/1001 at 192.168.1.50) Ended > 2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1088 Close > Channel > sofia/external/1001 at 192.168.1.50 [CS_DESTROY] > 2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1086 Session 2 > (sofia/external/1000 at 192.168.1.50) Ended > 2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1088 Close > Channel > sofia/external/1000 at 192.168.1.50 [CS_DESTROY] > > > with regards > Pankaj anand > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Fri Sep 18 07:01:42 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Sep 2009 09:01:42 -0500 Subject: [Freeswitch-users] Simple call waiting question In-Reply-To: <33c87fa30909171715v439a14d1x2f6ff8675128b0f5@mail.gmail.com> References: <33c87fa30909170558v4b032da2w6f8202b187d3bcf6@mail.gmail.com> <33c87fa30909171715v439a14d1x2f6ff8675128b0f5@mail.gmail.com> Message-ID: <191c3a030909180701y67b45e62x5dbed0a52635545f@mail.gmail.com> This feature is handled by the phone itself. (remember this is digital phone not analog) When C calls A the phone should indicate to you that there is a 2nd call, you press flash and put B on hold and talk to C when you press flash again it either goes back and forth between b and C, conferences them or hangs up on them all depending on the phone. The phone is doing all the work in this case. Trying to do it with FreeSWITCH is over-complicating the situation and I don't think we even have a way to emulate that behaviour. On Thu, Sep 17, 2009 at 7:15 PM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Thanks Brian, (I bought a dud phone.. and its a new DECT! - crazy) > > I am using the 5900 and 5901 for parking/unparking. That > functionality works fine and I can park/unpark the B leg as I wish. > > The problem is that if I park the B-leg, the A-leg then gets a busy > signal. If the A leg is then hung up, a user-busy signal is sent to > the C-leg, so the call goes to voicemail. > > What I want to happen is park B and answer C directly. > > Is this possible? > > > On Thu, Sep 17, 2009 at 11:21 PM, Brian West wrote: > > Personally I would throw the phone in the trash. :P > > > > In the default dialplan look at 5900 for park and 5901 for unpark. > > > > /b > > > > > > On Sep 17, 2009, at 7:58 AM, Mark Campbell-Smith wrote: > > > >> I am trying to create a simple call waiting dialplan as my phone does > >> not have Recall button. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/1fe99383/attachment.html From rupa at rupa.com Fri Sep 18 07:07:14 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 18 Sep 2009 09:07:14 -0500 Subject: [Freeswitch-users] mod_lcr and indexes In-Reply-To: References: Message-ID: Hmm.... This is because mysql is "dumb" :( Anyway, if you wanted quoted digits, there is an option to enable that in the mod_lcr config file. http://wiki.freeswitch.org/wiki/Mod_lcr#Advanced_Usage Specifically, look at the parameter: quote_in_list The most efficient way (that I know of) to use mod_lcr is to use postgresql and the prefix postgres module which uses a custom datatype and a GIST index for the prefix column. On Thu, Sep 17, 2009 at 9:24 PM, Marcelo Sosa - LST wrote: > Hello all, > > This is my first message on the list, i?m pretty new to FS. > I was playing a bit with mod_lcr and found that the sql query for fetching > the lowest rate can be changed to a better use of indexes, at least on > mysql. Anyone can do some test using other DBs? > > The change i've made was simple, the original query was something about "... > AND digits IN (12345, 1234, 123, 12, 1) ..." and using EXPLAIN i saw that it > was using carrier_id as key for the biggest table and not digits. I've > changed the code so the query is " AND (digits='12345' OR digits='1234' OR > digits='123' OR digits='12' OR digits='1') " and mysql uses the index from > the digits row, reducing the returned resultset of the subquery from all the > digits from a carrier to the number of "OR" in the query (in my case, from > 19850+ to 14). > > Anyone think that this may be a nice change? or it is just a bad use of > indexes by mysql? > > Regards, > Marcelo Sosa > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From pankajanand18 at gmail.com Fri Sep 18 07:11:11 2009 From: pankajanand18 at gmail.com (pankaj anand) Date: Fri, 18 Sep 2009 19:41:11 +0530 Subject: [Freeswitch-users] Not able to make call using external profile In-Reply-To: <809ad7ab0909180313n5538c1c0ve1b8070c0d1da2cc@mail.gmail.com> References: <809ad7ab0909180313n5538c1c0ve1b8070c0d1da2cc@mail.gmail.com> Message-ID: <809ad7ab0909180711v73f15899v8e9dc64cb191e82c@mail.gmail.com> I m using default configuration of freeswitch.. I m not using any gateway for authentication. in the $INSTALLDIR/conf/sip_profiles/external/ directory, there exist only one file which example.xml , this files contains as you can see, all the lines are commented. So i m not using any gateways. On Fri, Sep 18, 2009 at 3:43 PM, pankaj anand wrote: > hi folks, I m not able to make SIP calls using external profile. > > i have added the following lines to the > $installdir/conf/dialplan/public.xml > > > > > > > > > > > > > > > I m able to connect using 1000 and 1001 from public Internet. I am able to > make an echo call. > > *when i type :* > > $: sofia status profile external reg > > It shows the list of the connected clients and their information. > > but when I m trying to make a call from one user to other user, it > generates the following error > > > 2009-09-18 15:41:45.675054 [NOTICE] switch_channel.c:602 New Channel > sofia/external/1001 at 192.168.1.50 [fcb6c23e-bdcd-41dd-b73e-df07b71252be] > 2009-09-18 15:41:45.677063 [INFO] mod_dialplan_xml.c:315 Processing > 1000->1000 in context public > 2009-09-18 15:41:45.679071 [NOTICE] switch_channel.c:602 New Channel > sofia/external/1000 at 192.168.1.50 [1a537865-be53-42ce-b8f5-cc183f4f1306] > 2009-09-18 15:41:45.688161 [ERR] sofia_reg.c:1568 No Matching gateway found > 2009-09-18 15:41:45.688161 [NOTICE] sofia_reg.c:1588 Hangup sofia/external/ > 1000 at 192.168.1.50 [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] > 2009-09-18 15:41:45.688161 [INFO] mod_dptools.c:2093 Originate Failed. > Cause: MANDATORY_IE_MISSING > 2009-09-18 15:41:45.689090 [NOTICE] mod_dptools.c:2125 Hangup > sofia/external/1001 at 192.168.1.50 [CS_EXECUTE] [MANDATORY_IE_MISSING] > 2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1086 Session 1 > (sofia/external/1001 at 192.168.1.50) Ended > 2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/external/1001 at 192.168.1.50 [CS_DESTROY] > 2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1086 Session 2 > (sofia/external/1000 at 192.168.1.50) Ended > 2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/external/1000 at 192.168.1.50 [CS_DESTROY] > > > with regards > Pankaj anand > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/87de407c/attachment-0001.html From brian at freeswitch.org Fri Sep 18 07:15:26 2009 From: brian at freeswitch.org (Brian West) Date: Fri, 18 Sep 2009 09:15:26 -0500 Subject: [Freeswitch-users] Not able to make call using external profile In-Reply-To: <809ad7ab0909180711v73f15899v8e9dc64cb191e82c@mail.gmail.com> References: <809ad7ab0909180313n5538c1c0ve1b8070c0d1da2cc@mail.gmail.com> <809ad7ab0909180711v73f15899v8e9dc64cb191e82c@mail.gmail.com> Message-ID: <7F516C2A-5410-4E9E-9AA3-BEB3C1AFCFE9@freeswitch.org> OK pay attention this time. See this line: 2009-09-18 15:41:45.688161 [ERR] sofia_reg.c:1568 No Matching gateway found You sent a call out the profile the far side sent you a challenge since you're not calling via a gateway we can't answer the challenge because we do not know HOW. What is the far end you're calling? /b On Sep 18, 2009, at 9:11 AM, pankaj anand wrote: > I m using default configuration of freeswitch.. I m not using any > gateway for authentication. > > in the $INSTALLDIR/conf/sip_profiles/external/ directory, there > exist only one file which example.xml , this files contains From martin at epbx.cz Fri Sep 18 02:40:41 2009 From: martin at epbx.cz (Martin Dvorak) Date: Fri, 18 Sep 2009 11:40:41 +0200 Subject: [Freeswitch-users] DB table sip_dialogs is always empty Message-ID: <4AB35599.8020709@epbx.cz> Hello, let me apologize for long reply delay, but my f***** mail server "eat" nearly all messages from FS mailing list (I have received just three e-mails in the past five days...). So, lets try another one :-) Now back to the thema: Thank you very much, Brian, Anthony and Michael for your answers! If I catch it right, the recommended way to get current calls info is to use "show channels" command. Am I right? BTW: If this kind of info is stored in the internal SQLite, isn't be useful to make it configurable => SQLite x ODBC? Thanks once more! Best regards kokoska.rokoska anthm wrote: that table is specific to the manage-presence option On Thu, Sep 17, 2009 at 6:08 AM, kokoska.rokoska wrote: > I'm sorry to resend this post, but even after few hours I can't see it > in the mailing-list... > Thanks. > > > Best regards, > > kokoska.rokoska > > > kokoska rokoska napsal(a): > > Hello, > > > > I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and based > > on logs FS connected to the dsns correctly. But during the calls I can't > > see any rows in sip_dialogs table. When I run "show channels" from > > console it works fine. > > All other tables are "populated and maintained" as I expect :-) > > > > I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB... > > > > Could you, please, point what I'm missing, or where I'm wrong? Thank you > > very much! > > > > > > Best regards, > > > > kokoska.rokoska > > > From lyncker at lyth.de Fri Sep 18 07:22:00 2009 From: lyncker at lyth.de (Filip Lyncker) Date: Fri, 18 Sep 2009 16:22:00 +0200 Subject: [Freeswitch-users] Some Newbie questions about dialplan and local Sip registration Message-ID: <4AB39788.3000907@lyth.de> Hi List, for the first experiments with freeswitch I downloaded the Windows installation. Now Im trying to get my 2 Sipphones get connected to. Later I want connect the freeswitch to my asterisk gateway. I find the examples pretty complex therfore Im trying to build up a simple solution to understand the functions from the scratch .. my current problem is , that I cant route my local sips to each other ( registration seems to work now). the next is , that freeshwitch is not able to connect to asterisk. but I will describe this later. I installed in the Directory a xml file ( called 22.xml) with the following content : This seems to be ok now. Now I want to dial from 22 to 24 , wherefore I configured this dialplan : wich doesnt work , mybe b/c the user/${dialed_extension} I dont know... Freeswitch says: [INFO] switch_core_state_machine.c:136 No Route, Aborting [NOTICE] switch_core_state_machine.c:137 Hangup sofia/internal/24 at 192.168.1.34 [CS_ROUTING] [NO_ROUTE_DESTINATION] [NOTICE] switch_core_session.c:1086 Session 17 (sofia/internal/24 at 192.168.1.34) Ended [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/24 at 192.168.1.34 [CS_DESTROY] Im sure , for you guys this cant be a big deal;) Next Point is my Asterisk registration , mybe you can help me out here to .. : In the sip-profiles/external I installed the my_asterisk.xml with that content : Freeswitch allways complains a timeout error : [ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request Timeout [408]. failure #17 [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry to 540 seconds. it seems that It cant connect ( I also tried out to explicit set the port to 5060 b/c I read something about 5080 .. : but this didnt help) In my Asterisk I set in the sip.conf the entry 28 with the pw test .... If someone could help me with my first steps I would be verrry thankful ;)) cheers Filip -- _________________________________ Filip Lyncker, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Gesch?ftsf?hrer: Filip Lyncker, Armin Theis From anthony.minessale at gmail.com Fri Sep 18 07:46:16 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Sep 2009 09:46:16 -0500 Subject: [Freeswitch-users] Hangup: Always the same Q.850 cause code In-Reply-To: <4AB33C76.5050702@gmx.net> References: <4AB33C76.5050702@gmx.net> Message-ID: <191c3a030909180746t326e06aakef4e26a0945de195@mail.gmail.com> Here is what I get when I test it. you may want to look at your console for the blue Hangup lines and confirm it's your call to hangup send 644 bytes to udp/[72.128.89.126]:42988 at 14:34:29.043915: ------------------------------------------------------------------------ SIP/2.0 480 Temporarily Unavailable Via: SIP/2.0/UDP 10.0.1.8:50606 ;branch=z9hG4bK-d8754z-6077f64f00ed157e-1---d8754z-;rport=42988;received=72.128.89.126 From: "tony" >;tag=352a2a46 To: "7016" >;tag=HUye00UQZKySQ Call-ID: N2Y1MWYwZjA2YzJlY2ZhY2VjYzRhNDZmMzczYWMwN2Q. CSeq: 1 INVITE User-Agent: The Guy In IRC IS WRONG Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=19;text="NO_ANSWER" Content-Length: 0 send 630 bytes to udp/[72.128.89.126]:42988 at 14:35:31.286436: ------------------------------------------------------------------------ SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 10.0.1.8:50606 ;branch=z9hG4bK-d8754z-223ae00e1829097e-1---d8754z-;rport=42988;received=72.128.89.126 From: "tony" >;tag=aa3b2b1d To: "7016" >;tag=j4Q71UcUvvmcK Call-ID: NDcyNmQyYjY5YWQwOTI3MjZiZWFlZDQyNDIyZjZlMDA. CSeq: 1 INVITE User-Agent: The Guy In IRC IS WRONG Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, refer Reason: Q.850;cause=17;text="USER_BUSY" Content-Length: 0 On Fri, Sep 18, 2009 at 2:53 AM, Peter P GMX wrote: > Hello , > > I try to hangup aa call with a certain cause code. > > If the user dials a number which we currently do not serve we send > > > which gives a > "SIP/2.0 480 Temporarily Unavailable." Message , which is fine. > > For the target number being busy or having another state, we use this. > > > which gives a > "SIP/2.0 486 Busy Here."" Message , which is fine in case of Busy. > > However in both cases the SIP mssage has the following cause code: > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > which can lead to problems when forwarding to a PSTN Gateway. > > How can we achieve, that the cause code is in sync with the deiivered > message? > > We are on Trunk 14741M. > > Best regards > Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/2c13f5d3/attachment.html From listas at askterisk.com Fri Sep 18 07:47:21 2009 From: listas at askterisk.com (Marcelo Sosa - LST) Date: Fri, 18 Sep 2009 11:47:21 -0300 Subject: [Freeswitch-users] mod_lcr and indexes References: Message-ID: Hello, I know that mysql is dumb, but i?m postgresql-fobic :) (i was happy using pgsql, until i found a bug in the restore of backups, that makes the backup unable to be restored, very bad day) :-) Anyway, I was refering to the change from "digits IN (a list of digits)" to "(digits='xx' OR digits='xxx')", not the quote_in_list option that i found when i was touching a bit of code. For the list-archives then, mysql may prefer a different query format to speed up lcr matches. I?ll test with postgres and check it has any differences by using one method or another, if not may be we can change the code so it uses the fastest way for mysql. Regards, Marcelo Sosa ----- Original Message ----- From: "Rupa Schomaker" To: Sent: Friday, September 18, 2009 11:07 AM Subject: Re: [Freeswitch-users] mod_lcr and indexes > Hmm.... This is because mysql is "dumb" :( Anyway, if you wanted > quoted digits, there is an option to enable that in the mod_lcr config > file. > > http://wiki.freeswitch.org/wiki/Mod_lcr#Advanced_Usage > > Specifically, look at the parameter: quote_in_list > > The most efficient way (that I know of) to use mod_lcr is to use > postgresql and the prefix postgres module which uses a custom datatype > and a GIST index for the prefix column. > > On Thu, Sep 17, 2009 at 9:24 PM, Marcelo Sosa - LST > wrote: >> Hello all, >> >> This is my first message on the list, i?m pretty new to FS. >> I was playing a bit with mod_lcr and found that the sql query for >> fetching >> the lowest rate can be changed to a better use of indexes, at least on >> mysql. Anyone can do some test using other DBs? >> >> The change i've made was simple, the original query was something about >> "... >> AND digits IN (12345, 1234, 123, 12, 1) ..." and using EXPLAIN i saw that >> it >> was using carrier_id as key for the biggest table and not digits. I've >> changed the code so the query is " AND (digits='12345' OR digits='1234' >> OR >> digits='123' OR digits='12' OR digits='1') " and mysql uses the index >> from >> the digits row, reducing the returned resultset of the subquery from all >> the >> digits from a carrier to the number of "OR" in the query (in my case, >> from >> 19850+ to 14). >> >> Anyone think that this may be a nice change? or it is just a bad use of >> indexes by mysql? >> >> Regards, >> Marcelo Sosa >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From anthony.minessale at gmail.com Fri Sep 18 07:53:08 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 18 Sep 2009 09:53:08 -0500 Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! In-Reply-To: <85b1dd14700c0857d483d74807a5c6d2.squirrel@correo.nodo50.org> References: <6E8D2069C08AA84A83D336E996AE4C6702DCBE4F41@mse17be1.mse17.exchange.ms> <85b1dd14700c0857d483d74807a5c6d2.squirrel@correo.nodo50.org> Message-ID: <191c3a030909180753h77337721r2066c49b7f403d0a@mail.gmail.com> You could put an api_hangup_hook on the channel to jsrun your script. What you want with javascript is not going to happen as long as you execute the script *WITH* the channel. it's not a problem it's just misuse/misunderstanding on your part. On Fri, Sep 18, 2009 at 5:03 AM, Alberto Escudero wrote: > Thanks for all the tips. I tried to run apiExecute(bgapi, jsrun originate) > and still Javascript?? does not let the thread go. > > No matter the combination of session.hangup(), exit, apiExecute with or > without bgapi, the state remains in CS_EXECUTE. > > So at the end i am triggering an event that i can later use to execute a > originate callback. It is nicer with ESL but i still think that will be > nice to have a real way to expunge a second Javascript and let the first > one die. > > The GSM channel/modem needs to be free-free (as I am a serial port-free) > to handle the outgoing call. The callback script worked perfect with SIP > because it does not care how many sessions are running in parallel. It can > always place a call back event the channel is not properly close. > > /aep > > > -- > Stopping junk mailers is good for the environment > > > So, what happens is that when you are executing an app, the state is > > CS_EXECUTE. Even if the session is hungup, the state machine doesn't go > > through all the hangup code until your app executes. > > > > The easiest workaround is probably to start a background api (bgapi?) > call > > to a script. This will happen on another thread, then allow your current > > thread to execute and the hangup code will execute. This should work just > > fine, I think. (You can stop reading here.) > > > > But wait, there's even more fun! anthm recently checked in a change a > > couple days that lets you work around this. Don't call destroy, call > > hangup on the session, on that session's thread. This will perform a > > hangup, then progress the state machine. Then the session will truly be > > hungup. Maybe you need update your freeswitch code, if this is not > > happening for you. > > > > If you updated and hangup still isn't hanging up, you might want to ask > > specifically about that. Or, you may need to call > > switch_core_session_hangup_state directly -- just hangup alone might not > > do the trick. This is a C function, and not exposed to languages by > > default - you can either patch javascript plugin to expose this safely > > (and I have no idea what this means for the javascript runtime), or use a > > more capable plugin like mod_managed which _does_ expose all the C > > functions, and lets you call in and out of them as you please. > > > > And now, someone who knows what they're talking about will chime in and > > point out what I got wrong. > > > > Thanks, > > -Michael > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Alberto Escudero > > Sent: Thursday, September 17, 2009 3:20 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: [Freeswitch-users] Callback in Javascript, session.destroy() > does > > not free the channel! > > > > We are trying to create a callback application in Javascript. We get the > > callerid from the unanswered call and after destroying the session, we > > initiate a callback to the user to conenct it to a local extension in the > > dialplan. > > > > Although we have tried to destroy the first session, or even invoke a > > second script using apiExecute("jsrun",dialer.js"), tried > session.hangup() > > or exit()... the first session does not seem to close properly until the > > whole chain of scripts are completed. > > > > Here is a piece of code that shows the concept (yes!, the sleep function > > is far from ideal. CPU loves it! ) > > > > function sleep(milliseconds) { > > var start = new Date().getTime(); > > for (var i = 0; i < 1e7; i++) { > > if ((new Date().getTime() - start) > milliseconds){ > > break; > > } > > } > > } > > > > if (session.ready()) { > > //We catch the caller_id > > caller_id_num = session.caller_id_num; > > > > console_log("Now we got your Caller ID\n"); > > > > //How long we want to wait to trigger a call back > > session.execute("sleep",5000); > > > > console_log("We have waited a while... time to create the > > callback\n"); > > > > //apiExecute("jsrun", "callback.js"); > > } > > > > //Destroy the session... > > session.destroy(); > > session=undefined; > > > > sleep(10000); > > > > //Preparing callback > > session2 = new > > Session('{ignore_early_media=true}celliax/interface1/600464646'); > > session2.setAutoHangup(false); > > session2.answer(); > > exit(); > > > > ++ > > Wisdom thoughts? > > > > -- > > Stopping junk mailers is good for the environment > > > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/c0ac66ec/attachment-0001.html From rupa at rupa.com Fri Sep 18 08:00:26 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 18 Sep 2009 10:00:26 -0500 Subject: [Freeswitch-users] mod_lcr and indexes In-Reply-To: References: Message-ID: On Fri, Sep 18, 2009 at 9:47 AM, Marcelo Sosa - LST wrote: > Hello, > > I know that mysql is dumb, but i?m postgresql-fobic :) (i was happy using > pgsql, until i found a bug in the restore of backups, that makes the backup > unable to be restored, very bad day) :-) huh, really? > Anyway, I was refering to the change from "digits IN (a list of digits)" to > "(digits='xx' OR digits='xxx')", not the quote_in_list option that i found > when i was touching a bit of code. > For the list-archives then, mysql may prefer a different query format to > speed up lcr matches. I?ll test with postgres and check it has any > differences by using one method or another, if not may be we can change the > code so it uses the fastest way for mysql. OR list instead of IN list? What is the size of your rate table? When trying both ways, what is the measured difference in performance? Does mysql have a way to analyze the table to ensure it's statistics are up to date? > > Regards, > Marcelo Sosa > > ----- Original Message ----- > From: "Rupa Schomaker" > To: > Sent: Friday, September 18, 2009 11:07 AM > Subject: Re: [Freeswitch-users] mod_lcr and indexes > > >> Hmm.... ?This is because mysql is "dumb" :( ?Anyway, if you wanted >> quoted digits, there is an option to enable that in the mod_lcr config >> file. >> >> http://wiki.freeswitch.org/wiki/Mod_lcr#Advanced_Usage >> >> Specifically, look at the parameter: quote_in_list >> >> The most efficient way (that I know of) to use mod_lcr is to use >> postgresql and the prefix postgres module which uses a custom datatype >> and a GIST index for the prefix column. >> >> On Thu, Sep 17, 2009 at 9:24 PM, Marcelo Sosa - LST >> wrote: >>> Hello all, >>> >>> This is my first message on the list, i?m pretty new to FS. >>> I was playing a bit with mod_lcr and found that the sql query for >>> fetching >>> the lowest rate can be changed to a better use of indexes, at least on >>> mysql. Anyone can do some test using other DBs? >>> >>> The change i've made was simple, the original query was something about >>> "... >>> AND digits IN (12345, 1234, 123, 12, 1) ..." and using EXPLAIN i saw that >>> it >>> was using carrier_id as key for the biggest table and not digits. I've >>> changed the code so the query is " AND (digits='12345' OR digits='1234' >>> OR >>> digits='123' OR digits='12' OR digits='1') " and mysql uses the index >>> from >>> the digits row, reducing the returned resultset of the subquery from all >>> the >>> digits from a carrier to the number of "OR" in the query (in my case, >>> from >>> 19850+ to 14). >>> >>> Anyone think that this may be a nice change? or it is just a bad use of >>> indexes by mysql? >>> >>> Regards, >>> Marcelo Sosa >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> >> -- >> -Rupa >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From valentin.doroga at pronexus.com Fri Sep 18 08:26:25 2009 From: valentin.doroga at pronexus.com (Valentin Doroga) Date: Fri, 18 Sep 2009 11:26:25 -0400 Subject: [Freeswitch-users] Nokia N800 Message-ID: <20090918152635.UEHC273.tomts27-srv.bellnexxia.net@toip41-bus.srvr.bell.ca> I'm coming again with the same question: does anybody have any instructions how to build FreeSWITCH for Nokia N800? Or, does anybody have a recently built version? Val we relocated the machine with the build env for that, I'll try to find the time to resurrect it and make a new one. On Wed, Apr 1, 2009 at 4:00 PM, Valentin Doroga wrote: There are some old binaries at: http://www.freeswitch.org/downloads/n800/ Is there a newer version? Any place with instruction to build? Val. _______________________________________________ Freeswitch-users mailing list Freeswitch-users at 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/ AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/87b506e2/attachment.html From mike at jerris.com Fri Sep 18 08:36:17 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 18 Sep 2009 11:36:17 -0400 Subject: [Freeswitch-users] Nokia N800 In-Reply-To: <20090918152635.UEHC273.tomts27-srv.bellnexxia.net@toip41-bus.srvr.bell.ca> References: <20090918152635.UEHC273.tomts27-srv.bellnexxia.net@toip41-bus.srvr.bell.ca> Message-ID: <8B0A5CCF-8C21-45AA-8148-86A046A672F4@jerris.com> We don't currently have the build environment setup, but I can tell you it was a pretty basic scratchbox setup and then a normal build just enabling mod_alsa and other appropriate modules. Mike On Sep 18, 2009, at 11:26 AM, Valentin Doroga wrote: > I'm coming again with the same question: does anybody have any > instructions how to build FreeSWITCH for Nokia N800? > Or, does anybody have a recently built version? > Val > > > > we relocated the machine with the build env for that, > I'll try to find the time to resurrect it and make a new one. > > > On Wed, Apr 1, 2009 at 4:00 PM, Valentin Doroga > wrote: > There are some old binaries at: > http://www.freeswitch.org/downloads/n800/ > > Is there a newer version? Any place with instruction to build? > Val. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/075d0783/attachment.html From msc at freeswitch.org Fri Sep 18 08:55:31 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Sep 2009 08:55:31 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call, Sept 18, Starting Shortly Message-ID: <87f2f3b90909180855l6de71215vce8a712e33542246@mail.gmail.com> Get ready to call in! Agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_18 Talk to you all shortly! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/c36a38d8/attachment.html From listas at askterisk.com Fri Sep 18 09:31:24 2009 From: listas at askterisk.com (Marcelo Sosa - LST) Date: Fri, 18 Sep 2009 13:31:24 -0300 Subject: [Freeswitch-users] mod_lcr and indexes References: Message-ID: Hello, >> I know that mysql is dumb, but i?m postgresql-fobic :) (i was happy using >> pgsql, until i found a bug in the restore of backups, that makes the >> backup >> unable to be restored, very bad day) :-) > > huh, really? Bug ID #885 :-) >> Anyway, I was refering to the change from "digits IN (a list of digits)" >> to >> "(digits='xx' OR digits='xxx')", not the quote_in_list option that i >> found >> when i was touching a bit of code. >> For the list-archives then, mysql may prefer a different query format to >> speed up lcr matches. I?ll test with postgres and check it has any >> differences by using one method or another, if not may be we can change >> the >> code so it uses the fastest way for mysql. > > OR list instead of IN list? Yup, kinda weird, isn?t it? IN list should be faster. > What is the size of your rate table? When trying both ways, what is > the measured difference in performance? Does mysql have a way to > analyze the table to ensure it's statistics are up to date? Rate table has about 19k records, i?m just testing. No measured difference with real calls (haven?t tested it), but mysql has a EXPLAIN command that analyzes the query and returns information about it (key used, possible keys, number of rows returned by the subquery, etc). I?m emailing you by private the EXPLAIN query result, i don?t want to clobber this list with mysql problems :-) Regards, Marcelo Sosa From rupa at rupa.com Fri Sep 18 10:35:30 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Fri, 18 Sep 2009 12:35:30 -0500 Subject: [Freeswitch-users] mod_lcr and indexes In-Reply-To: References: Message-ID: On Fri, Sep 18, 2009 at 11:31 AM, Marcelo Sosa - LST wrote: > Hello, > >>> I know that mysql is dumb, but i?m postgresql-fobic :) (i was happy using >>> pgsql, until i found a bug in the restore of backups, that makes the >>> backup >>> unable to be restored, very bad day) :-) >> >> huh, really? > > Bug ID #885 :-) What bug repository. :) >> What is the size of your rate table? ?When trying both ways, what is >> the measured difference in performance? ?Does mysql have a way to >> analyze the table to ensure it's statistics are up to date? > > Rate table has about 19k records, i?m just testing. No measured difference > with real calls (haven?t tested it), but mysql has a EXPLAIN command that > analyzes the query and returns information about it (key used, possible > keys, number of rows returned by the subquery, etc). > I?m emailing you by private the EXPLAIN query result, i don?t want to > clobber this list with mysql problems :-) heh, ok, I replied to your email before seeing this. So you can ignore that part. umm... lets see. We'll keep in private email until we have an answer an followup here. At 19,000 records, that is probably just a few blocks on disk -- not large at all. But we can still try to get this working ok. > > Regards, > Marcelo Sosa > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- -Rupa From tculjaga at gmail.com Fri Sep 18 10:41:08 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 18 Sep 2009 19:41:08 +0200 Subject: [Freeswitch-users] Not able to make call using external profile In-Reply-To: <7F516C2A-5410-4E9E-9AA3-BEB3C1AFCFE9@freeswitch.org> References: <809ad7ab0909180313n5538c1c0ve1b8070c0d1da2cc@mail.gmail.com> <809ad7ab0909180711v73f15899v8e9dc64cb191e82c@mail.gmail.com> <7F516C2A-5410-4E9E-9AA3-BEB3C1AFCFE9@freeswitch.org> Message-ID: <65d96fc80909181041h6c1eb1a1nfef46326f9c57220@mail.gmail.com> in other works, what are you trying to achieve? where do you want send calls? what is 192.168.1.50? where/how are you originating calls from? basically can you please tell us what is your call flow scenario otherwise we can't help? T. On Fri, Sep 18, 2009 at 4:15 PM, Brian West wrote: > OK pay attention this time. > > See this line: > > 2009-09-18 15:41:45.688161 [ERR] sofia_reg.c:1568 No Matching gateway > found > > You sent a call out the profile the far side sent you a challenge > since you're not calling via a gateway we can't answer the challenge > because we do not know HOW. > > What is the far end you're calling? > > /b > > > On Sep 18, 2009, at 9:11 AM, pankaj anand wrote: > > > I m using default configuration of freeswitch.. I m not using any > > gateway for authentication. > > > > in the $INSTALLDIR/conf/sip_profiles/external/ directory, there > > exist only one file which example.xml , this files contains > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/a8e1b52e/attachment-0001.html From tculjaga at gmail.com Fri Sep 18 11:04:24 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 18 Sep 2009 20:04:24 +0200 Subject: [Freeswitch-users] Some Newbie questions about dialplan and local Sip registration In-Reply-To: <4AB39788.3000907@lyth.de> References: <4AB39788.3000907@lyth.de> Message-ID: <65d96fc80909181104v2dc2a9ag268a51cd45c28e6a@mail.gmail.com> hi Filip, for calling a user... please read this first: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User for making a GW register into e.g. asterisk please use this: this should be enough to register the GW... after that please read this: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways in your case it will be something like this: On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker wrote: > Hi List, > > for the first experiments with freeswitch I downloaded the Windows > installation. > Now Im trying to get my 2 Sipphones get connected to. Later I want > connect the freeswitch to my asterisk gateway. > > I find the examples pretty complex therfore Im trying to build up a > simple solution to understand the functions from the scratch .. > > my current problem is , that I cant route my local sips to each other ( > registration seems to work now). > the next is , that freeshwitch is not able to connect to asterisk. but I > will describe this later. > > I installed in the Directory a xml file ( called 22.xml) with the > following content : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > This seems to be ok now. Now I want to dial from 22 to 24 , wherefore I > configured this dialplan : > > > > > > > > > > > wich doesnt work , mybe b/c the user/${dialed_extension} I dont know... > Freeswitch says: > [INFO] switch_core_state_machine.c:136 No Route, Aborting > [NOTICE] switch_core_state_machine.c:137 Hangup > sofia/internal/24 at 192.168.1.34 [CS_ROUTING] [NO_ROUTE_DESTINATION] > [NOTICE] switch_core_session.c:1086 Session 17 > (sofia/internal/24 at 192.168.1.34) Ended > [NOTICE] switch_core_session.c:1088 Close Channel > sofia/internal/24 at 192.168.1.34 [CS_DESTROY] > > Im sure , for you guys this cant be a big deal;) > > > Next Point is my Asterisk registration , mybe you can help me out here > to .. : > > In the sip-profiles/external I installed the my_asterisk.xml with that > content : > > > > > > > > > > > > Freeswitch allways complains a timeout error : > [ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request > Timeout [408]. failure #17 > [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry > to 540 seconds. > > it seems that It cant connect ( I also tried out to explicit set the > port to 5060 b/c I read something about 5080 .. : value="5060"> but this didnt help) > In my Asterisk I set in the sip.conf the entry 28 with the pw test .... > > > If someone could help me with my first steps I would be verrry thankful ;)) > > cheers > > > Filip > > -- > _________________________________ > Filip Lyncker, Dipl.-Inform. (FH) > > > Lyncker & Theis GmbH > Wilhelmstr. 16 > 65185 Wiesbaden > Germany > > Fon +49 611/9006951 > Fax +49 611/9406125 > > > Handelsregister: HRB 23156 Amtsgericht Wiesbaden > Steuernummer: 4023897051 > USt-IdNr.: DE255806399 > > Gesch?ftsf?hrer: > Filip Lyncker, > Armin Theis > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/47dbaa8b/attachment.html From email.list.subscriber at gmail.com Fri Sep 18 11:26:33 2009 From: email.list.subscriber at gmail.com (email lists) Date: Fri, 18 Sep 2009 14:26:33 -0400 Subject: [Freeswitch-users] FreeSWITCH 64bit compilation error (Solaris 10) Message-ID: <4ab3d091.9453f10a.0891.2fa4@mx.google.com> Forwarding the issue below to see if anyone is familiar with this issue, and/or what our next steps should be. Thanks, Vladimir Looks like a problem with a Makefile not honoring CFLAGS,etc. Perhaps you can report this to the dev team. Other components built fine but this damn spidermonkey is buggering. # file /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch*.o | head /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-g711.o : ELF 64-bit LSB relocatable AMD64 Version 1 /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-getgat eway.o: ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-igd_de sc_parse.o: ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-libtel etone_detect.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE2 SSE AMD_3DNow CMOV FPU] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-libtel etone_generate.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE2 SSE AMD_3DNow CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniso ap.o: ELF 64-bit LSB relocatable AMD64 Version 1 /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniss dpc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniup npc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE AMD_3DNow] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniwg et.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-minixm l.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] # file /opt/freeradius-client/sbin/radacct /opt/freeradius-client/sbin/radacct: ELF 64-bit LSB executable AMD64 Version 1 [SSE2 SSE FXSR AMD_3DNow CMOV FPU], dynamically linked, not stripped Build output: Making all in nua LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la LINK libsofia-sip-ua.la libtool: link: warning: `-version-info/-version-number' is ignored for convenience libraries Making all in packages Creating mod_sofia_la-mod_sofia.lo mkdir .libs Compiling mod_sofia.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia.lo Compiling sofia.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) "sofia.c", line 3522: warning: enum type mismatch: arg #2 (E_ENUM_TYPE_MISMATCH_ARG) Creating mod_sofia_la-sofia_glue.lo Compiling sofia_glue.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_presence.lo Compiling sofia_presence.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_reg.lo Compiling sofia_reg.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_sla.lo Compiling sofia_sla.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofi a-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia.la making all mod_speex Compiling mod_speex.c... mkdir .libs Compiling mod_speex.c ... Creating mod_speex.so... making all mod_spidermonkey cd config; /usr/sfw/bin/gmake -j1 export ld: fatal: file now.o: wrong ELF class: ELFCLASS64 ld: fatal: File processing errors. No output written to now gmake[7]: *** [now] Error 1 gmake[6]: *** [export] Error 2 gmake[5]: *** [/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/js/libjs.la] Error 2 gmake[4]: *** [all] Error 1 gmake[3]: *** [mod_spidermonkey-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /usr/sfw/bin/gmake install + +----------------------------------------------+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/7c284fe2/attachment-0001.html From andrew at hijacked.us Fri Sep 18 11:55:59 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Fri, 18 Sep 2009 14:55:59 -0400 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> Message-ID: <20090918185559.GJ20978@hijacked.us> On Thu, Sep 17, 2009 at 11:20:22AM -0700, Michael Collins wrote: > I was curious about this myself. Even if someone has built a non-free > skills-based ACD using FS I'd like to know about it. > -MC I guess nobody paid any attention to my Cluecon presentation... :( http://wiki.opencsm.org/wiki/index.php/Spice_Telephony is a skill-based ACD that uses FS for its voice components. I havent pimped it here in quite a while but here's some of its major features * Skill based routing * Priority Queues (instead of just FIFO) * Multiple call types (voice, voicemail and email are currently supported, instant message support (via libpurple) is prototyped) * Outbound call support (no autodialer though) * Distributed system so you can aggregate multiple FS instances/locations into one big 'virtual' callcenter * Web-based agent and administrative interface There's quite a bit more, but that's the overview. The project is finally approaching a 1.0 after over a year of development - I hope to deploy it in production sometime around the end of this year or the beginning of 2010 (replacing my previous custom asterisk solution). You can grab the code at http://git.opencsm.org/index.cgi/spice-telephony/ (you can browse or git clone that URL). All you should need to run it is a modern erlang release (R12B5 or newer) and ruby/rake to run the build. Andrew From grevenx at me.com Fri Sep 18 12:06:46 2009 From: grevenx at me.com (=?iso-8859-1?Q?Even_Andr=E9_Fiskvik?=) Date: Fri, 18 Sep 2009 21:06:46 +0200 Subject: [Freeswitch-users] FreeSWITCH 64bit compilation error (Solaris 10) In-Reply-To: <4ab3d091.9453f10a.0891.2fa4@mx.google.com> References: <4ab3d091.9453f10a.0891.2fa4@mx.google.com> Message-ID: I've been having problems of somewhat the same, but for my case it has been forcing a 32-bit version in Mac OS X Snow Leopard rather than the default for GCC which is to build it in 64-bit. I have not been able to get this done yet, and I also end up with a mix of two different architectures across the modules. Did he pass --enable-64 to the ./configure command and make clean ? Best regards, Even Andr? Fiskvik / grEvenX On 18. sep. 2009, at 20.26, email lists wrote: > Forwarding the issue below to see if anyone is familiar with this > issue, and/or what our next steps should be. > > Thanks, > Vladimir > > > Looks like a problem with a Makefile not honoring CFLAGS,etc. > Perhaps you can report this to the dev team. Other components built > fine but this damn spidermonkey is buggering. > > # file /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/ > libfreeswitch*.o | head > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > g711.o: ELF 64-bit LSB relocatable AMD64 Version 1 > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > getgateway.o: ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > igd_desc_parse.o: ELF 64-bit LSB relocatable AMD64 Version 1 > [CMOV] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > libteletone_detect.o: ELF 64-bit LSB relocatable AMD64 Version 1 > [SSE2 SSE AMD_3DNow CMOV FPU] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > libteletone_generate.o: ELF 64-bit LSB relocatable AMD64 Version 1 > [SSE2 SSE AMD_3DNow CMOV] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > minisoap.o: ELF 64-bit LSB relocatable AMD64 Version 1 > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > minissdpc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > miniupnpc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE > AMD_3DNow] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > miniwget.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > minixml.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] > > # file /opt/freeradius-client/sbin/radacct > /opt/freeradius-client/sbin/radacct: ELF 64-bit LSB executable > AMD64 Version 1 [SSE2 SSE FXSR AMD_3DNow CMOV FPU], dynamically > linked, not stripped > > Build output: > Making all in nua > LTCOMPILE nua.lo > LTCOMPILE nua_common.lo > LTCOMPILE nua_stack.lo > LTCOMPILE nua_server.lo > LTCOMPILE nua_client.lo > LTCOMPILE nua_extension.lo > LTCOMPILE nua_dialog.lo > LTCOMPILE outbound.lo > LTCOMPILE nua_params.lo > LTCOMPILE nua_register.lo > LTCOMPILE nua_registrar.lo > LTCOMPILE nua_session.lo > LTCOMPILE nua_options.lo > LTCOMPILE nua_message.lo > LTCOMPILE nua_publish.lo > LTCOMPILE nua_subnotref.lo > LTCOMPILE nua_notifier.lo > LTCOMPILE nua_event_server.lo > LTCOMPILE nua_tag.lo > LTCOMPILE nua_tag_ref.lo > LINK libnua.la > LINK libsofia-sip-ua.la > libtool: link: warning: `-version-info/-version-number' is ignored > for convenience libraries > Making all in packages > Creating mod_sofia_la-mod_sofia.lo > mkdir .libs > Compiling mod_sofia.c ... > "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/ > libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable > bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) > Creating mod_sofia_la-sofia.lo > Compiling sofia.c ... > "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/ > libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable > bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) > "sofia.c", line 3522: warning: enum type mismatch: arg #2 > (E_ENUM_TYPE_MISMATCH_ARG) > Creating mod_sofia_la-sofia_glue.lo > Compiling sofia_glue.c ... > "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/ > libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable > bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) > Creating mod_sofia_la-sofia_presence.lo > Compiling sofia_presence.c ... > "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/ > libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable > bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) > Creating mod_sofia_la-sofia_reg.lo > Compiling sofia_reg.c ... > "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/ > libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable > bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) > Creating mod_sofia_la-sofia_sla.lo > Compiling sofia_sla.c ... > "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/ > libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable > bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) > Creating mod_sofia.la > > making all mod_speex > Compiling mod_speex.c... > mkdir .libs > Compiling mod_speex.c ... > Creating mod_speex.so... > > making all mod_spidermonkey > cd config; /usr/sfw/bin/gmake -j1 export > ld: fatal: file now.o: wrong ELF class: ELFCLASS64 > ld: fatal: File processing errors. No output written to now > gmake[7]: *** [now] Error 1 > gmake[6]: *** [export] Error 2 > gmake[5]: *** [/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/ > js/libjs.la] Error 2 > gmake[4]: *** [all] Error 1 > gmake[3]: *** [mod_spidermonkey-all] Error 1 > gmake[2]: *** [all-recursive] Error 1 > Making all in build > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + /usr/sfw/bin/gmake install + > +----------------------------------------------+ > gmake[1]: *** [all-recursive] Error 1 > gmake: *** [all] Error 2 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/fd91390e/attachment-0001.html From msc at freeswitch.org Fri Sep 18 12:35:16 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Sep 2009 12:35:16 -0700 Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! In-Reply-To: <191c3a030909180753h77337721r2066c49b7f403d0a@mail.gmail.com> References: <6E8D2069C08AA84A83D336E996AE4C6702DCBE4F41@mse17be1.mse17.exchange.ms> <85b1dd14700c0857d483d74807a5c6d2.squirrel@correo.nodo50.org> <191c3a030909180753h77337721r2066c49b7f403d0a@mail.gmail.com> Message-ID: <87f2f3b90909181235q482bbc39oc6b7d4a4a0ed9f80@mail.gmail.com> FYI, I did a POC on this: dump_arg.lua: -- dump_args.lua -- print out the args freeswitch.consoleLog("info", "Arg1: " .. argv[1] .. \n") freeswitch.consoleLog("info", "Arg2: " .. argv[2] .. "\n") >From there you can do whatever you want in the target script. I'm sure perlrun, pyrun, and jsrun are all the same in terms of accepting args and running whatever you want, like generating an originate API, etc. Just remember that the caller needs to hangup before you can call him back. :) -MC On Fri, Sep 18, 2009 at 7:53 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > You could put an api_hangup_hook on the channel to jsrun your script. > > What you want with javascript is not going to happen as long as you execute > the script *WITH* the channel. > it's not a problem it's just misuse/misunderstanding on your part. > > > > > On Fri, Sep 18, 2009 at 5:03 AM, Alberto Escudero wrote: > >> Thanks for all the tips. I tried to run apiExecute(bgapi, jsrun originate) >> and still Javascript?? does not let the thread go. >> >> No matter the combination of session.hangup(), exit, apiExecute with or >> without bgapi, the state remains in CS_EXECUTE. >> >> So at the end i am triggering an event that i can later use to execute a >> originate callback. It is nicer with ESL but i still think that will be >> nice to have a real way to expunge a second Javascript and let the first >> one die. >> >> The GSM channel/modem needs to be free-free (as I am a serial port-free) >> to handle the outgoing call. The callback script worked perfect with SIP >> because it does not care how many sessions are running in parallel. It can >> always place a call back event the channel is not properly close. >> >> /aep >> >> >> -- >> Stopping junk mailers is good for the environment >> >> > So, what happens is that when you are executing an app, the state is >> > CS_EXECUTE. Even if the session is hungup, the state machine doesn't go >> > through all the hangup code until your app executes. >> > >> > The easiest workaround is probably to start a background api (bgapi?) >> call >> > to a script. This will happen on another thread, then allow your current >> > thread to execute and the hangup code will execute. This should work >> just >> > fine, I think. (You can stop reading here.) >> > >> > But wait, there's even more fun! anthm recently checked in a change a >> > couple days that lets you work around this. Don't call destroy, call >> > hangup on the session, on that session's thread. This will perform a >> > hangup, then progress the state machine. Then the session will truly be >> > hungup. Maybe you need update your freeswitch code, if this is not >> > happening for you. >> > >> > If you updated and hangup still isn't hanging up, you might want to ask >> > specifically about that. Or, you may need to call >> > switch_core_session_hangup_state directly -- just hangup alone might not >> > do the trick. This is a C function, and not exposed to languages by >> > default - you can either patch javascript plugin to expose this safely >> > (and I have no idea what this means for the javascript runtime), or use >> a >> > more capable plugin like mod_managed which _does_ expose all the C >> > functions, and lets you call in and out of them as you please. >> > >> > And now, someone who knows what they're talking about will chime in and >> > point out what I got wrong. >> > >> > Thanks, >> > -Michael >> > >> > -----Original Message----- >> > From: freeswitch-users-bounces at lists.freeswitch.org >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> > Alberto Escudero >> > Sent: Thursday, September 17, 2009 3:20 PM >> > To: freeswitch-users at lists.freeswitch.org >> > Subject: [Freeswitch-users] Callback in Javascript, session.destroy() >> does >> > not free the channel! >> > >> > We are trying to create a callback application in Javascript. We get the >> > callerid from the unanswered call and after destroying the session, we >> > initiate a callback to the user to conenct it to a local extension in >> the >> > dialplan. >> > >> > Although we have tried to destroy the first session, or even invoke a >> > second script using apiExecute("jsrun",dialer.js"), tried >> session.hangup() >> > or exit()... the first session does not seem to close properly until the >> > whole chain of scripts are completed. >> > >> > Here is a piece of code that shows the concept (yes!, the sleep function >> > is far from ideal. CPU loves it! ) >> > >> > function sleep(milliseconds) { >> > var start = new Date().getTime(); >> > for (var i = 0; i < 1e7; i++) { >> > if ((new Date().getTime() - start) > milliseconds){ >> > break; >> > } >> > } >> > } >> > >> > if (session.ready()) { >> > //We catch the caller_id >> > caller_id_num = session.caller_id_num; >> > >> > console_log("Now we got your Caller ID\n"); >> > >> > //How long we want to wait to trigger a call back >> > session.execute("sleep",5000); >> > >> > console_log("We have waited a while... time to create the >> > callback\n"); >> > >> > //apiExecute("jsrun", "callback.js"); >> > } >> > >> > //Destroy the session... >> > session.destroy(); >> > session=undefined; >> > >> > sleep(10000); >> > >> > //Preparing callback >> > session2 = new >> > Session('{ignore_early_media=true}celliax/interface1/600464646'); >> > session2.setAutoHangup(false); >> > session2.answer(); >> > exit(); >> > >> > ++ >> > Wisdom thoughts? >> > >> > -- >> > Stopping junk mailers is good for the environment >> > >> > >> > >> > >> > _______________________________________________ >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 at 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 at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/be78a1d3/attachment.html From mgende at gendesign.com Fri Sep 18 12:49:48 2009 From: mgende at gendesign.com (Michael Gende) Date: Fri, 18 Sep 2009 14:49:48 -0500 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call, Sept 18, Starting Shortly In-Reply-To: <87f2f3b90909180855l6de71215vce8a712e33542246@mail.gmail.com> References: <87f2f3b90909180855l6de71215vce8a712e33542246@mail.gmail.com> Message-ID: Sorry I didn't make the meeting! Business. But, I'm still producing the "home grown" doc I mentioned (and was referenced in the Agenda for this meeting). Available as soon as I get a complete v1.0. Have a FS and * Biz Ed box working in tandem in our office at the moment. As soon as I get a few more details covered, the * is getting turned down. Really like the quality of sound on our FS. Mike G. On Fri, Sep 18, 2009 at 10:55 AM, Michael Collins wrote: > Get ready to call in! Agenda is here: > http://wiki.freeswitch.org/wiki/FS_weekly_2009_09_18 > > Talk to you all shortly! > -MC > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/c943abc2/attachment.html From msc at freeswitch.org Fri Sep 18 13:04:12 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Sep 2009 13:04:12 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly Conference Call, Sept 18, Starting Shortly In-Reply-To: References: <87f2f3b90909180855l6de71215vce8a712e33542246@mail.gmail.com> Message-ID: <87f2f3b90909181304g438aafc0ya61e96256e598d8d@mail.gmail.com> On Fri, Sep 18, 2009 at 12:49 PM, Michael Gende wrote: > Sorry I didn't make the meeting! Business. > > But, I'm still producing the "home grown" doc I mentioned (and was > referenced in the Agenda for this meeting). Available as soon as I get a > complete v1.0. > > Have a FS and * Biz Ed box working in tandem in our office at the moment. > As soon as I get a few more details covered, the * is getting turned down. > Really like the quality of sound on our FS. > > Glad to hear it! Looking forward to your information. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/0d753591/attachment-0001.html From msc at freeswitch.org Fri Sep 18 13:05:29 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Sep 2009 13:05:29 -0700 Subject: [Freeswitch-users] Checking Dial Status in FS In-Reply-To: References: Message-ID: <87f2f3b90909181305k16b5b0d7u115eeeffc7b27e20@mail.gmail.com> On Mon, Sep 7, 2009 at 9:14 PM, Ahmed Munir wrote: > Hi, > > In FS, which function can be used for checking the dial status (in channel > variables as well as mod_perl function/class)? In asterisk $DIALSTATUS is > used to check the status i.e. busy, answer, etc. > > Kindly do let me know. > Are you building an application with the event socket? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/78b1cf4c/attachment.html From msc at freeswitch.org Fri Sep 18 13:24:35 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Sep 2009 13:24:35 -0700 Subject: [Freeswitch-users] Zaptel Analog FXO Outbound Dialing In-Reply-To: <20090914003529.784c4fc4@neutrino.joshua-colp.com> References: <20090914003529.784c4fc4@neutrino.joshua-colp.com> Message-ID: <87f2f3b90909181324k104e5bf1t5b456c19cf83525@mail.gmail.com> FYI, I forgot to mention that Tony fixed this on Monday. http://fisheye.freeswitch.org/changelog/OpenZAP/?cs=834 -MC On Sun, Sep 13, 2009 at 5:35 PM, wrote: > > ------------------------------ > My suspicion is that this is only for zaptel type cards. Our tests > with Sangoma analog cards have all been pretty successful. But thanks > for info! Anyone else using Rhino, Digium, or compatible analog cards? > > I am not experiencing an audio delay. My configuration is exactly as > documented on the Zaptel Tutorial wiki page ( > http://wiki.freeswitch.org/wiki/Zaptel_Tutorial). I'm using a Digium > TDM400P, Zaptel 1.4 revision 4630, and FreeSWITCH trunk revision 14842. If > you want me to try anything for you, I'm 'Deeewayne' on IRC. > > -Dwayne. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/060ef681/attachment.html From msc at freeswitch.org Fri Sep 18 13:27:08 2009 From: msc at freeswitch.org (Michael Collins) Date: Fri, 18 Sep 2009 13:27:08 -0700 Subject: [Freeswitch-users] Unknown call drops.. INFO DTMF(3) In-Reply-To: <33c87fa30909130234u21541ae7o61c522d014e3dacd@mail.gmail.com> References: <33c87fa30909130234u21541ae7o61c522d014e3dacd@mail.gmail.com> Message-ID: <87f2f3b90909181327p78361ffflfabe3c57edab01f2@mail.gmail.com> Is this still happening? If so please make sure that you are on latest trunk and re-test. Get a pcap of the traffic (SIP and RTP) for review and then report back. Thanks, MC On Sun, Sep 13, 2009 at 2:34 AM, Mark Campbell-Smith < mcampbellsmith at gmail.com> wrote: > Hi! > > I have just experienced some call drops and each time the sequence is > the same in the freeswitch.log file. Both parties are sure that they > did not accidentally hit the 3 button to send the DTMF tone (and the > same thing has happened four times already after ~5 minutes). > > 2009-09-13 19:28:23.835216 [DEBUG] sofia.c:4329 INFO DTMF(3) > 2009-09-13 19:28:23.835216 [DEBUG] sofia.c:4450 dispatched freeswitch > event for INFO > 2009-09-13 19:28:23.859408 [DEBUG] switch_rtp.c:1624 Send start packet > for [3] ts=2591120 dur=160/160/13120 seq=64923 > 2009-09-13 19:28:23.879439 [DEBUG] switch_rtp.c:1560 Send middle > packet for [3] ts=2591120 dur=320/320/13120 seq=64924 > 2009-09-13 19:28:23.899455 [DEBUG] switch_rtp.c:1560 Send middle > packet for [3] ts=2591120 dur=480/480/13120 seq=64925 > : > : > 2009-09-13 19:28:25.439404 [DEBUG] switch_rtp.c:1560 Send middle > packet for [3] ts=2591120 dur=12800/12800/13120 seq=65002 > 2009-09-13 19:28:25.459312 [DEBUG] switch_rtp.c:1560 Send middle > packet for [3] ts=2591120 dur=12960/12960/13120 seq=65003 > 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet > for [3] ts=2591120 dur=13120/13120/13120 seq=65004 > 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet > for [3] ts=2591120 dur=13120/13120/13120 seq=65005 > 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet > for [3] ts=2591120 dur=13120/13120/13120 seq=65006 > 2009-09-13 19:28:33.879341 [NOTICE] sofia.c:322 Hangup > sofia/external/ [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] > 2009-09-13 19:28:33.879341 [DEBUG] switch_channel.c:1683 Send signal > sofia/external/ [KILL] > 2009-09-13 19:28:33.879341 [DEBUG] switch_core_session.c:932 Send > signal sofia/external/ [BREAK] > 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:371 > sofia/external/ ending bridge by request from write function > 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:426 > sofia/internal_nat/1000 at 192.168.1.120 receive message [UNBRIDGE] > 2009-09-13 19:28:33.900940 [DEBUG] switch_core_session.c:630 Send > signal sofia/internal_nat/1000 at 192.168.1.120 [BREAK] > 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:452 BRIDGE > THREAD DONE [sofia/internal_nat/1000 at 192.168.1.120] > 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:454 Send signal > sofia/external/ [BREAK] > 2009-09-13 19:28:33.912049 [NOTICE] switch_core_state_machine.c:179 > Hangup sofia/internal_nat/1000 at 192.168.1.120 [CS_EXECUTE] > [NORMAL_CLEARING] > > Anyone have any idea what this sequence means and why I am getting > this? Is it my sip provider or something in FreeSwitch? What does > the 'Send end packet for [3] ts=2591120 dur=13120/13120/13120 > seq=65006' mean? Notice that dur (duration?) is increasing a lot > until the call drops. > > Thanks! > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090918/8887f67c/attachment.html From amran23 at hotmail.com Fri Sep 18 16:27:22 2009 From: amran23 at hotmail.com (Amran Sarkar) Date: Sat, 19 Sep 2009 10:27:22 +1100 Subject: [Freeswitch-users] I have good quality Bangladesh Mobile White route (8801) In-Reply-To: References: <4ab3d091.9453f10a.0891.2fa4@mx.google.com> Message-ID: Dear All, I have good quality Bangldesh white route with good rates. if any one interested please contact with me. Thanks Md Amran Ali Sarkar OMEGA Technology. From: grevenx at me.com Date: Fri, 18 Sep 2009 21:06:46 +0200 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSWITCH 64bit compilation error (Solaris 10) I've been having problems of somewhat the same, but for my case it has been forcing a 32-bit version in Mac OS X Snow Leopard rather than the default for GCC which is to build it in 64-bit. I have not been able to get this done yet, and I also end up with a mix of two different architectures across the modules. Did he pass --enable-64 to the ./configure command and make clean ? Best regards, Even Andr? Fiskvik / grEvenX On 18. sep. 2009, at 20.26, email lists wrote: Forwarding the issue below to see if anyone is familiar with this issue, and/or what our next steps should be. Thanks, Vladimir Looks like a problem with a Makefile not honoring CFLAGS,etc. Perhaps you can report this to the dev team. Other components built fine but this damn spidermonkey is buggering. # file /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch*.o | head /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-g711.o: ELF 64-bit LSB relocatable AMD64 Version 1 /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-getgateway.o: ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-igd_desc_parse.o: ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-libteletone_detect.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE2 SSE AMD_3DNow CMOV FPU] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-libteletone_generate.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE2 SSE AMD_3DNow CMOV] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-minisoap.o: ELF 64-bit LSB relocatable AMD64 Version 1 /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-minissdpc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniupnpc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE AMD_3DNow] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-miniwget.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la-minixml.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] # file /opt/freeradius-client/sbin/radacct /opt/freeradius-client/sbin/radacct: ELF 64-bit LSB executable AMD64 Version 1 [SSE2 SSE FXSR AMD_3DNow CMOV FPU], dynamically linked, not stripped Build output: Making all in nua LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la LINK libsofia-sip-ua.la libtool: link: warning: `-version-info/-version-number' is ignored for convenience libraries Making all in packages Creating mod_sofia_la-mod_sofia.lo mkdir .libs Compiling mod_sofia.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia.lo Compiling sofia.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) "sofia.c", line 3522: warning: enum type mismatch: arg #2 (E_ENUM_TYPE_MISMATCH_ARG) Creating mod_sofia_la-sofia_glue.lo Compiling sofia_glue.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_presence.lo Compiling sofia_presence.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_reg.lo Compiling sofia_reg.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia_la-sofia_sla.lo Compiling sofia_sla.c ... "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) Creating mod_sofia.la making all mod_speex Compiling mod_speex.c... mkdir .libs Compiling mod_speex.c ... Creating mod_speex.so... making all mod_spidermonkey cd config; /usr/sfw/bin/gmake -j1 export ld: fatal: file now.o: wrong ELF class: ELFCLASS64 ld: fatal: File processing errors. No output written to now gmake[7]: *** [now] Error 1 gmake[6]: *** [export] Error 2 gmake[5]: *** [/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/js/libjs.la] Error 2 gmake[4]: *** [all] Error 1 gmake[3]: *** [mod_spidermonkey-all] Error 1 gmake[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + /usr/sfw/bin/gmake install + +----------------------------------------------+ gmake[1]: *** [all-recursive] Error 1 gmake: *** [all] Error 2_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________ Show them the way! Add maps and directions to your party invites. http://www.microsoft.com/windows/windowslive/products/events.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090919/97573dc1/attachment-0001.html From william.suffill at gmail.com Fri Sep 18 17:08:22 2009 From: william.suffill at gmail.com (William Suffill) Date: Fri, 18 Sep 2009 20:08:22 -0400 Subject: [Freeswitch-users] I have good quality Bangladesh Mobile White route (8801) In-Reply-To: References: <4ab3d091.9453f10a.0891.2fa4@mx.google.com> Message-ID: <6b65470d0909181708t1e3cad2cm4e1b872d9b98f9c5@mail.gmail.com> Probably best to post these type of offers on the freeswitch-biz list http://lists.freeswitch.org/mailman/listinfo/freeswitch-biz (Hopefully that list will eventually grow with business posts. Still pretty slow at present) Also best to create a new e-mail versus reply to another thread and changing the subject. -- W From kjv at ken-ton.com Fri Sep 18 21:21:40 2009 From: kjv at ken-ton.com (Karl Vesterling) Date: Sat, 19 Sep 2009 00:21:40 -0400 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: <9AC967E2-2807-4FB7-BA8E-F5C128BCDE76@freeswitch.org> References: <35265AD6-3DC1-4BB7-971F-F43D3934C430@ken-ton.com> <00FEDAA6-5061-4AFD-A8E2-039A351AA10C@freeswitch.org> <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> <8A58CD3E-F0C3-4BAD-8DB9-3C9D3A22A70F@ken-ton.com> <191c3a030909140617v396063e7n86cd8fdb39d75fa9@mail.gmail.com> <0E10F348-B983-48F9-87DB-19BE52CEFD50@ken-ton.com> <20090917040742.GA24786@jdc.jasonjgw.net> <9AC967E2-2807-4FB7-BA8E-F5C128BCDE76@freeswitch.org> Message-ID: <9C617DD2-DDFD-4160-81EE-B0FD5E2892EA@ken-ton.com> No penguin is perfect... There's issues w/ 2.6.X - 2.6.27.X with respect to timing for things like packet shaping, which is a requirement for me. 2.6.29.X onward, well, I might be inclined to try the latest revision, but last I tested was 2.6.30, and it was truly all round badness, with everything. They seem to have the IRQ handling problems re-worked, but even if you compile WITHOUT dynamic ticks, it's still horrible on the CPU. I'll wait until 2.6.32... What I got now is working w/ the workaround. Thanks Brian. (thinking to self, you know... Maybe there's a reason for the bias of these BSD zealots???? It ran fine on my old 11/780 which doubled as a forced air furnas...) Best Regards, Karl J. Vesterling kjv at ken-ton.com 202-461-3231 x0 On Sep 17, 2009, at 9:11 AM, Brian West wrote: > Its a bug in 2.6.26 thru 2.6.28 kernels that impact the performance of > SQLite. He was specifically running SUSE. > > /b > > On Sep 16, 2009, at 11:07 PM, Jason White wrote: > >> Please take this up with your Linux distribution as a bug report >> related to >> the kernel, and persist with it until it's sorted out. >> >> The more that users do this, the more kernel bugs will get fixed. >> >> We're all responsible to some extent for the quality of our free/ >> open-source >> operating systems. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > From pankajanand18 at gmail.com Fri Sep 18 22:50:56 2009 From: pankajanand18 at gmail.com (pankaj anand) Date: Sat, 19 Sep 2009 11:20:56 +0530 Subject: [Freeswitch-users] Not able to make call using external profile In-Reply-To: <809ad7ab0909180711v73f15899v8e9dc64cb191e82c@mail.gmail.com> References: <809ad7ab0909180313n5538c1c0ve1b8070c0d1da2cc@mail.gmail.com> <809ad7ab0909180711v73f15899v8e9dc64cb191e82c@mail.gmail.com> Message-ID: <809ad7ab0909182250u6fbfa25pe871bcdb7dcc91c@mail.gmail.com> @Tihomir Culjaga HI folks, thanx for such a quick reply. Q. what I want to achieve with FreeSwitch ? A: I want to enable the outside users ( from internet) to have video chat on peer2peer using freeSwitch for signaling. External Profile is being used to for this. External profile is using 5080 port. That port is forwarded on the NAT server. Users are able to connect using 5080 port. They get registered with no issues. Q. where do you want to send calls ? A. I want to send call from one extension to another extension ( both extension exist on the are on public internet). Right now i m trying with 1000 and 1001 user available in the default directory. 1. What is 192.168.1.50 ? Ans: well , this is my domain name which is by default the local-ip address of the machine. My current setup is like this: FreeSwitch ( 192.168.1.50) ---->NAT(122.162.153.224)-->Internet<----(122.80.0.180)NAT<--(192.168.1.15)1001(user) 2. Where/how are you originating calls from ? 1. I am using X-lite, Phoner , LinPhone to make calls. All these phones have stun server enabled . For the public dial plan I have added these lines in the file public.xml which is used by the external profile Now the echo calls works through the external profile. But when a call is being made to some other user, for example if user 1000 makes a call to the 1001 it reaches to the "public_extensions " but it generates the error which I have already mentioned. For the gateway thing , not gateway is being used. On Fri, Sep 18, 2009 at 7:41 PM, pankaj anand wrote: > I m using default configuration of freeswitch.. I m not using any gateway > for authentication. > in the $INSTALLDIR/conf/sip_profiles/external/ directory, there exist only > one file which example.xml , this files contains > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > as you can see, all the lines are commented. So i m not using any gateways. > > > > On Fri, Sep 18, 2009 at 3:43 PM, pankaj anand wrote: > >> hi folks, I m not able to make SIP calls using external profile. >> >> i have added the following lines to the >> $installdir/conf/dialplan/public.xml >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> I m able to connect using 1000 and 1001 from public Internet. I am able >> to make an echo call. >> >> *when i type :* >> >> $: sofia status profile external reg >> >> It shows the list of the connected clients and their information. >> >> but when I m trying to make a call from one user to other user, it >> generates the following error >> >> >> 2009-09-18 15:41:45.675054 [NOTICE] switch_channel.c:602 New Channel >> sofia/external/1001 at 192.168.1.50 [fcb6c23e-bdcd-41dd-b73e-df07b71252be] >> 2009-09-18 15:41:45.677063 [INFO] mod_dialplan_xml.c:315 Processing >> 1000->1000 in context public >> 2009-09-18 15:41:45.679071 [NOTICE] switch_channel.c:602 New Channel >> sofia/external/1000 at 192.168.1.50 [1a537865-be53-42ce-b8f5-cc183f4f1306] >> 2009-09-18 15:41:45.688161 [ERR] sofia_reg.c:1568 No Matching gateway >> found >> 2009-09-18 15:41:45.688161 [NOTICE] sofia_reg.c:1588 Hangup >> sofia/external/1000 at 192.168.1.50 [CS_CONSUME_MEDIA] >> [MANDATORY_IE_MISSING] >> 2009-09-18 15:41:45.688161 [INFO] mod_dptools.c:2093 Originate Failed. >> Cause: MANDATORY_IE_MISSING >> 2009-09-18 15:41:45.689090 [NOTICE] mod_dptools.c:2125 Hangup >> sofia/external/1001 at 192.168.1.50 [CS_EXECUTE] [MANDATORY_IE_MISSING] >> 2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1086 Session 1 >> (sofia/external/1001 at 192.168.1.50) Ended >> 2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1088 Close >> Channel sofia/external/1001 at 192.168.1.50 [CS_DESTROY] >> 2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1086 Session 2 >> (sofia/external/1000 at 192.168.1.50) Ended >> 2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1088 Close >> Channel sofia/external/1000 at 192.168.1.50 [CS_DESTROY] >> >> >> with regards >> Pankaj anand >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090919/b2810f01/attachment-0001.html From jason at jasonjgw.net Fri Sep 18 23:49:01 2009 From: jason at jasonjgw.net (Jason White) Date: Sat, 19 Sep 2009 16:49:01 +1000 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: <9C617DD2-DDFD-4160-81EE-B0FD5E2892EA@ken-ton.com> References: <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> <8A58CD3E-F0C3-4BAD-8DB9-3C9D3A22A70F@ken-ton.com> <191c3a030909140617v396063e7n86cd8fdb39d75fa9@mail.gmail.com> <0E10F348-B983-48F9-87DB-19BE52CEFD50@ken-ton.com> <20090917040742.GA24786@jdc.jasonjgw.net> <9AC967E2-2807-4FB7-BA8E-F5C128BCDE76@freeswitch.org> <9C617DD2-DDFD-4160-81EE-B0FD5E2892EA@ken-ton.com> Message-ID: <20090919064901.GA2129@jdc.jasonjgw.net> Karl Vesterling wrote: > No penguin is perfect... > There's issues w/ 2.6.X - 2.6.27.X with respect to timing for things > like packet shaping, which is a requirement for me. Two suggestions: 1. Your distribution's bug tracker. 2. http://ltp.sourceforge.net/ (If they get test coverage of the relevant interfaces there will be quicker detection of problems and, we hope, prevention of regressions.) From tculjaga at gmail.com Sat Sep 19 00:04:33 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sat, 19 Sep 2009 09:04:33 +0200 Subject: [Freeswitch-users] Not able to make call using external profile In-Reply-To: <809ad7ab0909182250u6fbfa25pe871bcdb7dcc91c@mail.gmail.com> References: <809ad7ab0909180313n5538c1c0ve1b8070c0d1da2cc@mail.gmail.com> <809ad7ab0909180711v73f15899v8e9dc64cb191e82c@mail.gmail.com> <809ad7ab0909182250u6fbfa25pe871bcdb7dcc91c@mail.gmail.com> Message-ID: <65d96fc80909190004v6a0ec669h3c63db00c2762360@mail.gmail.com> check this: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User dial registered user: dial external endpoint: another issue you might have with RTP so check the wiki for NAT config as well. T. On Sat, Sep 19, 2009 at 7:50 AM, pankaj anand wrote: > @Tihomir Culjaga > > HI folks, > > thanx for such a quick reply. > > > > Q. what I want to achieve with FreeSwitch ? > > A: I want to enable the outside users ( from internet) to have video > chat on peer2peer using freeSwitch for signaling. External Profile is being > used to for this. External profile is using 5080 port. That port is > forwarded on the NAT server. Users are able to connect using 5080 port. They > get registered with no issues. > > > > Q. where do you want to send calls ? > > A. I want to send call from one extension to another extension ( both > extension exist on the are on public internet). Right now i m trying with > 1000 and 1001 user available in the default directory. > > > 1. What is 192.168.1.50 ? > > Ans: well , this is my domain name which is by default the local-ip > address of the machine. My current setup is like this: > > FreeSwitch ( 192.168.1.50) > ---->NAT(122.162.153.224)-->Internet<----(122.80.0.180)NAT<--(192.168.1.15)1001(user) > > > 2. > > Where/how are you originating calls from ? > > > 1. I am using X-lite, Phoner , LinPhone to make calls. All these phones > have stun server enabled . > > > > For the public dial plan I have added these lines in the file > public.xml which is used by the external profile > > > > > > expression="^(10[01][0-9])$"> > > > > > > > > > > > > > > > > > > > > > > > > > > Now the echo calls works through the external profile. But when a call > is being made to some other user, for example if user 1000 makes a call to > the 1001 it reaches to the "public_extensions " but it generates the > error which I have already mentioned. For the gateway thing , not gateway is > being used. > > > > > > > On Fri, Sep 18, 2009 at 7:41 PM, pankaj anand wrote: > >> I m using default configuration of freeswitch.. I m not using any gateway >> for authentication. >> in the $INSTALLDIR/conf/sip_profiles/external/ directory, there exist >> only one file which example.xml , this files contains >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> as you can see, all the lines are commented. So i m not using any >> gateways. >> >> >> >> On Fri, Sep 18, 2009 at 3:43 PM, pankaj anand wrote: >> >>> hi folks, I m not able to make SIP calls using external profile. >>> >>> i have added the following lines to the >>> $installdir/conf/dialplan/public.xml >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> I m able to connect using 1000 and 1001 from public Internet. I am able >>> to make an echo call. >>> >>> *when i type :* >>> >>> $: sofia status profile external reg >>> >>> It shows the list of the connected clients and their information. >>> >>> but when I m trying to make a call from one user to other user, it >>> generates the following error >>> >>> >>> 2009-09-18 15:41:45.675054 [NOTICE] switch_channel.c:602 New Channel >>> sofia/external/1001 at 192.168.1.50 [fcb6c23e-bdcd-41dd-b73e-df07b71252be] >>> 2009-09-18 15:41:45.677063 [INFO] mod_dialplan_xml.c:315 Processing >>> 1000->1000 in context public >>> 2009-09-18 15:41:45.679071 [NOTICE] switch_channel.c:602 New Channel >>> sofia/external/1000 at 192.168.1.50 [1a537865-be53-42ce-b8f5-cc183f4f1306] >>> 2009-09-18 15:41:45.688161 [ERR] sofia_reg.c:1568 No Matching gateway >>> found >>> 2009-09-18 15:41:45.688161 [NOTICE] sofia_reg.c:1588 Hangup >>> sofia/external/1000 at 192.168.1.50 [CS_CONSUME_MEDIA] >>> [MANDATORY_IE_MISSING] >>> 2009-09-18 15:41:45.688161 [INFO] mod_dptools.c:2093 Originate Failed. >>> Cause: MANDATORY_IE_MISSING >>> 2009-09-18 15:41:45.689090 [NOTICE] mod_dptools.c:2125 Hangup >>> sofia/external/1001 at 192.168.1.50 [CS_EXECUTE] [MANDATORY_IE_MISSING] >>> 2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1086 Session 1 >>> (sofia/external/1001 at 192.168.1.50) Ended >>> 2009-09-18 15:41:45.690064 [NOTICE] switch_core_session.c:1088 Close >>> Channel sofia/external/1001 at 192.168.1.50 [CS_DESTROY] >>> 2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1086 Session 2 >>> (sofia/external/1000 at 192.168.1.50) Ended >>> 2009-09-18 15:41:45.692078 [NOTICE] switch_core_session.c:1088 Close >>> Channel sofia/external/1000 at 192.168.1.50 [CS_DESTROY] >>> >>> >>> with regards >>> Pankaj anand >>> >>> >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090919/02de55ec/attachment.html From aep.lists at it46.se Sat Sep 19 02:08:26 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Sat, 19 Sep 2009 11:08:26 +0200 Subject: [Freeswitch-users] Callback in Javascript, session.destroy() does not free the channel! In-Reply-To: <87f2f3b90909181235q482bbc39oc6b7d4a4a0ed9f80@mail.gmail.com> References: <6E8D2069C08AA84A83D336E996AE4C6702DCBE4F41@mse17be1.mse17.exchange.ms> <85b1dd14700c0857d483d74807a5c6d2.squirrel@correo.nodo50.org> <191c3a030909180753h77337721r2066c49b7f403d0a@mail.gmail.com> <87f2f3b90909181235q482bbc39oc6b7d4a4a0ed9f80@mail.gmail.com> Message-ID: Hi Michael, I will like to get a few RINGS back to the user and sleep a bit before the call back. The second i can do using the app sleep. What about the first thing? http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready Will test i let you know... Crazy Callbacker aka aep -- Stopping junk mailers is good for the environment > FYI, > > I did a POC on this: > > > > > > > > dump_arg.lua: > > -- > dump_args.lua > > -- print out the > args > > > > freeswitch.consoleLog("info", "Arg1: " .. argv[1] .. > \n") > > freeswitch.consoleLog("info", "Arg2: " .. argv[2] .. > "\n") > > > >>From there you can do whatever you want in the target script. I'm sure > perlrun, pyrun, and jsrun are all the same in terms of accepting args and > running whatever you want, like generating an originate API, etc. Just > remember that the caller needs to hangup before you can call him back. :) > > -MC > > On Fri, Sep 18, 2009 at 7:53 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> You could put an api_hangup_hook on the channel to jsrun your script. >> >> What you want with javascript is not going to happen as long as you >> execute >> the script *WITH* the channel. >> it's not a problem it's just misuse/misunderstanding on your part. >> >> >> >> >> On Fri, Sep 18, 2009 at 5:03 AM, Alberto Escudero >> wrote: >> >>> Thanks for all the tips. I tried to run apiExecute(bgapi, jsrun >>> originate) >>> and still Javascript?? does not let the thread go. >>> >>> No matter the combination of session.hangup(), exit, apiExecute with or >>> without bgapi, the state remains in CS_EXECUTE. >>> >>> So at the end i am triggering an event that i can later use to execute >>> a >>> originate callback. It is nicer with ESL but i still think that will be >>> nice to have a real way to expunge a second Javascript and let the >>> first >>> one die. >>> >>> The GSM channel/modem needs to be free-free (as I am a serial >>> port-free) >>> to handle the outgoing call. The callback script worked perfect with >>> SIP >>> because it does not care how many sessions are running in parallel. It >>> can >>> always place a call back event the channel is not properly close. >>> >>> /aep >>> >>> >>> -- >>> Stopping junk mailers is good for the environment >>> >>> > So, what happens is that when you are executing an app, the state is >>> > CS_EXECUTE. Even if the session is hungup, the state machine doesn't >>> go >>> > through all the hangup code until your app executes. >>> > >>> > The easiest workaround is probably to start a background api (bgapi?) >>> call >>> > to a script. This will happen on another thread, then allow your >>> current >>> > thread to execute and the hangup code will execute. This should work >>> just >>> > fine, I think. (You can stop reading here.) >>> > >>> > But wait, there's even more fun! anthm recently checked in a change a >>> > couple days that lets you work around this. Don't call destroy, call >>> > hangup on the session, on that session's thread. This will perform a >>> > hangup, then progress the state machine. Then the session will truly >>> be >>> > hungup. Maybe you need update your freeswitch code, if this is not >>> > happening for you. >>> > >>> > If you updated and hangup still isn't hanging up, you might want to >>> ask >>> > specifically about that. Or, you may need to call >>> > switch_core_session_hangup_state directly -- just hangup alone might >>> not >>> > do the trick. This is a C function, and not exposed to languages by >>> > default - you can either patch javascript plugin to expose this >>> safely >>> > (and I have no idea what this means for the javascript runtime), or >>> use >>> a >>> > more capable plugin like mod_managed which _does_ expose all the C >>> > functions, and lets you call in and out of them as you please. >>> > >>> > And now, someone who knows what they're talking about will chime in >>> and >>> > point out what I got wrong. >>> > >>> > Thanks, >>> > -Michael >>> > >>> > -----Original Message----- >>> > From: freeswitch-users-bounces at lists.freeswitch.org >>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >>> > Alberto Escudero >>> > Sent: Thursday, September 17, 2009 3:20 PM >>> > To: freeswitch-users at lists.freeswitch.org >>> > Subject: [Freeswitch-users] Callback in Javascript, session.destroy() >>> does >>> > not free the channel! >>> > >>> > We are trying to create a callback application in Javascript. We get >>> the >>> > callerid from the unanswered call and after destroying the session, >>> we >>> > initiate a callback to the user to conenct it to a local extension in >>> the >>> > dialplan. >>> > >>> > Although we have tried to destroy the first session, or even invoke a >>> > second script using apiExecute("jsrun",dialer.js"), tried >>> session.hangup() >>> > or exit()... the first session does not seem to close properly until >>> the >>> > whole chain of scripts are completed. >>> > >>> > Here is a piece of code that shows the concept (yes!, the sleep >>> function >>> > is far from ideal. CPU loves it! ) >>> > >>> > function sleep(milliseconds) { >>> > var start = new Date().getTime(); >>> > for (var i = 0; i < 1e7; i++) { >>> > if ((new Date().getTime() - start) > milliseconds){ >>> > break; >>> > } >>> > } >>> > } >>> > >>> > if (session.ready()) { >>> > //We catch the caller_id >>> > caller_id_num = session.caller_id_num; >>> > >>> > console_log("Now we got your Caller ID\n"); >>> > >>> > //How long we want to wait to trigger a call back >>> > session.execute("sleep",5000); >>> > >>> > console_log("We have waited a while... time to create the >>> > callback\n"); >>> > >>> > //apiExecute("jsrun", "callback.js"); >>> > } >>> > >>> > //Destroy the session... >>> > session.destroy(); >>> > session=undefined; >>> > >>> > sleep(10000); >>> > >>> > //Preparing callback >>> > session2 = new >>> > Session('{ignore_early_media=true}celliax/interface1/600464646'); >>> > session2.setAutoHangup(false); >>> > session2.answer(); >>> > exit(); >>> > >>> > ++ >>> > Wisdom thoughts? >>> > >>> > -- >>> > Stopping junk mailers is good for the environment >>> > >>> > >>> > >>> > >>> > _______________________________________________ >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at 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 at 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 at 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_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nagalenoj at gmail.com Sat Sep 19 02:34:45 2009 From: nagalenoj at gmail.com (Nagalenoj) Date: Sat, 19 Sep 2009 02:34:45 -0700 (PDT) Subject: [Freeswitch-users] Mod_perl or ESL Message-ID: <25520023.post@talk.nabble.com> Dear friends, I want to know which is the better way to do route calls and control calls. I've did a experiment which can be done in both ways, Mod_perl and ESL. I don't know which one is better to take. When I see some earlier posts, It is given like Mod_perl has some limitations and I don't know what kind of limitations they are., Can someone say which is better to use and how it is better? -- View this message in context: http://www.nabble.com/Mod_perl-or-ESL-tp25520023p25520023.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From codecomplete at free.fr Sat Sep 19 03:34:25 2009 From: codecomplete at free.fr (Fred-145) Date: Sat, 19 Sep 2009 03:34:25 -0700 (PDT) Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? Message-ID: <25520404.post@talk.nabble.com> Hello I'm selling a basic solution for SOHO customers (FS is installed on their work computer running Windows or Macs) to handle an analog phone line. When they're on the road, in addition or instead of getting a notification by e-mail when someone calls their office, some users might want to have the Freeswitch server actually ring their cellphone so they can take calls. Besides taking a subscription with a VoIP provider that the Freeswitch server will use to ring their cellphone, I'd like to know what my options are when it comes to setting up a GSM gateway on the customer's premises, in case they don't want to depend on the Internet. Are there Freeswitch-compatible, affordable solutions to handle a single GSM subscription? I guess all it takes is having them take a second subscription with their GSM provider and inserting the SIM chip inside the gateway to have Freeswitch ring their cellphone, but I've never used those things. Thank you. -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25520404.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From aep.lists at it46.se Sat Sep 19 04:47:06 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Sat, 19 Sep 2009 13:47:06 +0200 Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <25520404.post@talk.nabble.com> References: <25520404.post@talk.nabble.com> Message-ID: <883102f54ceed3b94dd3c6362573e280.squirrel@correo.nodo50.org> If you can wait a few weeks, it will be one :) available and documented. /aep -- Stopping junk mailers is good for the environment > > Hello > > I'm selling a basic solution for SOHO customers (FS is installed on their > work computer running Windows or Macs) to handle an analog phone line. > When they're on the road, in addition or instead of getting a notification > by e-mail when someone calls their office, some users might want to have > the > Freeswitch server actually ring their cellphone so they can take calls. > > Besides taking a subscription with a VoIP provider that the Freeswitch > server will use to ring their cellphone, I'd like to know what my options > are when it comes to setting up a GSM gateway on the customer's premises, > in > case they don't want to depend on the Internet. > > Are there Freeswitch-compatible, affordable solutions to handle a single > GSM > subscription? I guess all it takes is having them take a second > subscription > with their GSM provider and inserting the SIM chip inside the gateway to > have Freeswitch ring their cellphone, but I've never used those things. > > Thank you. > -- > View this message in context: > http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25520404.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From mattdfong at gmail.com Sat Sep 19 07:36:43 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Sat, 19 Sep 2009 21:36:43 +0700 Subject: [Freeswitch-users] Trouble Getting session:getVariable("state") in Lua Message-ID: <4256bf830909190736p4bf0a10dla3de7b11096ffcd8@mail.gmail.com> I'm having trouble getting the channel variable state in my Lua ivr example. I have tried both session:getVariable("state") session:getVariable("Channel-State") session:getVariable("answer_state") session:getVariable("Answer-State") but lua reports nil for all attempts I did a uuid_dump and it appears normal....and both Channel-State and Answer-State Variables are present...does anyone know why my Lua IVR can not get these channel variables? Thanks --matt uuid_dump:Event-Name: CHANNEL_DATA Core-UUID: ed5556a8-060f-4ce4-85bb-0a70b08120cf FreeSWITCH-Hostname: matthew-laptop FreeSWITCH-IPv4: 192.168.2.2 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2009-09-19%2012%3A47%3A20 Event-Date-GMT: Sat,%2019%20Sep%202009%2012%3A47%3A20%20GMT Event-Date-Timestamp: 1253364440904749 Event-Calling-File: mod_commands.c Event-Calling-Function: uuid_dump_function Event-Calling-Line-Number: 3298 Channel-State: CS_EXECUTE Channel-State-Number: 4 Channel-Name: sofia/internal/1001 Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 Call-Direction: outbound Presence-Call-Direction: outbound Answer-State: answered Channel-Read-Codec-Name: PCMU Channel-Read-Codec-Rate: 8000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Caller-Caller-ID-Name: FreeSWITCH Caller-Caller-ID-Number: 0000000000 Caller-Network-Addr: 192.168.2.4 Caller-Destination-Number: 1001 Caller-Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 Caller-Source: src/switch_ivr_originate.c Caller-Context: default Caller-Channel-Name: sofia/internal/1001 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1253364439936068 Caller-Channel-Created-Time: 1253364439936068 Caller-Channel-Answered-Time: 1253364440900612 Caller-Channel-Progress-Time: 1253364439976071 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false variable_channel_name: sofia/internal/1001 variable_sip_local_url: 1001%40192.168.2.2 variable_sip_destination_url: %22user%22%20%3Csip%3A1001%40192.168.2.4%3A5061%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1001%2540192.168.2.4%253A5061%3E variable_is_outbound: true variable_ignore_early_media: true variable_originate_early_media: false variable_sip_nat_detected: true variable_sofia_profile_name: internal variable_sip_call_id: 690ad846-1fbd-122d-1599-0010c6ceb785 variable_sip_reply_host: 192.168.2.4 variable_sip_reply_port: 5061 variable_sip_user_agent: Linksys/PAP2T-5.1.6(LS) variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%201231630%201231630%20IN%20IP4%20192.168.2.4%0D%0As%3D-%0D%0Ac%3DIN%20IP4%20192.168.2.4%0D%0At%3D0%200%0D%0Am%3Daudio%2016406%20RTP/AVP%200%20100%20101%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A100%20NSE/8000%0D%0Aa%3Dfmtp%3A100%20192-193%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dptime%3A20%0D%0A variable_remote_media_ip: 192.168.2.4 variable_remote_media_port: 16406 variable_read_codec: PCMU variable_read_rate: 8000 variable_write_codec: PCMU variable_write_rate: 8000 variable_local_media_ip: 192.168.2.2 variable_local_media_port: 20442 variable_endpoint_disposition: ANSWER variable_current_application_data: api_epik_pocket.lua variable_current_application: lua -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090919/70fb043e/attachment.html From tculjaga at gmail.com Sat Sep 19 08:24:13 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sat, 19 Sep 2009 17:24:13 +0200 Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <883102f54ceed3b94dd3c6362573e280.squirrel@correo.nodo50.org> References: <25520404.post@talk.nabble.com> <883102f54ceed3b94dd3c6362573e280.squirrel@correo.nodo50.org> Message-ID: <65d96fc80909190824k50c03defwf4d4cf2af931a1dd@mail.gmail.com> btw, you can check this GW: http://www.edgepbx.cn/shop/index.php?controller=product&product_id=12 i have it on my desk and it works as a charm... T. On Sat, Sep 19, 2009 at 1:47 PM, Alberto Escudero wrote: > If you can wait a few weeks, it will be one :) available and documented. > > /aep > -- > Stopping junk mailers is good for the environment > > > > > Hello > > > > I'm selling a basic solution for SOHO customers (FS is installed on their > > work computer running Windows or Macs) to handle an analog phone line. > > When they're on the road, in addition or instead of getting a > notification > > by e-mail when someone calls their office, some users might want to have > > the > > Freeswitch server actually ring their cellphone so they can take calls. > > > > Besides taking a subscription with a VoIP provider that the Freeswitch > > server will use to ring their cellphone, I'd like to know what my options > > are when it comes to setting up a GSM gateway on the customer's premises, > > in > > case they don't want to depend on the Internet. > > > > Are there Freeswitch-compatible, affordable solutions to handle a single > > GSM > > subscription? I guess all it takes is having them take a second > > subscription > > with their GSM provider and inserting the SIM chip inside the gateway to > > have Freeswitch ring their cellphone, but I've never used those things. > > > > Thank you. > > -- > > View this message in context: > > > http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25520404.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090919/d40798b6/attachment.html From anthony.minessale at gmail.com Sat Sep 19 08:27:40 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 19 Sep 2009 10:27:40 -0500 Subject: [Freeswitch-users] Trouble Getting session:getVariable("state") in Lua In-Reply-To: <4256bf830909190736p4bf0a10dla3de7b11096ffcd8@mail.gmail.com> References: <4256bf830909190736p4bf0a10dla3de7b11096ffcd8@mail.gmail.com> Message-ID: <191c3a030909190827j3c4ba67cxfaf8f0589adc7d32@mail.gmail.com> state is not a variable. I added a session:getState() for you to trunk but I am not sure why you need it. On Sat, Sep 19, 2009 at 9:36 AM, Matthew Fong wrote: > I'm having trouble getting the channel variable state in my Lua ivr > example. > I have tried both > > session:getVariable("state") > session:getVariable("Channel-State") > session:getVariable("answer_state") > session:getVariable("Answer-State") > > but lua reports nil for all attempts > > I did a uuid_dump and it appears normal....and both Channel-State and > Answer-State Variables are present...does anyone know why my Lua IVR can not > get these channel variables? Thanks > > --matt > > uuid_dump:Event-Name: CHANNEL_DATA > Core-UUID: ed5556a8-060f-4ce4-85bb-0a70b08120cf > FreeSWITCH-Hostname: matthew-laptop > FreeSWITCH-IPv4: 192.168.2.2 > FreeSWITCH-IPv6: %3A%3A1 > Event-Date-Local: 2009-09-19%2012%3A47%3A20 > Event-Date-GMT: Sat,%2019%20Sep%202009%2012%3A47%3A20%20GMT > Event-Date-Timestamp: 1253364440904749 > Event-Calling-File: mod_commands.c > Event-Calling-Function: uuid_dump_function > Event-Calling-Line-Number: 3298 > Channel-State: CS_EXECUTE > Channel-State-Number: 4 > Channel-Name: sofia/internal/1001 > Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 > Call-Direction: outbound > Presence-Call-Direction: outbound > Answer-State: answered > Channel-Read-Codec-Name: PCMU > Channel-Read-Codec-Rate: 8000 > Channel-Write-Codec-Name: PCMU > Channel-Write-Codec-Rate: 8000 > Caller-Caller-ID-Name: FreeSWITCH > Caller-Caller-ID-Number: 0000000000 > Caller-Network-Addr: 192.168.2.4 > Caller-Destination-Number: 1001 > Caller-Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 > Caller-Source: src/switch_ivr_originate.c > Caller-Context: default > Caller-Channel-Name: sofia/internal/1001 > Caller-Profile-Index: 1 > Caller-Profile-Created-Time: 1253364439936068 > Caller-Channel-Created-Time: 1253364439936068 > Caller-Channel-Answered-Time: 1253364440900612 > Caller-Channel-Progress-Time: 1253364439976071 > Caller-Channel-Progress-Media-Time: 0 > Caller-Channel-Hangup-Time: 0 > Caller-Channel-Transfer-Time: 0 > Caller-Screen-Bit: true > Caller-Privacy-Hide-Name: false > Caller-Privacy-Hide-Number: false > variable_channel_name: sofia/internal/1001 > variable_sip_local_url: 1001%40192.168.2.2 > variable_sip_destination_url: > %22user%22%20%3Csip%3A1001%40192.168.2.4%3A5061%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1001%2540192.168.2.4%253A5061%3E > variable_is_outbound: true > variable_ignore_early_media: true > variable_originate_early_media: false > variable_sip_nat_detected: true > variable_sofia_profile_name: internal > variable_sip_call_id: 690ad846-1fbd-122d-1599-0010c6ceb785 > variable_sip_reply_host: 192.168.2.4 > variable_sip_reply_port: 5061 > variable_sip_user_agent: Linksys/PAP2T-5.1.6(LS) > variable_switch_r_sdp: > v%3D0%0D%0Ao%3D-%201231630%201231630%20IN%20IP4%20192.168.2.4%0D%0As%3D-%0D%0Ac%3DIN%20IP4%20192.168.2.4%0D%0At%3D0%200%0D%0Am%3Daudio%2016406%20RTP/AVP%200%20100%20101%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A100%20NSE/8000%0D%0Aa%3Dfmtp%3A100%20192-193%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dptime%3A20%0D%0A > variable_remote_media_ip: 192.168.2.4 > variable_remote_media_port: 16406 > variable_read_codec: PCMU > variable_read_rate: 8000 > variable_write_codec: PCMU > variable_write_rate: 8000 > variable_local_media_ip: 192.168.2.2 > variable_local_media_port: 20442 > variable_endpoint_disposition: ANSWER > variable_current_application_data: api_epik_pocket.lua > variable_current_application: lua > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090919/913f7206/attachment-0001.html From jmesquita at freeswitch.org Sat Sep 19 09:21:30 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sat, 19 Sep 2009 13:21:30 -0300 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <20090918185559.GJ20978@hijacked.us> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail.com> <20090918185559.GJ20978@hijacked.us> Message-ID: Andrew, I am sorry for forgetting about you. This is exactly why asked if you were you on IRC the other day... Can you tell me if this is going to stay open source when production ready? jmesquita On 9/18/09, Andrew Thompson wrote: > On Thu, Sep 17, 2009 at 11:20:22AM -0700, Michael Collins wrote: >> I was curious about this myself. Even if someone has built a non-free >> skills-based ACD using FS I'd like to know about it. >> -MC > > I guess nobody paid any attention to my Cluecon presentation... :( > > http://wiki.opencsm.org/wiki/index.php/Spice_Telephony is a skill-based > ACD that uses FS for its voice components. I havent pimped it here in > quite a while but here's some of its major features > > * Skill based routing > * Priority Queues (instead of just FIFO) > * Multiple call types (voice, voicemail and email are currently > supported, instant message support (via libpurple) is prototyped) > * Outbound call support (no autodialer though) > * Distributed system so you can aggregate multiple FS > instances/locations into one big 'virtual' callcenter > * Web-based agent and administrative interface > > There's quite a bit more, but that's the overview. The project is > finally approaching a 1.0 after over a year of development - I hope to > deploy it in production sometime around the end of this year or the > beginning of 2010 (replacing my previous custom asterisk solution). > > You can grab the code at > http://git.opencsm.org/index.cgi/spice-telephony/ (you can browse or > git clone that URL). All you should need to run it is a modern erlang > release (R12B5 or newer) and ruby/rake to run the build. > > Andrew > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sent from my mobile device From matt at hellohunter.com Sat Sep 19 09:41:49 2009 From: matt at hellohunter.com (Matt Hunter) Date: Sat, 19 Sep 2009 23:41:49 +0700 Subject: [Freeswitch-users] Trouble Getting session:getVariable("state") in Lua In-Reply-To: <191c3a030909190827j3c4ba67cxfaf8f0589adc7d32@mail.gmail.com> References: <4256bf830909190736p4bf0a10dla3de7b11096ffcd8@mail.gmail.com> <191c3a030909190827j3c4ba67cxfaf8f0589adc7d32@mail.gmail.com> Message-ID: <4256bf830909190941g515b46ecr476cb4ad7196a162@mail.gmail.com> I think this is probably also the problem that this user on Jira thought was a bug at http://jira.freeswitch.org/browse/MODLANG-128 Anyway, thanks! I had wanted the state of the channel because after hang-up of a channel being controlled by a lua script, the script continues executing. My lua script has a few loops, so if a caller hangups during a loop, the lua script never exits (gets caught in the loop). So I was trying to get the state variable to see if the call still exists, and if not exist the loop and close the lua script. Is there an easier way that I'm missing to accomplish this? Also when using onInput and a dtmf_callback within a luascript, you can interrupt a session:sleep and/or a playmsg, but it seems once the onInput execution is finished, the sleep and playmsg continue. Is the correct method to have the onInput return break; to stop the old sleep and playmsg from Q'ing? Thanks so much. --matt On Sat, Sep 19, 2009 at 10:27 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > state is not a variable. > I added a session:getState() for you to trunk but I am not sure why you > need it. > > > On Sat, Sep 19, 2009 at 9:36 AM, Matthew Fong wrote: > >> I'm having trouble getting the channel variable state in my Lua ivr >> example. >> I have tried both >> >> session:getVariable("state") >> session:getVariable("Channel-State") >> session:getVariable("answer_state") >> session:getVariable("Answer-State") >> >> but lua reports nil for all attempts >> >> I did a uuid_dump and it appears normal....and both Channel-State and >> Answer-State Variables are present...does anyone know why my Lua IVR can not >> get these channel variables? Thanks >> >> --matt >> >> uuid_dump:Event-Name: CHANNEL_DATA >> Core-UUID: ed5556a8-060f-4ce4-85bb-0a70b08120cf >> FreeSWITCH-Hostname: matthew-laptop >> FreeSWITCH-IPv4: 192.168.2.2 >> FreeSWITCH-IPv6: %3A%3A1 >> Event-Date-Local: 2009-09-19%2012%3A47%3A20 >> Event-Date-GMT: Sat,%2019%20Sep%202009%2012%3A47%3A20%20GMT >> Event-Date-Timestamp: 1253364440904749 >> Event-Calling-File: mod_commands.c >> Event-Calling-Function: uuid_dump_function >> Event-Calling-Line-Number: 3298 >> Channel-State: CS_EXECUTE >> Channel-State-Number: 4 >> Channel-Name: sofia/internal/1001 >> Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 >> Call-Direction: outbound >> Presence-Call-Direction: outbound >> Answer-State: answered >> Channel-Read-Codec-Name: PCMU >> Channel-Read-Codec-Rate: 8000 >> Channel-Write-Codec-Name: PCMU >> Channel-Write-Codec-Rate: 8000 >> Caller-Caller-ID-Name: FreeSWITCH >> Caller-Caller-ID-Number: 0000000000 >> Caller-Network-Addr: 192.168.2.4 >> Caller-Destination-Number: 1001 >> Caller-Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 >> Caller-Source: src/switch_ivr_originate.c >> Caller-Context: default >> Caller-Channel-Name: sofia/internal/1001 >> Caller-Profile-Index: 1 >> Caller-Profile-Created-Time: 1253364439936068 >> Caller-Channel-Created-Time: 1253364439936068 >> Caller-Channel-Answered-Time: 1253364440900612 >> Caller-Channel-Progress-Time: 1253364439976071 >> Caller-Channel-Progress-Media-Time: 0 >> Caller-Channel-Hangup-Time: 0 >> Caller-Channel-Transfer-Time: 0 >> Caller-Screen-Bit: true >> Caller-Privacy-Hide-Name: false >> Caller-Privacy-Hide-Number: false >> variable_channel_name: sofia/internal/1001 >> variable_sip_local_url: 1001%40192.168.2.2 >> variable_sip_destination_url: >> %22user%22%20%3Csip%3A1001%40192.168.2.4%3A5061%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1001%2540192.168.2.4%253A5061%3E >> variable_is_outbound: true >> variable_ignore_early_media: true >> variable_originate_early_media: false >> variable_sip_nat_detected: true >> variable_sofia_profile_name: internal >> variable_sip_call_id: 690ad846-1fbd-122d-1599-0010c6ceb785 >> variable_sip_reply_host: 192.168.2.4 >> variable_sip_reply_port: 5061 >> variable_sip_user_agent: Linksys/PAP2T-5.1.6(LS) >> variable_switch_r_sdp: >> v%3D0%0D%0Ao%3D-%201231630%201231630%20IN%20IP4%20192.168.2.4%0D%0As%3D-%0D%0Ac%3DIN%20IP4%20192.168.2.4%0D%0At%3D0%200%0D%0Am%3Daudio%2016406%20RTP/AVP%200%20100%20101%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A100%20NSE/8000%0D%0Aa%3Dfmtp%3A100%20192-193%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dptime%3A20%0D%0A >> variable_remote_media_ip: 192.168.2.4 >> variable_remote_media_port: 16406 >> variable_read_codec: PCMU >> variable_read_rate: 8000 >> variable_write_codec: PCMU >> variable_write_rate: 8000 >> variable_local_media_ip: 192.168.2.2 >> variable_local_media_port: 20442 >> variable_endpoint_disposition: ANSWER >> variable_current_application_data: api_epik_pocket.lua >> variable_current_application: lua >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090919/2e5859dc/attachment.html From mattdfong at gmail.com Sat Sep 19 09:42:10 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Sat, 19 Sep 2009 23:42:10 +0700 Subject: [Freeswitch-users] Trouble Getting session:getVariable("state") in Lua In-Reply-To: <191c3a030909190827j3c4ba67cxfaf8f0589adc7d32@mail.gmail.com> References: <4256bf830909190736p4bf0a10dla3de7b11096ffcd8@mail.gmail.com> <191c3a030909190827j3c4ba67cxfaf8f0589adc7d32@mail.gmail.com> Message-ID: <4256bf830909190942g26f781e0y41e0e93f8cd393db@mail.gmail.com> I think this is probably also the problem that this user on Jira thought was a bug at http://jira.freeswitch.org/browse/MODLANG-128 Anyway, thanks! I had wanted the state of the channel because after hang-up of a channel being controlled by a lua script, the script continues executing. My lua script has a few loops, so if a caller hangups during a loop, the lua script never exits (gets caught in the loop). So I was trying to get the state variable to see if the call still exists, and if not exist the loop and close the lua script. Is there an easier way that I'm missing to accomplish this? Also when using onInput and a dtmf_callback within a luascript, you can interrupt a session:sleep and/or a playmsg, but it seems once the onInput execution is finished, the sleep and playmsg continue. Is the correct method to have the onInput return break; to stop the old sleep and playmsg from Q'ing? Thanks so much. --matt On Sat, Sep 19, 2009 at 10:27 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > state is not a variable. > I added a session:getState() for you to trunk but I am not sure why you > need it. > > > On Sat, Sep 19, 2009 at 9:36 AM, Matthew Fong wrote: > >> I'm having trouble getting the channel variable state in my Lua ivr >> example. >> I have tried both >> >> session:getVariable("state") >> session:getVariable("Channel-State") >> session:getVariable("answer_state") >> session:getVariable("Answer-State") >> >> but lua reports nil for all attempts >> >> I did a uuid_dump and it appears normal....and both Channel-State and >> Answer-State Variables are present...does anyone know why my Lua IVR can not >> get these channel variables? Thanks >> >> --matt >> >> uuid_dump:Event-Name: CHANNEL_DATA >> Core-UUID: ed5556a8-060f-4ce4-85bb-0a70b08120cf >> FreeSWITCH-Hostname: matthew-laptop >> FreeSWITCH-IPv4: 192.168.2.2 >> FreeSWITCH-IPv6: %3A%3A1 >> Event-Date-Local: 2009-09-19%2012%3A47%3A20 >> Event-Date-GMT: Sat,%2019%20Sep%202009%2012%3A47%3A20%20GMT >> Event-Date-Timestamp: 1253364440904749 >> Event-Calling-File: mod_commands.c >> Event-Calling-Function: uuid_dump_function >> Event-Calling-Line-Number: 3298 >> Channel-State: CS_EXECUTE >> Channel-State-Number: 4 >> Channel-Name: sofia/internal/1001 >> Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 >> Call-Direction: outbound >> Presence-Call-Direction: outbound >> Answer-State: answered >> Channel-Read-Codec-Name: PCMU >> Channel-Read-Codec-Rate: 8000 >> Channel-Write-Codec-Name: PCMU >> Channel-Write-Codec-Rate: 8000 >> Caller-Caller-ID-Name: FreeSWITCH >> Caller-Caller-ID-Number: 0000000000 >> Caller-Network-Addr: 192.168.2.4 >> Caller-Destination-Number: 1001 >> Caller-Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 >> Caller-Source: src/switch_ivr_originate.c >> Caller-Context: default >> Caller-Channel-Name: sofia/internal/1001 >> Caller-Profile-Index: 1 >> Caller-Profile-Created-Time: 1253364439936068 >> Caller-Channel-Created-Time: 1253364439936068 >> Caller-Channel-Answered-Time: 1253364440900612 >> Caller-Channel-Progress-Time: 1253364439976071 >> Caller-Channel-Progress-Media-Time: 0 >> Caller-Channel-Hangup-Time: 0 >> Caller-Channel-Transfer-Time: 0 >> Caller-Screen-Bit: true >> Caller-Privacy-Hide-Name: false >> Caller-Privacy-Hide-Number: false >> variable_channel_name: sofia/internal/1001 >> variable_sip_local_url: 1001%40192.168.2.2 >> variable_sip_destination_url: >> %22user%22%20%3Csip%3A1001%40192.168.2.4%3A5061%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1001%2540192.168.2.4%253A5061%3E >> variable_is_outbound: true >> variable_ignore_early_media: true >> variable_originate_early_media: false >> variable_sip_nat_detected: true >> variable_sofia_profile_name: internal >> variable_sip_call_id: 690ad846-1fbd-122d-1599-0010c6ceb785 >> variable_sip_reply_host: 192.168.2.4 >> variable_sip_reply_port: 5061 >> variable_sip_user_agent: Linksys/PAP2T-5.1.6(LS) >> variable_switch_r_sdp: >> v%3D0%0D%0Ao%3D-%201231630%201231630%20IN%20IP4%20192.168.2.4%0D%0As%3D-%0D%0Ac%3DIN%20IP4%20192.168.2.4%0D%0At%3D0%200%0D%0Am%3Daudio%2016406%20RTP/AVP%200%20100%20101%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A100%20NSE/8000%0D%0Aa%3Dfmtp%3A100%20192-193%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dptime%3A20%0D%0A >> variable_remote_media_ip: 192.168.2.4 >> variable_remote_media_port: 16406 >> variable_read_codec: PCMU >> variable_read_rate: 8000 >> variable_write_codec: PCMU >> variable_write_rate: 8000 >> variable_local_media_ip: 192.168.2.2 >> variable_local_media_port: 20442 >> variable_endpoint_disposition: ANSWER >> variable_current_application_data: api_epik_pocket.lua >> variable_current_application: lua >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090919/bdfe0382/attachment-0001.html From anthony.minessale at gmail.com Sat Sep 19 09:47:12 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 19 Sep 2009 11:47:12 -0500 Subject: [Freeswitch-users] Trouble Getting session:getVariable("state") in Lua In-Reply-To: <4256bf830909190941g515b46ecr476cb4ad7196a162@mail.gmail.com> References: <4256bf830909190736p4bf0a10dla3de7b11096ffcd8@mail.gmail.com> <191c3a030909190827j3c4ba67cxfaf8f0589adc7d32@mail.gmail.com> <4256bf830909190941g515b46ecr476cb4ad7196a162@mail.gmail.com> Message-ID: <191c3a030909190947j2fcbb43at23252b1feeb0654@mail.gmail.com> you should always check session:ready() in all loops if session:ready() fails it means you must exit your script because the call has either been transferred or hungup. On Sat, Sep 19, 2009 at 11:41 AM, Matt Hunter wrote: > I think this is probably also the problem that this user on Jira thought > was a bug at > http://jira.freeswitch.org/browse/MODLANG-128 > > Anyway, thanks! > > I had wanted the state of the channel because after hang-up of a channel > being controlled by a lua script, the script continues executing. My lua > script has a few loops, so if a caller hangups during a loop, the lua script > never exits (gets caught in the loop). So I was trying to get the state > variable to see if the call still exists, and if not exist the loop and > close the lua script. > > Is there an easier way that I'm missing to accomplish this? > > Also when using onInput and a dtmf_callback within a luascript, you can > interrupt a session:sleep and/or a playmsg, but it seems once the onInput > execution is finished, the sleep and playmsg continue. Is the correct method > to have the onInput return break; to stop the old sleep and playmsg from > Q'ing? > > Thanks so much. > > --matt > > > On Sat, Sep 19, 2009 at 10:27 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> state is not a variable. >> I added a session:getState() for you to trunk but I am not sure why you >> need it. >> >> >> On Sat, Sep 19, 2009 at 9:36 AM, Matthew Fong wrote: >> >>> I'm having trouble getting the channel variable state in my Lua ivr >>> example. >>> I have tried both >>> >>> session:getVariable("state") >>> session:getVariable("Channel-State") >>> session:getVariable("answer_state") >>> session:getVariable("Answer-State") >>> >>> but lua reports nil for all attempts >>> >>> I did a uuid_dump and it appears normal....and both Channel-State and >>> Answer-State Variables are present...does anyone know why my Lua IVR can not >>> get these channel variables? Thanks >>> >>> --matt >>> >>> uuid_dump:Event-Name: CHANNEL_DATA >>> Core-UUID: ed5556a8-060f-4ce4-85bb-0a70b08120cf >>> FreeSWITCH-Hostname: matthew-laptop >>> FreeSWITCH-IPv4: 192.168.2.2 >>> FreeSWITCH-IPv6: %3A%3A1 >>> Event-Date-Local: 2009-09-19%2012%3A47%3A20 >>> Event-Date-GMT: Sat,%2019%20Sep%202009%2012%3A47%3A20%20GMT >>> Event-Date-Timestamp: 1253364440904749 >>> Event-Calling-File: mod_commands.c >>> Event-Calling-Function: uuid_dump_function >>> Event-Calling-Line-Number: 3298 >>> Channel-State: CS_EXECUTE >>> Channel-State-Number: 4 >>> Channel-Name: sofia/internal/1001 >>> Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 >>> Call-Direction: outbound >>> Presence-Call-Direction: outbound >>> Answer-State: answered >>> Channel-Read-Codec-Name: PCMU >>> Channel-Read-Codec-Rate: 8000 >>> Channel-Write-Codec-Name: PCMU >>> Channel-Write-Codec-Rate: 8000 >>> Caller-Caller-ID-Name: FreeSWITCH >>> Caller-Caller-ID-Number: 0000000000 >>> Caller-Network-Addr: 192.168.2.4 >>> Caller-Destination-Number: 1001 >>> Caller-Unique-ID: 12ee98af-d76d-483c-b9a9-59e7f08ca4e9 >>> Caller-Source: src/switch_ivr_originate.c >>> Caller-Context: default >>> Caller-Channel-Name: sofia/internal/1001 >>> Caller-Profile-Index: 1 >>> Caller-Profile-Created-Time: 1253364439936068 >>> Caller-Channel-Created-Time: 1253364439936068 >>> Caller-Channel-Answered-Time: 1253364440900612 >>> Caller-Channel-Progress-Time: 1253364439976071 >>> Caller-Channel-Progress-Media-Time: 0 >>> Caller-Channel-Hangup-Time: 0 >>> Caller-Channel-Transfer-Time: 0 >>> Caller-Screen-Bit: true >>> Caller-Privacy-Hide-Name: false >>> Caller-Privacy-Hide-Number: false >>> variable_channel_name: sofia/internal/1001 >>> variable_sip_local_url: 1001%40192.168.2.2 >>> variable_sip_destination_url: >>> %22user%22%20%3Csip%3A1001%40192.168.2.4%3A5061%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1001%2540192.168.2.4%253A5061%3E >>> variable_is_outbound: true >>> variable_ignore_early_media: true >>> variable_originate_early_media: false >>> variable_sip_nat_detected: true >>> variable_sofia_profile_name: internal >>> variable_sip_call_id: 690ad846-1fbd-122d-1599-0010c6ceb785 >>> variable_sip_reply_host: 192.168.2.4 >>> variable_sip_reply_port: 5061 >>> variable_sip_user_agent: Linksys/PAP2T-5.1.6(LS) >>> variable_switch_r_sdp: >>> v%3D0%0D%0Ao%3D-%201231630%201231630%20IN%20IP4%20192.168.2.4%0D%0As%3D-%0D%0Ac%3DIN%20IP4%20192.168.2.4%0D%0At%3D0%200%0D%0Am%3Daudio%2016406%20RTP/AVP%200%20100%20101%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A100%20NSE/8000%0D%0Aa%3Dfmtp%3A100%20192-193%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0Aa%3Dptime%3A20%0D%0A >>> variable_remote_media_ip: 192.168.2.4 >>> variable_remote_media_port: 16406 >>> variable_read_codec: PCMU >>> variable_read_rate: 8000 >>> variable_write_codec: PCMU >>> variable_write_rate: 8000 >>> variable_local_media_ip: 192.168.2.2 >>> variable_local_media_port: 20442 >>> variable_endpoint_disposition: ANSWER >>> variable_current_application_data: api_epik_pocket.lua >>> variable_current_application: lua >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090919/2a4b1bd8/attachment.html From klaus.teller at gmx.net Sat Sep 19 19:14:56 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Sun, 20 Sep 2009 04:14:56 +0200 Subject: [Freeswitch-users] Call Tracing Message-ID: <20090920021456.250460@gmx.net> Hi, Say i have an inbound VoIP/SIP call that hits my FS box. Is it possible to to extract information about the intermediate hops that the call or the signaling went through? If so, what information can i get? Thanks, Gregoire. -- Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 - sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser From jmesquita at freeswitch.org Sat Sep 19 21:03:02 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 20 Sep 2009 01:03:02 -0300 Subject: [Freeswitch-users] mod_nibblebill Message-ID: Guys, I have been testing mod_nibblebill lately and there are 2 params that I could not make work. Looking at code, I could not find a single line that would actually test those. Is this confirmed to be implemented? If not, this should be removed from the configs so it won't get ppl lured. Regards, jmesquita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090920/b9be92fd/attachment.html From tculjaga at gmail.com Sun Sep 20 01:33:01 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sun, 20 Sep 2009 10:33:01 +0200 Subject: [Freeswitch-users] Call Tracing In-Reply-To: <20090920021456.250460@gmx.net> References: <20090920021456.250460@gmx.net> Message-ID: <65d96fc80909200133q1875708ewdbbcf49290295b9@mail.gmail.com> switch.conf.xml (btw: in console you can enable/disable logging on the fly - F8/F7) your relevant sip profile: T. On Sun, Sep 20, 2009 at 4:14 AM, Klaus Teller wrote: > Hi, > > Say i have an inbound VoIP/SIP call that hits my FS box. Is it possible to > to extract information about the intermediate hops that the call or the > signaling went through? If so, what information can i get? > > Thanks, > Gregoire. > -- > Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 - > sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090920/2c3ec820/attachment-0001.html From mcampbellsmith at gmail.com Sun Sep 20 02:50:16 2009 From: mcampbellsmith at gmail.com (Mark Campbell-Smith) Date: Sun, 20 Sep 2009 19:50:16 +1000 Subject: [Freeswitch-users] Unknown call drops.. INFO DTMF(3) In-Reply-To: <87f2f3b90909181327p78361ffflfabe3c57edab01f2@mail.gmail.com> References: <33c87fa30909130234u21541ae7o61c522d014e3dacd@mail.gmail.com> <87f2f3b90909181327p78361ffflfabe3c57edab01f2@mail.gmail.com> Message-ID: <33c87fa30909200250x55c8158bt796b97a9278fdf5c@mail.gmail.com> The problem hasn't been seen again, but the exact call case has not been performed again. Will update freeswitch and monitor next time. Cheers MCS On Sat, Sep 19, 2009 at 6:27 AM, Michael Collins wrote: > Is this still happening? If so please make sure that you are on latest trunk > and re-test. Get a pcap of the traffic (SIP and RTP) for review and then > report back. > > Thanks, > MC > > On Sun, Sep 13, 2009 at 2:34 AM, Mark Campbell-Smith > wrote: >> >> Hi! >> >> I have just experienced some call drops and each time the sequence is >> the same in the freeswitch.log file. ?Both parties are sure that they >> did not accidentally hit the 3 button to send the DTMF tone (and the >> same thing has happened four times already after ~5 minutes). >> >> 2009-09-13 19:28:23.835216 [DEBUG] sofia.c:4329 INFO DTMF(3) >> 2009-09-13 19:28:23.835216 [DEBUG] sofia.c:4450 dispatched freeswitch >> event for INFO >> 2009-09-13 19:28:23.859408 [DEBUG] switch_rtp.c:1624 Send start packet >> for [3] ts=2591120 dur=160/160/13120 seq=64923 >> 2009-09-13 19:28:23.879439 [DEBUG] switch_rtp.c:1560 Send middle >> packet for [3] ts=2591120 dur=320/320/13120 seq=64924 >> 2009-09-13 19:28:23.899455 [DEBUG] switch_rtp.c:1560 Send middle >> packet for [3] ts=2591120 dur=480/480/13120 seq=64925 >> : >> : >> 2009-09-13 19:28:25.439404 [DEBUG] switch_rtp.c:1560 Send middle >> packet for [3] ts=2591120 dur=12800/12800/13120 seq=65002 >> 2009-09-13 19:28:25.459312 [DEBUG] switch_rtp.c:1560 Send middle >> packet for [3] ts=2591120 dur=12960/12960/13120 seq=65003 >> 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet >> for [3] ts=2591120 dur=13120/13120/13120 seq=65004 >> 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet >> for [3] ts=2591120 dur=13120/13120/13120 seq=65005 >> 2009-09-13 19:28:25.479183 [DEBUG] switch_rtp.c:1560 Send end packet >> for [3] ts=2591120 dur=13120/13120/13120 seq=65006 >> 2009-09-13 19:28:33.879341 [NOTICE] sofia.c:322 Hangup >> sofia/external/ [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] >> 2009-09-13 19:28:33.879341 [DEBUG] switch_channel.c:1683 Send signal >> sofia/external/ [KILL] >> 2009-09-13 19:28:33.879341 [DEBUG] switch_core_session.c:932 Send >> signal sofia/external/ [BREAK] >> 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:371 >> sofia/external/ ending bridge by request from write function >> 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:426 >> sofia/internal_nat/1000 at 192.168.1.120 receive message [UNBRIDGE] >> 2009-09-13 19:28:33.900940 [DEBUG] switch_core_session.c:630 Send >> signal sofia/internal_nat/1000 at 192.168.1.120 [BREAK] >> 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:452 BRIDGE >> THREAD DONE [sofia/internal_nat/1000 at 192.168.1.120] >> 2009-09-13 19:28:33.900940 [DEBUG] switch_ivr_bridge.c:454 Send signal >> sofia/external/ [BREAK] >> 2009-09-13 19:28:33.912049 [NOTICE] switch_core_state_machine.c:179 >> Hangup sofia/internal_nat/1000 at 192.168.1.120 [CS_EXECUTE] >> [NORMAL_CLEARING] >> >> Anyone have any idea what this sequence means and why I am getting >> this? ?Is it my sip provider or something in FreeSwitch? ?What does >> the 'Send end packet for [3] ts=2591120 dur=13120/13120/13120 >> seq=65006' mean? ?Notice that dur (duration?) is increasing a lot >> until the call drops. >> >> Thanks! >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From klaus.teller at gmx.net Sun Sep 20 03:44:16 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Sun, 20 Sep 2009 12:44:16 +0200 Subject: [Freeswitch-users] Call Tracing In-Reply-To: <65d96fc80909200133q1875708ewdbbcf49290295b9@mail.gmail.com> References: <20090920021456.250460@gmx.net> <65d96fc80909200133q1875708ewdbbcf49290295b9@mail.gmail.com> Message-ID: <20090920104416.32730@gmx.net> Hi T., I just tried that but i don't see anything different on the console. My test call is going via callcentric and les.net, but besides the final hop which i normally see in the channel name, there is nothing else. Any idea what i might be doing wrong here? Thanks, Klaus. -------- Original-Nachricht -------- > Datum: Sun, 20 Sep 2009 10:33:01 +0200 > Von: Tihomir Culjaga > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] Call Tracing > switch.conf.xml (btw: in console you can enable/disable logging on the fly > - > F8/F7) > > > > > your relevant sip profile: > > > > T. > > > On Sun, Sep 20, 2009 at 4:14 AM, Klaus Teller > wrote: > > > Hi, > > > > Say i have an inbound VoIP/SIP call that hits my FS box. Is it possible > to > > to extract information about the intermediate hops that the call or the > > signaling went through? If so, what information can i get? > > > > Thanks, > > Gregoire. > > -- > > Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 > - > > sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 From shaheryarkh at googlemail.com Sun Sep 20 04:11:50 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Sun, 20 Sep 2009 17:11:50 +0600 Subject: [Freeswitch-users] Call Tracing In-Reply-To: <20090920104416.32730@gmx.net> References: <20090920021456.250460@gmx.net> <65d96fc80909200133q1875708ewdbbcf49290295b9@mail.gmail.com> <20090920104416.32730@gmx.net> Message-ID: there are a few variable that you can set in /usr/local/freeswitch/conf/vars.xml. * * You can change it to something like (and then restart FS), * * Usually it will give you enough information about call processing, however just in case you are looking for SIP trace of a call only then you can enable it on per-profile basis at run-time, for example, *sofia profile internal siptrace on* this will enable SIP trace for all calls to / from sofia internal profile (which also includes directory users). You can run following command on FS console to get information on what profile etc. are available as well as their status. *sofia status* For more info consult Wiki page at, http://wiki.freeswitch.org/wiki/Sofia Thank you. On Sun, Sep 20, 2009 at 4:44 PM, Klaus Teller wrote: > Hi T., > > I just tried that but i don't see anything different on the console. My > test call is going via callcentric and les.net, but besides the final hop > which i normally see in the channel name, there is nothing else. > > Any idea what i might be doing wrong here? > > Thanks, > Klaus. > -------- Original-Nachricht -------- > > Datum: Sun, 20 Sep 2009 10:33:01 +0200 > > Von: Tihomir Culjaga > > An: freeswitch-users at lists.freeswitch.org > > Betreff: Re: [Freeswitch-users] Call Tracing > > > switch.conf.xml (btw: in console you can enable/disable logging on the > fly > > - > > F8/F7) > > > > > > > > > > your relevant sip profile: > > > > > > > > T. > > > > > > On Sun, Sep 20, 2009 at 4:14 AM, Klaus Teller > > wrote: > > > > > Hi, > > > > > > Say i have an inbound VoIP/SIP call that hits my FS box. Is it possible > > to > > > to extract information about the intermediate hops that the call or the > > > signaling went through? If so, what information can i get? > > > > > > Thanks, > > > Gregoire. > > > -- > > > Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox > 3 > > - > > > sicherer, schneller und einfacher! > http://portal.gmx.net/de/go/chbrowser > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > -- > GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090920/92496ee2/attachment.html From klaus.teller at gmx.net Sun Sep 20 04:49:08 2009 From: klaus.teller at gmx.net (Klaus Teller) Date: Sun, 20 Sep 2009 13:49:08 +0200 Subject: [Freeswitch-users] Call Tracing In-Reply-To: References: <20090920021456.250460@gmx.net> <65d96fc80909200133q1875708ewdbbcf49290295b9@mail.gmail.com> <20090920104416.32730@gmx.net> Message-ID: <20090920114908.250460@gmx.net> Thanks. I tried that and what it shows me is the trace between my peer and the SIP provider (i.e. les.net). The call is actually coming from callcentric and i don't see that in the trace. Is it supposed to show this? Klaus. -------- Original-Nachricht -------- > Datum: Sun, 20 Sep 2009 17:11:50 +0600 > Von: Muhammad Shahzad > An: freeswitch-users at lists.freeswitch.org > Betreff: Re: [Freeswitch-users] Call Tracing > there are a few variable that you can set in > /usr/local/freeswitch/conf/vars.xml. > > * > > * > You can change it to something like (and then restart FS), > > * > > * > Usually it will give you enough information about call processing, however > just in case you are looking for SIP trace of a call only then you can > enable it on per-profile basis at run-time, > > for example, > > *sofia profile internal siptrace on* > > this will enable SIP trace for all calls to / from sofia internal profile > (which also includes directory users). > > You can run following command on FS console to get information on what > profile etc. are available as well as their status. > > *sofia status* > > For more info consult Wiki page at, > > http://wiki.freeswitch.org/wiki/Sofia > > Thank you. > > > On Sun, Sep 20, 2009 at 4:44 PM, Klaus Teller > wrote: > > > Hi T., > > > > I just tried that but i don't see anything different on the console. My > > test call is going via callcentric and les.net, but besides the final > hop > > which i normally see in the channel name, there is nothing else. > > > > Any idea what i might be doing wrong here? > > > > Thanks, > > Klaus. > > -------- Original-Nachricht -------- > > > Datum: Sun, 20 Sep 2009 10:33:01 +0200 > > > Von: Tihomir Culjaga > > > An: freeswitch-users at lists.freeswitch.org > > > Betreff: Re: [Freeswitch-users] Call Tracing > > > > > switch.conf.xml (btw: in console you can enable/disable logging on the > > fly > > > - > > > F8/F7) > > > > > > > > > > > > > > > your relevant sip profile: > > > > > > > > > > > > T. > > > > > > > > > On Sun, Sep 20, 2009 at 4:14 AM, Klaus Teller > > > wrote: > > > > > > > Hi, > > > > > > > > Say i have an inbound VoIP/SIP call that hits my FS box. Is it > possible > > > to > > > > to extract information about the intermediate hops that the call or > the > > > > signaling went through? If so, what information can i get? > > > > > > > > Thanks, > > > > Gregoire. > > > > -- > > > > Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla > Firefox > > 3 > > > - > > > > sicherer, schneller und einfacher! > > http://portal.gmx.net/de/go/chbrowser > > > > > > > > _______________________________________________ > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > -- > > GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > > Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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_786pk at hotmail.com > Email: shaheryarkh at googlemail.com -- GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 From frank at impactfax.com Sun Sep 20 09:10:20 2009 From: frank at impactfax.com (Frank @ Impact) Date: Sun, 20 Sep 2009 12:10:20 -0400 Subject: [Freeswitch-users] Bind extention to a different Dialplan andcdr php? In-Reply-To: <87f2f3b90909080037q7027bb1cme36aa68dcc42c882@mail.gmail.com> Message-ID: <639179436F2F4CAF9151191AF779B663@ws4> Currently all incoming calls to my FS to all extensions are sent off by curl to a particular php script called dialplan.php. I would like to have certain extensions that are called to have their xml dialplan built by a curl to a different php script, say dialplan2.php. Is there a way to have certain extensions get their dialplan by calling a different php script other than the default? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, September 08, 2009 3:37 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Bind extention to a different Dialplan andcdr php? On Sun, Sep 6, 2009 at 8:43 AM, Frank @ Impact wrote: Is there a way to bind a particular extension to a different dialplan php and a different cdr php script than the default one? Could you re-phrase this question with a bit more detail? Thanks. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090920/e4815e65/attachment-0001.html From jmesquita at freeswitch.org Sun Sep 20 09:40:07 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 20 Sep 2009 13:40:07 -0300 Subject: [Freeswitch-users] Bind extention to a different Dialplan andcdr php? In-Reply-To: <639179436F2F4CAF9151191AF779B663@ws4> References: <87f2f3b90909080037q7027bb1cme36aa68dcc42c882@mail.gmail.com> <639179436F2F4CAF9151191AF779B663@ws4> Message-ID: Frank, That kind of logic needs to be performed at your application, if I am not wrong. I will do some testing here, but I think that mod_xml_cURL sends purpose and other information regarding the type of data it is requesting so you can decide on your application exactly what to do. Regards, jmesquita On Sun, Sep 20, 2009 at 1:10 PM, Frank @ Impact wrote: > Currently all incoming calls to my FS to all extensions are sent off by > curl to a particular php script called dialplan.php. > > I would like to have certain extensions that are called to have their xml > dialplan built by a curl to a different php script, say dialplan2.php. > > > > Is there a way to have certain extensions get their dialplan by calling a > different php script other than the default? > > > > -----Original Message----- > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Tuesday, September 08, 2009 3:37 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] Bind extention to a different Dialplan > andcdr php? > > > > > > On Sun, Sep 6, 2009 at 8:43 AM, Frank @ Impact > wrote: > > Is there a way to bind a particular extension to a different dialplan php > and a different cdr php script than the default one? > > > > Could you re-phrase this question with a bit more detail? Thanks. > -MC > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090920/b15eca29/attachment.html From sprice at gmail.com Sun Sep 20 09:49:17 2009 From: sprice at gmail.com (SP) Date: Sun, 20 Sep 2009 11:49:17 -0500 Subject: [Freeswitch-users] Bind extention to a different Dialplan and cdr php? Message-ID: <7e2ac3270909200949y41fc65b7qb3b63f5a5750e75a@mail.gmail.com> you can use different php includes On Sunday, September 20, 2009, Jo?o Mesquita wrote: > Frank, > > That kind of logic needs to be performed at your application, if I am not wrong. I will do some testing here, but I think that mod_xml_cURL sends purpose and other information regarding the type of data it is requesting so you can decide on your application exactly what to do. > > Regards, > > jmesquita > > On Sun, Sep 20, 2009 at 1:10 PM, Frank @ Impact > wrote: > > > > > > > > > > > > > > > > > > > > > Currently all incoming calls to my FS to > all extensions are sent off by curl to a particular php > script called dialplan.php. > > I would like to have certain extensions > that are called to have their xml dialplan built by a > curl to a different php script, say dialplan2.php. > > > > Is there a way to have certain extensions > get their dialplan by calling a different php script other than the default? > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins > Sent: Tuesday, September 08, 2009 > 3:37 AM > To: > freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] > Bind extention to a different Dialplan andcdr php? > > > > > > > > On Sun, Sep 6, 2009 at 8:43 AM, Frank @ Impact > wrote: > > > > > > > Is there a way to bind a particular extension to a > different dialplan php and a different cdr php script than the default one? > > > > > > > > > > Could you re-phrase this question with a bit more > detail? Thanks. > -MC > > > > > > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- Shannon From R.Kloosterman at mtel.nl Sun Sep 20 11:42:40 2009 From: R.Kloosterman at mtel.nl (Remko Kloosterman) Date: Sun, 20 Sep 2009 20:42:40 +0200 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <20090918185559.GJ20978@hijacked.us> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com><87f2f3b90909171120m1827ed5fmbba68ae973feb3e1@mail.gmail .com> <20090918185559.GJ20978@hijacked.us> Message-ID: <11372C8B9E603F4FACDE6AB18256DEC695A9EF@srvmtel.office.mtel.nl> This actually sounds very good Andrew. You even have an agent interface. Do you have plans for a outbound campaign dialer? I know of a commercial dialer that is good in it's predictive algotithm, but very bad when it comes to campaign management. -----Oorspronkelijk bericht----- Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Andrew Thompson Verzonden: vrijdag 18 september 2009 20:56 Aan: freeswitch-users at lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] skill-based ACD On Thu, Sep 17, 2009 at 11:20:22AM -0700, Michael Collins wrote: > I was curious about this myself. Even if someone has built a non-free > skills-based ACD using FS I'd like to know about it. > -MC I guess nobody paid any attention to my Cluecon presentation... :( http://wiki.opencsm.org/wiki/index.php/Spice_Telephony is a skill-based ACD that uses FS for its voice components. I havent pimped it here in quite a while but here's some of its major features * Skill based routing * Priority Queues (instead of just FIFO) * Multiple call types (voice, voicemail and email are currently supported, instant message support (via libpurple) is prototyped) * Outbound call support (no autodialer though) * Distributed system so you can aggregate multiple FS instances/locations into one big 'virtual' callcenter * Web-based agent and administrative interface There's quite a bit more, but that's the overview. The project is finally approaching a 1.0 after over a year of development - I hope to deploy it in production sometime around the end of this year or the beginning of 2010 (replacing my previous custom asterisk solution). You can grab the code at http://git.opencsm.org/index.cgi/spice-telephony/ (you can browse or git clone that URL). All you should need to run it is a modern erlang release (R12B5 or newer) and ruby/rake to run the build. Andrew _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From gavin.henry at gmail.com Sun Sep 20 12:47:05 2009 From: gavin.henry at gmail.com (Gavin Henry) Date: Sun, 20 Sep 2009 20:47:05 +0100 Subject: [Freeswitch-users] Any FreeSWITCH training courses out there? Message-ID: <13ca621c0909201247t509fca27h9b62aa37af7e757b@mail.gmail.com> Hi all, Is there anyone out there doing beginner courses or conversion courses from an Asterisk mindset? Cheers. -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com From jmesquita at freeswitch.org Sun Sep 20 13:07:52 2009 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Sun, 20 Sep 2009 17:07:52 -0300 Subject: [Freeswitch-users] Any FreeSWITCH training courses out there? In-Reply-To: <13ca621c0909201247t509fca27h9b62aa37af7e757b@mail.gmail.com> References: <13ca621c0909201247t509fca27h9b62aa37af7e757b@mail.gmail.com> Message-ID: You could always talk to consulting at freeswitch.org. They can help you with that! ;) And, this is probably best to be sent to the -biz, isn't it? (really asking, not being ironic) jmesquita On Sun, Sep 20, 2009 at 4:47 PM, Gavin Henry wrote: > Hi all, > > Is there anyone out there doing beginner courses or conversion courses > from an Asterisk mindset? > > Cheers. > > -- > Sent from my mobile device > > http://www.suretecsystems.com/services/openldap/ > http://www.suretectelecom.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090920/e3e081c1/attachment.html From sranil at gmail.com Sun Sep 20 13:13:51 2009 From: sranil at gmail.com (Anil Kumar S. R.) Date: Mon, 21 Sep 2009 01:43:51 +0530 Subject: [Freeswitch-users] User Creation with DB in Freeswitch Message-ID: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> Hi All, Is there any way in which user information can be stored in database instead of the xml files. Regards, Anil -- Anil Kumar S. R. http://sranil.googlepages.com/ "The best way to succeed in this world is to act on the advice you give to others." -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/392beffc/attachment.html From brian at freeswitch.org Sun Sep 20 13:24:53 2009 From: brian at freeswitch.org (Brian West) Date: Sun, 20 Sep 2009 15:24:53 -0500 Subject: [Freeswitch-users] User Creation with DB in Freeswitch In-Reply-To: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> References: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> Message-ID: <47C5757E-AB3B-4978-A0E8-3239B36CB39A@freeswitch.org> Using XML CURL http://wiki.freeswitch.org/wiki/Mod_xml_curl /b On Sep 20, 2009, at 3:13 PM, Anil Kumar S. R. wrote: > Hi All, > > Is there any way in which user information can be stored in database > instead of the xml files. > > Regards, > Anil > > -- > Anil Kumar S. R. > http://sranil.googlepages.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090920/3d186c82/attachment.html From gavin.henry at gmail.com Sun Sep 20 13:32:09 2009 From: gavin.henry at gmail.com (Gavin Henry) Date: Sun, 20 Sep 2009 21:32:09 +0100 Subject: [Freeswitch-users] Any FreeSWITCH training courses out there? In-Reply-To: References: <13ca621c0909201247t509fca27h9b62aa37af7e757b@mail.gmail.com> Message-ID: <13ca621c0909201332n3d06186cjb90305cb3bce3c3a@mail.gmail.com> 2009/9/20 Jo?o Mesquita : > You could always talk to consulting at freeswitch.org. They can help you with > that! ;) Thanks. > And, this is probably best to be sent to the -biz, isn't it? (really asking, > not being ironic) Oops, forgot. I'm subscribed to that too. Should have remembered. From codecomplete at free.fr Sun Sep 20 13:57:10 2009 From: codecomplete at free.fr (Fred-145) Date: Sun, 20 Sep 2009 13:57:10 -0700 (PDT) Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <65d96fc80909190824k50c03defwf4d4cf2af931a1dd@mail.gmail.com> References: <25520404.post@talk.nabble.com> <883102f54ceed3b94dd3c6362573e280.squirrel@correo.nodo50.org> <65d96fc80909190824k50c03defwf4d4cf2af931a1dd@mail.gmail.com> Message-ID: <25530130.post@talk.nabble.com> Thanks Tihomir for the link. >From what I read, it appears that EdgePBX's FX02G is a full-fledged Asterisk server with a GSM module and an FXS module. Did you reflash its NAND to run Freeswitch? At $300, I guess customers will rather take a subscription with a VoIP provided and use their GSM gateway, but I'm interested in knowing whether the FX02G can be used as a PSTN/GSM gateway, possibly with FreeSwitch running on that unit as well. Thank you. -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530130.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From tculjaga at gmail.com Sun Sep 20 14:11:29 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sun, 20 Sep 2009 23:11:29 +0200 Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <25530130.post@talk.nabble.com> References: <25520404.post@talk.nabble.com> <883102f54ceed3b94dd3c6362573e280.squirrel@correo.nodo50.org> <65d96fc80909190824k50c03defwf4d4cf2af931a1dd@mail.gmail.com> <25530130.post@talk.nabble.com> Message-ID: <65d96fc80909201411u6aefec64oa75f98fe7707e8ca@mail.gmail.com> hi, well, yes, it should be possible to crosscompile freeswitch on that platofrm... this is a totally different topic and to be honest i really don't see the point doing this. When i did it last time (porting stuff to Blackfin), it took several days of hard work. This is an external device/endpoint to freeswitch. You don't need any FXS ports... it is enough to have the GSM one (or two). Just send calls from FS to FX02 via SIP and that's it. T. On Sun, Sep 20, 2009 at 10:57 PM, Fred-145 wrote: > > Thanks Tihomir for the link. > > >From what I read, it appears that EdgePBX's FX02G is a full-fledged > Asterisk > server with a GSM module and an FXS module. Did you reflash its NAND to run > Freeswitch? > > At $300, I guess customers will rather take a subscription with a VoIP > provided and use their GSM gateway, but I'm interested in knowing whether > the FX02G can be used as a PSTN/GSM gateway, possibly with FreeSwitch > running on that unit as well. > > Thank you. > -- > View this message in context: > http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530130.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090920/49d45758/attachment.html From mitul at enterux.com Sun Sep 20 20:43:41 2009 From: mitul at enterux.com (Mitul Limbani) Date: Mon, 21 Sep 2009 09:13:41 +0530 Subject: [Freeswitch-users] Any FreeSWITCH training courses out there? In-Reply-To: <13ca621c0909201332n3d06186cjb90305cb3bce3c3a@mail.gmail.com> References: <13ca621c0909201247t509fca27h9b62aa37af7e757b@mail.gmail.com> <13ca621c0909201332n3d06186cjb90305cb3bce3c3a@mail.gmail.com> Message-ID: <753D756A-1DBF-440F-B4A7-B5482EEB1475@enterux.com> Gavin, We do have some freeswitch training content and are planning to launch one in India our training facility is withi one of the largest telco in india - MTNL (http://CETTM.MTNL.in) Do let me know what sort of course are you looking for. Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 21-Sep-2009, at 2:02 AM, Gavin Henry wrote: > 2009/9/20 Jo?o Mesquita : >> You could always talk to consulting at freeswitch.org. They can help >> you with >> that! ;) > > Thanks. > >> And, this is probably best to be sent to the -biz, isn't it? >> (really asking, >> not being ironic) > > Oops, forgot. I'm subscribed to that too. Should have remembered. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mitul at enterux.com Sun Sep 20 20:44:31 2009 From: mitul at enterux.com (Mitul Limbani) Date: Mon, 21 Sep 2009 09:14:31 +0530 Subject: [Freeswitch-users] User Creation with DB in Freeswitch In-Reply-To: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> References: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> Message-ID: <566E110B-72CC-428E-AEE8-94AB600587B0@enterux.com> Yes use odbc in fs Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 21-Sep-2009, at 1:43 AM, "Anil Kumar S. R." wrote: > Hi All, > > Is there any way in which user information can be stored in database > instead of the xml files. > > Regards, > Anil > > -- > Anil Kumar S. R. > http://sranil.googlepages.com/ > > "The best way to succeed in this world is to act on the advice you > give to others." > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/24a9e534/attachment.html From mitul at enterux.com Sun Sep 20 20:53:25 2009 From: mitul at enterux.com (Mitul Limbani) Date: Mon, 21 Sep 2009 09:23:25 +0530 Subject: [Freeswitch-users] Any FreeSWITCH training courses out there? In-Reply-To: <13ca621c0909201247t509fca27h9b62aa37af7e757b@mail.gmail.com> References: <13ca621c0909201247t509fca27h9b62aa37af7e757b@mail.gmail.com> Message-ID: <83A73A2C-2AF9-4A9E-B86E-7D32C82867F0@enterux.com> Gavin, Sorry for the earlier mail, I can see that you mentioned Asterisk to Freeswitch course, we have pretty much under gone the same cycle and have put that as the part of our training course, it's named: FreeSWITCH for AstMasters Please do get in touch off the list, also if anyone else is interested in this course do get in touch with me. Thanks & Regards, Mitul Limbani, Founder & CEO, Enterux Solutions Pvt. Ltd., The Enterprise Linux Company (r), http://www.enterux.com http://www.entVoice.com On 21-Sep-2009, at 1:17 AM, Gavin Henry wrote: > Hi all, > > Is there anyone out there doing beginner courses or conversion courses > from an Asterisk mindset? > > Cheers. > > -- > Sent from my mobile device > > http://www.suretecsystems.com/services/openldap/ > http://www.suretectelecom.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From juanbackson at gmail.com Sun Sep 20 20:58:33 2009 From: juanbackson at gmail.com (Juan Backson) Date: Mon, 21 Sep 2009 11:58:33 +0800 Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <65d96fc80909201411u6aefec64oa75f98fe7707e8ca@mail.gmail.com> References: <25520404.post@talk.nabble.com> <883102f54ceed3b94dd3c6362573e280.squirrel@correo.nodo50.org> <65d96fc80909190824k50c03defwf4d4cf2af931a1dd@mail.gmail.com> <25530130.post@talk.nabble.com> <65d96fc80909201411u6aefec64oa75f98fe7707e8ca@mail.gmail.com> Message-ID: <27c25bc40909202058l30736b86p2291dc0cf5dcb49d@mail.gmail.com> Hi, Are you able to have freeswitch working on blackfin platform? thanks, jb On Mon, Sep 21, 2009 at 5:11 AM, Tihomir Culjaga wrote: > hi, > > well, yes, it should be possible to crosscompile freeswitch on that > platofrm... this is a totally different topic and to be honest i really > don't see the point doing this. When i did it last time (porting stuff to > Blackfin), it took several days of hard work. > > This is an external device/endpoint to freeswitch. You don't need any FXS > ports... it is enough to have the GSM one (or two). Just send calls from FS > to FX02 via SIP and that's it. > > > T. > > > On Sun, Sep 20, 2009 at 10:57 PM, Fred-145 wrote: > >> >> Thanks Tihomir for the link. >> >> >From what I read, it appears that EdgePBX's FX02G is a full-fledged >> Asterisk >> server with a GSM module and an FXS module. Did you reflash its NAND to >> run >> Freeswitch? >> >> At $300, I guess customers will rather take a subscription with a VoIP >> provided and use their GSM gateway, but I'm interested in knowing whether >> the FX02G can be used as a PSTN/GSM gateway, possibly with FreeSwitch >> running on that unit as well. >> >> Thank you. >> -- >> View this message in context: >> http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530130.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/dea762a1/attachment.html From brad.tuan at gmail.com Sun Sep 20 21:08:03 2009 From: brad.tuan at gmail.com (Brad Tuan) Date: Mon, 21 Sep 2009 12:08:03 +0800 Subject: [Freeswitch-users] How to set the IP of REGISTER message?? Message-ID: If my PC has two IP address, How to decide the IP which using to send REGISTER message?? For example, I have two IP , 172.30.30.XXX for External and 192.168.60.XXX for Internal, and now i want to add two gateway, one is 210.XXX.XXX.XXX and the other one is 192.168.60.30, How to change the IP that i want to use?? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/4061ce0c/attachment.html From hads at nice.net.nz Sun Sep 20 21:08:05 2009 From: hads at nice.net.nz (Hadley Rich) Date: Mon, 21 Sep 2009 16:08:05 +1200 Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <27c25bc40909202058l30736b86p2291dc0cf5dcb49d@mail.gmail.com> References: <25520404.post@talk.nabble.com> <65d96fc80909201411u6aefec64oa75f98fe7707e8ca@mail.gmail.com> <27c25bc40909202058l30736b86p2291dc0cf5dcb49d@mail.gmail.com> Message-ID: <200909211608.05748.hads@nice.net.nz> On Mon, 21 Sep 2009 15:58:33 Juan Backson wrote: > Are you able to have freeswitch working on blackfin platform? This has been covered many times on the list now, currently the answer is no. hads -- https://nicegear.co.nz VoIP and Open Source Hardware From frank at carmickle.com Sun Sep 20 21:27:54 2009 From: frank at carmickle.com (Frank Carmickle) Date: Mon, 21 Sep 2009 00:27:54 -0400 Subject: [Freeswitch-users] How to set the IP of REGISTER message?? In-Reply-To: References: Message-ID: <20090921042754.GZ30343@base.carmickle.com> Hello Brad On Mon, Sep 21, Brad Tuan wrote: > If my PC has two IP address, How to decide the IP which using to send > REGISTER message?? You need to create different sofia profiles for each. Then you can define the gateway in each. --FC From tculjaga at gmail.com Sun Sep 20 23:41:25 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 21 Sep 2009 08:41:25 +0200 Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <200909211608.05748.hads@nice.net.nz> References: <25520404.post@talk.nabble.com> <65d96fc80909201411u6aefec64oa75f98fe7707e8ca@mail.gmail.com> <27c25bc40909202058l30736b86p2291dc0cf5dcb49d@mail.gmail.com> <200909211608.05748.hads@nice.net.nz> Message-ID: <65d96fc80909202341k5a17dea4l7fe273779a9c4c4@mail.gmail.com> I didn't say i have a working FS on blackfin... i just said i've ported a lot of software to blackfin and it was always ....floating point, fork vs vfork ... main issues... but why do you think it cannot be done? T. On Mon, Sep 21, 2009 at 6:08 AM, Hadley Rich wrote: > On Mon, 21 Sep 2009 15:58:33 Juan Backson wrote: > > Are you able to have freeswitch working on blackfin platform? > > This has been covered many times on the list now, currently the answer is > no. > > hads > -- > https://nicegear.co.nz > VoIP and Open Source Hardware > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/db19c883/attachment.html From shaheryarkh at googlemail.com Mon Sep 21 01:40:31 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 21 Sep 2009 14:40:31 +0600 Subject: [Freeswitch-users] Call Tracing In-Reply-To: <20090920114908.250460@gmx.net> References: <20090920021456.250460@gmx.net> <65d96fc80909200133q1875708ewdbbcf49290295b9@mail.gmail.com> <20090920104416.32730@gmx.net> <20090920114908.250460@gmx.net> Message-ID: In that case you should turn on sip trace for profile where your callcentric peer is configured. By default FS comes with two profiles namely internal and external. If you haven't created any new profile and configured your users and peers in these two profiles then you should try turning on sip trace for external profile too (or just external profile alone). *sofia profile external siptrace on* Please check your peer configuration and turn on sip trace on appropriate profile. Thank you. On Sun, Sep 20, 2009 at 5:49 PM, Klaus Teller wrote: > Thanks. I tried that and what it shows me is the trace between my peer and > the SIP provider (i.e. les.net). The call is actually coming from > callcentric and i don't see that in the trace. Is it supposed to show this? > > Klaus. > > -------- Original-Nachricht -------- > > Datum: Sun, 20 Sep 2009 17:11:50 +0600 > > Von: Muhammad Shahzad > > An: freeswitch-users at lists.freeswitch.org > > Betreff: Re: [Freeswitch-users] Call Tracing > > > there are a few variable that you can set in > > /usr/local/freeswitch/conf/vars.xml. > > > > * > > > > * > > You can change it to something like (and then restart FS), > > > > * > > > > * > > Usually it will give you enough information about call processing, > however > > just in case you are looking for SIP trace of a call only then you can > > enable it on per-profile basis at run-time, > > > > for example, > > > > *sofia profile internal siptrace on* > > > > this will enable SIP trace for all calls to / from sofia internal profile > > (which also includes directory users). > > > > You can run following command on FS console to get information on what > > profile etc. are available as well as their status. > > > > *sofia status* > > > > For more info consult Wiki page at, > > > > http://wiki.freeswitch.org/wiki/Sofia > > > > Thank you. > > > > > > On Sun, Sep 20, 2009 at 4:44 PM, Klaus Teller > > wrote: > > > > > Hi T., > > > > > > I just tried that but i don't see anything different on the console. My > > > test call is going via callcentric and les.net, but besides the final > > hop > > > which i normally see in the channel name, there is nothing else. > > > > > > Any idea what i might be doing wrong here? > > > > > > Thanks, > > > Klaus. > > > -------- Original-Nachricht -------- > > > > Datum: Sun, 20 Sep 2009 10:33:01 +0200 > > > > Von: Tihomir Culjaga > > > > An: freeswitch-users at lists.freeswitch.org > > > > Betreff: Re: [Freeswitch-users] Call Tracing > > > > > > > switch.conf.xml (btw: in console you can enable/disable logging on > the > > > fly > > > > - > > > > F8/F7) > > > > > > > > > > > > > > > > > > > > your relevant sip profile: > > > > > > > > > > > > > > > > T. > > > > > > > > > > > > On Sun, Sep 20, 2009 at 4:14 AM, Klaus Teller > > > > wrote: > > > > > > > > > Hi, > > > > > > > > > > Say i have an inbound VoIP/SIP call that hits my FS box. Is it > > possible > > > > to > > > > > to extract information about the intermediate hops that the call or > > the > > > > > signaling went through? If so, what information can i get? > > > > > > > > > > Thanks, > > > > > Gregoire. > > > > > -- > > > > > Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla > > Firefox > > > 3 > > > > - > > > > > sicherer, schneller und einfacher! > > > http://portal.gmx.net/de/go/chbrowser > > > > > > > > > > _______________________________________________ > > > > > FreeSWITCH-users mailing list > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE: > > > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > http://www.freeswitch.org > > > > > > > > > > > -- > > > GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > > > Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > > > > > > _______________________________________________ > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at 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_786pk at hotmail.com > > Email: shaheryarkh at googlemail.com > > -- > GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT! > Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/fe227ec8/attachment.html From codecomplete at free.fr Mon Sep 21 01:56:29 2009 From: codecomplete at free.fr (Fred-145) Date: Mon, 21 Sep 2009 01:56:29 -0700 (PDT) Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <65d96fc80909190824k50c03defwf4d4cf2af931a1dd@mail.gmail.com> References: <25520404.post@talk.nabble.com> <883102f54ceed3b94dd3c6362573e280.squirrel@correo.nodo50.org> <65d96fc80909190824k50c03defwf4d4cf2af931a1dd@mail.gmail.com> Message-ID: <25530241.post@talk.nabble.com> Or as a more affordable solution... is it possible to connect an entry-level GSM phone to a PC running Freeswitch and use this as a poor man's gateway? -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530241.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From shaheryarkh at googlemail.com Mon Sep 21 02:28:04 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 21 Sep 2009 15:28:04 +0600 Subject: [Freeswitch-users] Any FreeSWITCH training courses out there? In-Reply-To: <83A73A2C-2AF9-4A9E-B86E-7D32C82867F0@enterux.com> References: <13ca621c0909201247t509fca27h9b62aa37af7e757b@mail.gmail.com> <83A73A2C-2AF9-4A9E-B86E-7D32C82867F0@enterux.com> Message-ID: With help from Pakistan Software Export Board (PSEB), we formed Asterisk Pakistan community forum in early 2008. This forum is still active and we arranged many workshops during last 18 months in all major cities of Pakistan. It was a great success and we effectively introduced Asterisk in so many government and private sectors. FreeSWITCH is very new in Pakistan and a very few people have heard its name here right now. So, we (me and some of my friends from Pakistan Open Source Software Foundation) are trying to develop some skilled personals for FreeSWITCH, before we approach Ministry of Information Technology to launch a campaign similar to Asterisk Pakistan Forum for FreeSWITCH. So, that if our proposal gets approval we would have enough resources to execute workshops all over Pakistan for FS training. All people in this mailing list (especially Pakistanis) who are interested in this, may contact me off list for participation and coordination in these efforts. The goal is to secure greatest share for Pakistan in this newly emerging technology and its benefits. Thank you. On Mon, Sep 21, 2009 at 9:53 AM, Mitul Limbani wrote: > Gavin, > > Sorry for the earlier mail, I can see that you mentioned Asterisk to > Freeswitch course, we have pretty much under gone the same cycle and > have put that as the part of our training course, it's named: > FreeSWITCH for AstMasters > > Please do get in touch off the list, also if anyone else is interested > in this course do get in touch with me. > > Thanks & Regards, > Mitul Limbani, > Founder & CEO, > Enterux Solutions Pvt. Ltd., > The Enterprise Linux Company (r), > http://www.enterux.com > http://www.entVoice.com > > On 21-Sep-2009, at 1:17 AM, Gavin Henry wrote: > > > Hi all, > > > > Is there anyone out there doing beginner courses or conversion courses > > from an Asterisk mindset? > > > > Cheers. > > > > -- > > Sent from my mobile device > > > > http://www.suretecsystems.com/services/openldap/ > > http://www.suretectelecom.com > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/82276531/attachment-0001.html From brian at freeswitch.org Mon Sep 21 02:29:59 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Sep 2009 04:29:59 -0500 Subject: [Freeswitch-users] How to set the IP of REGISTER message?? In-Reply-To: <20090921042754.GZ30343@base.carmickle.com> References: <20090921042754.GZ30343@base.carmickle.com> Message-ID: No you no longer have to do this. Please refer to the internal.xml profile in the default config. If you set the local-network-acl and then set ext-sip-ip and ext-rtp-ip then the profile will figure out which IP to use based on the destination or source of the request/ response. /b On Sep 20, 2009, at 11:27 PM, Frank Carmickle wrote: > You need to create different sofia profiles for each. Then you can > define the gateway in each. > > --FC From brian at freeswitch.org Mon Sep 21 02:30:41 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Sep 2009 04:30:41 -0500 Subject: [Freeswitch-users] Mod_perl or ESL In-Reply-To: <25520023.post@talk.nabble.com> References: <25520023.post@talk.nabble.com> Message-ID: <5BB3DC87-3475-457A-853D-FD477604B0D9@freeswitch.org> How far do you want things to scale? /b On Sep 19, 2009, at 4:34 AM, Nagalenoj wrote: > > Dear friends, > I want to know which is the better way to do route calls and > control > calls. I've did a experiment which can be done in both ways, > Mod_perl and > ESL. I don't know which one is better to take. > When I see some earlier posts, It is given like Mod_perl has some > limitations and I don't know what kind of limitations they are., From brian at freeswitch.org Mon Sep 21 02:32:40 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Sep 2009 04:32:40 -0500 Subject: [Freeswitch-users] Not able to make call using external profile In-Reply-To: <65d96fc80909190004v6a0ec669h3c63db00c2762360@mail.gmail.com> References: <809ad7ab0909180313n5538c1c0ve1b8070c0d1da2cc@mail.gmail.com> <809ad7ab0909180711v73f15899v8e9dc64cb191e82c@mail.gmail.com> <809ad7ab0909182250u6fbfa25pe871bcdb7dcc91c@mail.gmail.com> <65d96fc80909190004v6a0ec669h3c63db00c2762360@mail.gmail.com> Message-ID: If you refer to the latest internal.xml in the default config for sip profiles you'll see an example of how to use a single profile for phones inside and outside of NAT. So you no longer have to have two profiles thus cutting the confusion level to almost zero when you setup FreeSWITCH to talk inside and outside of nat. Key elements are local-network-acl, ext-sip-ip and ext-rtp-ip and you're all set. /b On Sep 19, 2009, at 2:04 AM, Tihomir Culjaga wrote: > another issue you might have with RTP so check the wiki for NAT > config as well. From brian at freeswitch.org Mon Sep 21 02:42:32 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Sep 2009 04:42:32 -0500 Subject: [Freeswitch-users] User Creation with DB in Freeswitch In-Reply-To: <566E110B-72CC-428E-AEE8-94AB600587B0@enterux.com> References: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> <566E110B-72CC-428E-AEE8-94AB600587B0@enterux.com> Message-ID: <347D3032-F778-46F4-AECE-94DECD27E849@freeswitch.org> You can't put the users directly into a db with FreeSWITCH you'll have to serve up the XML document via XML CURL or write your own module to do so via the module interfaces provided. /b On Sep 20, 2009, at 10:44 PM, Mitul Limbani wrote: > Yes use odbc in fs > > Thanks & Regards, > Mitul Limbani, > Founder & CEO, > Enterux Solutions Pvt. Ltd., > The Enterprise Linux Company (r), > http://www.enterux.com > http://www.entVoice.com > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/9662fb35/attachment.html From frank at carmickle.com Mon Sep 21 02:45:39 2009 From: frank at carmickle.com (Frank Carmickle) Date: Mon, 21 Sep 2009 05:45:39 -0400 Subject: [Freeswitch-users] How to set the IP of REGISTER message?? In-Reply-To: References: <20090921042754.GZ30343@base.carmickle.com> Message-ID: <20090921094539.GB30343@base.carmickle.com> On Mon, Sep 21, Brian West wrote: > No you no longer have to do this. Please refer to the internal.xml > profile in the default config. If you set the local-network-acl and > then set ext-sip-ip and ext-rtp-ip then the profile will figure out > which IP to use based on the destination or source of the request/ > response. With two interfaces? Isn't it required for both of the interfaces to be bound to sofia? If that is true then isn't it only possible to bind one address per profile? --FC From m.krivushin at imarto.net Mon Sep 21 02:55:26 2009 From: m.krivushin at imarto.net (Mikhail Krivushin) Date: Mon, 21 Sep 2009 16:55:26 +0700 Subject: [Freeswitch-users] Can I set "From:" field in originate command? Message-ID: <5be734a50909210255m6149e7eag31b5674fb9a76f27@mail.gmail.com> -- ? ?????????, ???????? ?????? ?. ????? ???. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME at jabber.ru skype: mkrivushin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/d0643784/attachment.html From shaheryarkh at googlemail.com Mon Sep 21 03:07:12 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 21 Sep 2009 16:07:12 +0600 Subject: [Freeswitch-users] User Creation with DB in Freeswitch In-Reply-To: <347D3032-F778-46F4-AECE-94DECD27E849@freeswitch.org> References: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> <566E110B-72CC-428E-AEE8-94AB600587B0@enterux.com> <347D3032-F778-46F4-AECE-94DECD27E849@freeswitch.org> Message-ID: Yes, XML CURL is really cool. I once bombed FS with 12,000 wrong registrations (bad username or password) in less then 50 seconds (49496 ms to be exact) and it processed all of them and gave correct responses using XML CURL. I am willing to do this test again soon, with correct registration data this time, to see how many registration Sofia SIP module configured with XML CURL module can handle at a time. Thank you. On Mon, Sep 21, 2009 at 3:42 PM, Brian West wrote: > You can't put the users directly into a db with FreeSWITCH you'll have to > serve up the XML document via XML CURL or write your own module to do so via > the module interfaces provided. > /b > > On Sep 20, 2009, at 10:44 PM, Mitul Limbani wrote: > > Yes use odbc in fs > > Thanks & Regards,Mitul Limbani, > Founder & CEO, > Enterux Solutions Pvt. Ltd., > The Enterprise Linux Company (r), > http://www.enterux.com > http://www.entVoice.com > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/7d358e40/attachment.html From shaheryarkh at googlemail.com Mon Sep 21 03:27:39 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Mon, 21 Sep 2009 16:27:39 +0600 Subject: [Freeswitch-users] User Creation with DB in Freeswitch In-Reply-To: References: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> <566E110B-72CC-428E-AEE8-94AB600587B0@enterux.com> <347D3032-F778-46F4-AECE-94DECD27E849@freeswitch.org> Message-ID: I searched my sent emails and found the results, copying it below (after removing some sensitive info), 1,000 Calls ================================================== Total 1000 REGISTER calls sent in 890 ms at rate of 1123/sec Total 1000 responses receieved in 4516 ms at rate of 221/sec: Detailed responses received: - 403 responses: 1000 (Forbidden) ------ TOTAL responses: 1000 (rate=221/sec) Maximum outstanding job: 894 Peak memory size: 15MB 5,000 Calls ================================================== Total 5000 REGISTER calls sent in 28539 ms at rate of 175/sec Total 5000 responses receieved in 36398 ms at rate of 137/sec: Detailed responses received: - 403 responses: 5000 (Forbidden) ------ TOTAL responses: 5000 (rate=137/sec) Maximum outstanding job: 1001 Peak memory size: 63MB 10,000 Calls ================================================== Total 10000 REGISTER calls sent in 60741 ms at rate of 164/sec Total 9289 responses receieved in 62740 ms at rate of 148/sec: Detailed responses received: - 403 responses: 9289 (Forbidden) ------ TOTAL responses: 9289 (rate=148/sec) Maximum outstanding job: 1047 Peak memory size: 78MB 12,000 Calls ================================================== Total 12000 REGISTER calls sent in 49496 ms at rate of 242/sec Total 12314 responses receieved in 60582 ms at rate of 203/sec: Detailed responses received: - 403 responses: 12314 (Forbidden) ------ TOTAL responses: 12314 (rate=203/sec) Maximum outstanding job: 1018 Peak memory size: 143MB So, FS doesn't crash even on 12,000 bad registrations (600 regs per second). I did tweak its configurations a little however no change was made to source code to make this happen. :-) Thank you. On Mon, Sep 21, 2009 at 4:07 PM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > Yes, XML CURL is really cool. I once bombed FS with 12,000 wrong > registrations (bad username or password) in less then 50 seconds (49496 ms > to be exact) and it processed all of them and gave correct responses using > XML CURL. > > I am willing to do this test again soon, with correct registration data > this time, to see how many registration Sofia SIP module configured with XML > CURL module can handle at a time. > > Thank you. > > > On Mon, Sep 21, 2009 at 3:42 PM, Brian West wrote: > >> You can't put the users directly into a db with FreeSWITCH you'll have to >> serve up the XML document via XML CURL or write your own module to do so via >> the module interfaces provided. >> /b >> >> On Sep 20, 2009, at 10:44 PM, Mitul Limbani wrote: >> >> Yes use odbc in fs >> >> Thanks & Regards,Mitul Limbani, >> Founder & CEO, >> Enterux Solutions Pvt. Ltd., >> The Enterprise Linux Company (r), >> http://www.enterux.com >> http://www.entVoice.com >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/4950369d/attachment-0001.html From m.krivushin at imarto.net Mon Sep 21 03:44:40 2009 From: m.krivushin at imarto.net (Mikhail Krivushin) Date: Mon, 21 Sep 2009 17:44:40 +0700 Subject: [Freeswitch-users] Can I set "From:" field in originate command? In-Reply-To: <5be734a50909210255m6149e7eag31b5674fb9a76f27@mail.gmail.com> References: <5be734a50909210255m6149e7eag31b5674fb9a76f27@mail.gmail.com> Message-ID: <5be734a50909210344w329981bdj48aadbb33bc03187@mail.gmail.com> I see that call_id_number placed in From:, but with wrong realm. I need a way for change realm in From:. Is any ability to do that? (I need to make calls over some telco from different accounts.) 2009/9/21 Mikhail Krivushin > > > -- > ? ?????????, ???????? ?????? > ?. ????? ???. +7 913 865 78 66 > icq: 218 744 127 > xmpp: KrivushinME at jabber.ru > skype: mkrivushin > -- ? ?????????, ???????? ?????? ?. ????? ???. +7 913 865 78 66 icq: 218 744 127 xmpp: KrivushinME at jabber.ru skype: mkrivushin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/11a431b5/attachment.html From tculjaga at gmail.com Mon Sep 21 03:44:45 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 21 Sep 2009 12:44:45 +0200 Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <25530241.post@talk.nabble.com> References: <25520404.post@talk.nabble.com> <883102f54ceed3b94dd3c6362573e280.squirrel@correo.nodo50.org> <65d96fc80909190824k50c03defwf4d4cf2af931a1dd@mail.gmail.com> <25530241.post@talk.nabble.com> Message-ID: <65d96fc80909210344y6ce0f2eanc58ada98d59de500@mail.gmail.com> its a waste of time ... i doubt it can be done. T. On Mon, Sep 21, 2009 at 10:56 AM, Fred-145 wrote: > > Or as a more affordable solution... is it possible to connect an > entry-level > GSM phone to a PC running Freeswitch and use this as a poor man's gateway? > -- > View this message in context: > http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530241.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/65eca7e9/attachment.html From demuel at thephinix.org Mon Sep 21 04:14:16 2009 From: demuel at thephinix.org (demuel at thephinix.org) Date: Mon, 21 Sep 2009 12:14:16 +0100 Subject: [Freeswitch-users] User Creation with DB in Freeswitch In-Reply-To: References: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> <566E110B-72CC-428E-AEE8-94AB600587B0@enterux.com> <347D3032-F778-46F4-AECE-94DECD27E849@freeswitch.org> Message-ID: Whoah what a term -> " ONCE BOMBED FS..."???? > Yes, XML CURL is really cool. I once bombed FS with 12,000 wrong > registrations (bad username or password) in less then 50 seconds (49496 ms > to be exact) and it processed all of them and gave correct responses using > XML CURL. > > I am willing to do this test again soon, with correct registration data this > time, to see how many registration Sofia SIP module configured with XML CURL > module can handle at a time. > > Thank you. > > > On Mon, Sep 21, 2009 at 3:42 PM, Brian West wrote: > >> You can't put the users directly into a db with FreeSWITCH you'll have to >> serve up the XML document via XML CURL or write your own module to do so via >> the module interfaces provided. >> /b >> >> On Sep 20, 2009, at 10:44 PM, Mitul Limbani wrote: >> >> Yes use odbc in fs >> >> Thanks & Regards,Mitul Limbani, >> Founder & CEO, >> Enterux Solutions Pvt. Ltd., >> The Enterprise Linux Company (r), >> http://www.enterux.com >> http://www.entVoice.com >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From nik.middleton at noblesolutions.co.uk Mon Sep 21 04:15:18 2009 From: nik.middleton at noblesolutions.co.uk (Nik Middleton) Date: Mon, 21 Sep 2009 12:15:18 +0100 Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <25520404.post@talk.nabble.com> References: <25520404.post@talk.nabble.com> Message-ID: Check out this range http://www.noblesolutions.co.uk/shop/index.php?main_page=index&manufactu rers_id=16 You should be able to find a local supplier We've used them for a couple of years now. They just plug into your network. Regards, -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fred-145 Sent: 19 September 2009 11:34 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? Hello I'm selling a basic solution for SOHO customers (FS is installed on their work computer running Windows or Macs) to handle an analog phone line. When they're on the road, in addition or instead of getting a notification by e-mail when someone calls their office, some users might want to have the Freeswitch server actually ring their cellphone so they can take calls. Besides taking a subscription with a VoIP provider that the Freeswitch server will use to ring their cellphone, I'd like to know what my options are when it comes to setting up a GSM gateway on the customer's premises, in case they don't want to depend on the Internet. Are there Freeswitch-compatible, affordable solutions to handle a single GSM subscription? I guess all it takes is having them take a second subscription with their GSM provider and inserting the SIM chip inside the gateway to have Freeswitch ring their cellphone, but I've never used those things. Thank you. -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp255204 04p25520404.html Sent from the Freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Mon Sep 21 06:37:10 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Sep 2009 08:37:10 -0500 Subject: [Freeswitch-users] How to set the IP of REGISTER message?? In-Reply-To: <20090921094539.GB30343@base.carmickle.com> References: <20090921042754.GZ30343@base.carmickle.com> <20090921094539.GB30343@base.carmickle.com> Message-ID: <70B2423A-8A7F-4D7A-BE9E-D40CEA41140A@freeswitch.org> yes but you can lie about IP's in the via/to and from if you set the local-network-acl ... I'm not talking two physical interfaces on FreeSWITCH... because that is one of the harder scenarios to setup... I'm talking single interface on FS sitting behind a nat router which is the most common. /b On Sep 21, 2009, at 4:45 AM, Frank Carmickle wrote: > With two interfaces? Isn't it required for both of the interfaces > to be bound to sofia? If that is true then isn't it only possible > to bind one address per profile? > > --FC From brian at freeswitch.org Mon Sep 21 06:39:25 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Sep 2009 08:39:25 -0500 Subject: [Freeswitch-users] Can I set "From:" field in originate command? In-Reply-To: <5be734a50909210344w329981bdj48aadbb33bc03187@mail.gmail.com> References: <5be734a50909210255m6149e7eag31b5674fb9a76f27@mail.gmail.com> <5be734a50909210344w329981bdj48aadbb33bc03187@mail.gmail.com> Message-ID: <9D725C34-4281-4582-8B87-B2FB1EE831C5@freeswitch.org> well first off you would setup a gateway and set the param 'from- domain' to what you wish it to be. /b On Sep 21, 2009, at 5:44 AM, Mikhail Krivushin wrote: > I see that call_id_number placed in From:, but with wrong realm. I > need a way for change realm in From:. Is any ability to do that? > > (I need to make calls over some telco from different accounts.) From pabx_freeswitch at telenet.be Mon Sep 21 09:51:31 2009 From: pabx_freeswitch at telenet.be (henkoegema) Date: Mon, 21 Sep 2009 09:51:31 -0700 (PDT) Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <25520404.post@talk.nabble.com> References: <25520404.post@talk.nabble.com> Message-ID: <25530400.post@talk.nabble.com> Fred-145 wrote: > > Hello > > I'm selling a basic solution for SOHO customers (FS is installed on their > work computer running Windows or Macs) to handle an analog phone line. > When they're on the road, in addition or instead of getting a notification > by e-mail when someone calls their office, some users might want to have > the Freeswitch server actually ring their cellphone so they can take > calls. > > Besides taking a subscription with a VoIP provider that the Freeswitch > server will use to ring their cellphone, I'd like to know what my options > are when it comes to setting up a GSM gateway on the customer's premises, > in case they don't want to depend on the Internet. > > Are there Freeswitch-compatible, affordable solutions to handle a single > GSM subscription? I guess all it takes is having them take a second > subscription with their GSM provider and inserting the SIM chip inside the > gateway to have Freeswitch ring their cellphone, but I've never used those > things. > > Thank you. > I have been using http://www.portech.com.tw/p3-product1_1.asp?Pid=13 for years with Asterisk and Freeswitch. -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530400.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From andrew at hijacked.us Mon Sep 21 09:56:01 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Mon, 21 Sep 2009 12:56:01 -0400 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <11372C8B9E603F4FACDE6AB18256DEC695A9EF@srvmtel.office.mtel.nl> References: <20090918185559.GJ20978@hijacked.us> <11372C8B9E603F4FACDE6AB18256DEC695A9EF@srvmtel.office.mtel.nl> Message-ID: <20090921165601.GA665@hijacked.us> On Sun, Sep 20, 2009 at 08:42:40PM +0200, Remko Kloosterman wrote: > This actually sounds very good Andrew. You even have an agent interface. > > Do you have plans for a outbound campaign dialer? I know of a commercial > dialer that is good in it's predictive algotithm, but very bad when it > comes to campaign management. > I don't have plans for an 'autodialer' in the traditional sense but I do have plans for some sort of campaign dialer - the idea is to use an API to load numbers to be called into a queue and the agents will just pop those stub calls off the queue and then the system will originate the call to the indicated number. This does mean that you'll be wasting agent time on voicemail/ringouts/whatever but hopefully you'll piss less people off. In addition, then you can farm out the system that decides the numbers to be called and in which order to an external system. An autodialer would certainly be possible under the current system, I just don't really care to implement one. Patches accepted, although really an autodialer might be better off remaining a binary-only module add-on (to prevent the doing of evil becoming too cheap :) ). And yes, to my knowledge it will remain under an open-source license for the forseeable future. Andrew From luismzuccolo at yahoo.com.ar Mon Sep 21 10:06:52 2009 From: luismzuccolo at yahoo.com.ar (Luis Manuel Zuccolo) Date: Mon, 21 Sep 2009 10:06:52 -0700 (PDT) Subject: [Freeswitch-users] Compile error In-Reply-To: <19FE43B3-7C34-49C5-8E10-DA66B66197BA@freeswitch.org> References: <1253162070.3583.14.camel@localhost.localdomain> <20090917054213.GH30343@base.carmickle.com> <19FE43B3-7C34-49C5-8E10-DA66B66197BA@freeswitch.org> Message-ID: <881903.13614.qm@web53103.mail.re2.yahoo.com> I' ve get the same error with a fresh tree Thanks in advance ________________________________ De: Brian West Para: freeswitch-users at lists.freeswitch.org Enviado: jueves 17 de septiembre de 2009, 10:12:36 Asunto: Re: [Freeswitch-users] Compile error NO you must not. The issue has been fixed in svn already please start with a fresh tree. /b PS: end users should NEVER have to reswig. On Sep 17, 2009, at 12:42 AM, Frank Carmickle wrote: On Thu, Sep 17, Luis M. Zuccolo wrote: > >Hi: >> > >> >Since svn version 13523 to current I get this error: >> > >> >make[5]: swig: Command not found >> >You must install swig. If your on debian apt-get install swig. If your not see http://www.swig.org/ > >HTH >--FC > Yahoo! Cocina Encontra las mejores recetas con Yahoo! Cocina. http://ar.mujer.yahoo.com/cocina/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/3f80ca6e/attachment-0001.html From sranil at gmail.com Mon Sep 21 11:27:34 2009 From: sranil at gmail.com (Anil Kumar S. R.) Date: Mon, 21 Sep 2009 23:57:34 +0530 Subject: [Freeswitch-users] Displaying matched extension during a call Message-ID: <1b2118200909211127i19998a9dm8d92078e04d21ac1@mail.gmail.com> Hi All, I am new to Freeswitch. So please bear with me if I ask any silly questions. * Can anyone of you please tell me how to display the extension name which has matched an incoming/outgoing call. * And can you please elaborate what does '' mean. * Suppose we have set a variable in the extension of the dialplan XML. Is there anyway we can display this variable on CLI for our debugging purposes. Regards, -- Anil Kumar S. R. http://sranil.googlepages.com/ "The best way to succeed in this world is to act on the advice you give to others." -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/72402e0c/attachment.html From frank at carmickle.com Mon Sep 21 11:59:00 2009 From: frank at carmickle.com (Frank Carmickle) Date: Mon, 21 Sep 2009 14:59:00 -0400 Subject: [Freeswitch-users] Displaying matched extension during a call In-Reply-To: <1b2118200909211127i19998a9dm8d92078e04d21ac1@mail.gmail.com> References: <1b2118200909211127i19998a9dm8d92078e04d21ac1@mail.gmail.com> Message-ID: <20090921185900.GG30343@base.carmickle.com> Hello Anil On Mon, Sep 21, Anil Kumar S. R. wrote: > * Can anyone of you please tell me how to display the extension name which > has matched an incoming/outgoing call. In the log you will find something like this 2009-09-21 14:36:15.574827 [INFO] mod_dialplan_xml.c:315 Processing fs->03977304 in context default > * And can you please elaborate what does '' > mean. Please see http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_info HTH --FC From svetikvoip at gmail.com Mon Sep 21 12:13:54 2009 From: svetikvoip at gmail.com (Svetik VOIP) Date: Mon, 21 Sep 2009 15:13:54 -0400 Subject: [Freeswitch-users] No ring tone while recording incoming call. Please help. Message-ID: <94790b850909211213j52aae598v861dd8ab2e9e7983@mail.gmail.com> Hi, I have trouble recording incoming calls with FreeSwitch. I have followed the instruction from Misc. Dialplan Tools record session (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session) It works well for outgoing calls, but I have the problem with incoming calls. The person who is calling does not hear ring tone, he hears just the silence until I pick up the phone. Everything else is working, we can talk, conversation is recorded. Here is a copy of my dialplan for incoming calls /usr/local/freeswitch/conf/dialplan/public/voipms.xml for outcoming calls I have a similar code added to the /usr/local/freeswitch/conf/dialplan/default/user1.xml and it works well. I have tried to move the line between the lines References: <94790b850909211213j52aae598v861dd8ab2e9e7983@mail.gmail.com> Message-ID: set ringback before record_session and also set transfer_ringback because record_session causes an pre-answer. /b On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote: > Hi, > > I have trouble recording incoming calls with FreeSwitch. > > I have followed the instruction from Misc. Dialplan Tools record > session > (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session) > It works well for outgoing calls, but I have the problem with > incoming calls. > > The person who is calling does not hear ring tone, he hears just the > silence until > I pick up the phone. Everything else is working, we can talk, > conversation is recorded. > > Here is a copy of my dialplan for incoming calls > /usr/local/freeswitch/conf/dialplan/public/voipms.xml > > > > expression="XXXXXXXXXX"> > > > data="RECORD_SOFTWARE=FreeSwitch"/> > data="RECORD_ARTIST=FreeSwitch"/> > data="RECORD_COMMENT=FreeSwitch"/> > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/4f457b7a/attachment.html From msc at freeswitch.org Mon Sep 21 12:30:43 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Sep 2009 12:30:43 -0700 Subject: [Freeswitch-users] Displaying matched extension during a call In-Reply-To: <1b2118200909211127i19998a9dm8d92078e04d21ac1@mail.gmail.com> References: <1b2118200909211127i19998a9dm8d92078e04d21ac1@mail.gmail.com> Message-ID: <87f2f3b90909211230v4a9c6eefxb28ff5d8fe02a969@mail.gmail.com> On Mon, Sep 21, 2009 at 11:27 AM, Anil Kumar S. R. wrote: > Hi All, > > I am new to Freeswitch. So please bear with me if I ask any silly > questions. > > * Can anyone of you please tell me how to display the extension name which > has matched an incoming/outgoing call. > * And can you please elaborate what does '' > mean. > * Suppose we have set a variable in the extension of the dialplan XML. Is > there anyway we can display this variable on CLI for our debugging purposes. > > Anil, Here are few links to get started: Handy tutorial: http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT Chan vars: http://wiki.freeswitch.org/wiki/Channel_Variables Log app: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_log Note: the info app dumps all sorts of information to the console and is a great way to learn about many of the channel variables that FS has. The log app will make it easy for you to pinpoint just a single channel variable: Have fun! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/30ca89bc/attachment.html From msc at freeswitch.org Mon Sep 21 12:32:49 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Sep 2009 12:32:49 -0700 Subject: [Freeswitch-users] CudaTel Communications Server Version 1.0 Released Message-ID: <87f2f3b90909211232o14b1a8b5w7369b47c8ab48274@mail.gmail.com> At ClueCon 2009 we had an exciting announcement: Barracuda Networks and the FreeSWITCH team have been working together to create a new PBX appliance. Dubbed the CudaTel Communications Server, this new communications platform is both feature-rich and easy-to-use. We are pleased to announce that version 1.0 of the CudaTel Communcations Server has been released! The feature list for this affordable system is impressive: Automatic phone provisioning Multi-party conferencing Group calling SIP phone and provider support Automated attendant Voicemail TMD hardware option High definition codec support (G.722, G.722.1, G.722.1c) Call recording Active Directory and LDAP integration Encrypted VoIP support Many more features are included, all of which are controlled by an intuitive Web-based interface. We invite you to visit the CudaTel website or call 989-720-4000 for more information or to request evaluation units. -The FreeSWITCH Team -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/3d382f94/attachment.html From dmitry.bely at gmail.com Mon Sep 21 14:33:00 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Tue, 22 Sep 2009 01:33:00 +0400 Subject: [Freeswitch-users] BLF docs/howto? Message-ID: <90823c940909211433p1952d053ta98c295424ac459c@mail.gmail.com> Have not found anything usable in the wiki/mail list archives. I'm trying to setup BLF (busy lamp field) for Grandstream GXP-2000 phone. It offers BLF/eventlist BLF modes. Does Freeswitch supports both including the latter (RFC4662)? How to setup BLF on Freeswitch side? Are there any examples? - Dmitry Bely From tculjaga at gmail.com Mon Sep 21 15:46:49 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 22 Sep 2009 00:46:49 +0200 Subject: [Freeswitch-users] recompile with gdb Message-ID: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> Hi Guys, I have an issue running FS... it crashes apparently without leaving any log ... not even a core dump is left. The machine is dual AMD opteron quad core with 8 GB RAM and i'm running 75 simultaneous calls (with media) with a rate of 5 calls per second. As i was not able to reproduce the issue on a real traffic so i went back to sipp and started generating some... sipp & scenario files are ok. after a while (few minutes)... on sipp i start getting retransmissions and when i check FS i see two situations: 1. freeswitch has died 2. freeswitch process is running but it doesn't respond to any call... as nothing has been sent ... and after a while it dies too. I'm using sip profile external (moved to port 5060) with some semi-complex dialplan... attached. well .. the point is that i cannot even tell where it crashes as there is no log. I have: fs is dumping the log to the log directory ... but nothing special can't bee seen there... I tried to recompile with gdb export CFLAGS="-g -ggdb" export MOD_CFLAGS="-g -ggdb" ./configure but without luck... ode1:/opt/freeswitch-trunk# node1:/opt/freeswitch-trunk# sudo make make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' make all-am make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2 node1:/opt/freeswitch-trunk# node1:/opt/freeswitch-trunk# Of course I'm using the latest trunk... Can anyone help? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/5db2a99c/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: dp.xml Type: text/xml Size: 7451 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/5db2a99c/attachment.xml From gavin.henry at gmail.com Mon Sep 21 16:13:13 2009 From: gavin.henry at gmail.com (Gavin Henry) Date: Tue, 22 Sep 2009 00:13:13 +0100 Subject: [Freeswitch-users] CudaTel Communications Server Version 1.0 Released In-Reply-To: <87f2f3b90909211232o14b1a8b5w7369b47c8ab48274@mail.gmail.com> References: <87f2f3b90909211232o14b1a8b5w7369b47c8ab48274@mail.gmail.com> Message-ID: <13ca621c0909211613r6aa5c683g1a777b37b3d32f39@mail.gmail.com> URL??? On 21/09/2009, Michael Collins wrote: > At ClueCon 2009 we had an exciting announcement: Barracuda Networks and the > FreeSWITCH team have been working together to create a new PBX appliance. > Dubbed the CudaTel Communications Server, this new communications platform > is both feature-rich and easy-to-use. We are pleased to announce that > version 1.0 of the CudaTel Communcations Server has been released! > > The feature list for this affordable system is impressive: > Automatic phone provisioning > Multi-party conferencing > Group calling > SIP phone and provider support > Automated attendant > Voicemail > TMD hardware option > High definition codec support (G.722, G.722.1, G.722.1c) > Call recording > Active Directory and LDAP integration > Encrypted VoIP support > > Many more features are included, all of which are controlled by an intuitive > Web-based interface. > > We invite you to visit the CudaTel website or call > 989-720-4000 for more information or to request evaluation units. > > -The FreeSWITCH Team > -- Sent from my mobile device http://www.suretecsystems.com/services/openldap/ http://www.suretectelecom.com From william.suffill at gmail.com Mon Sep 21 16:19:38 2009 From: william.suffill at gmail.com (William Suffill) Date: Mon, 21 Sep 2009 19:19:38 -0400 Subject: [Freeswitch-users] CudaTel Communications Server Version 1.0 Released In-Reply-To: <13ca621c0909211613r6aa5c683g1a777b37b3d32f39@mail.gmail.com> References: <87f2f3b90909211232o14b1a8b5w7369b47c8ab48274@mail.gmail.com> <13ca621c0909211613r6aa5c683g1a777b37b3d32f39@mail.gmail.com> Message-ID: <6b65470d0909211619p79bfd1fep70b9e739d16efcc1@mail.gmail.com> "We invite you to visit the CudaTel / website or call 989-720-4000 for more information or to request evaluation units." From brian at freeswitch.org Mon Sep 21 16:22:12 2009 From: brian at freeswitch.org (Brian West) Date: Mon, 21 Sep 2009 18:22:12 -0500 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> Message-ID: This looks like gcc is segfaulting can you provide me a complete backtrace of the core file that dumps from FreeSWITCH? http://wiki.freeswitch.org/wiki/Reporting_Bugs It sounds like you might have bad ram or bad hardware... gcc crashing is usually a sign something is really wrong with your machine. /b On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: > but without luck... > > ode1:/opt/freeswitch-trunk# > node1:/opt/freeswitch-trunk# sudo make > make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' > make all-am > make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' > g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP - > MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o > pcrecpp_unittest.cc > g++: Internal error: Segmentation fault (program cc1plus) > Please submit a full bug report. > See for instructions. > make[2]: *** [pcrecpp_unittest.o] Error 1 > make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' > make: *** [libs/pcre/libpcre.la] Error 2 > node1:/opt/freeswitch-trunk# > node1:/opt/freeswitch-trunk# > > > Of course I'm using the latest trunk... > > Can anyone help? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/f809521a/attachment.html From hads at nice.net.nz Mon Sep 21 16:55:16 2009 From: hads at nice.net.nz (Hadley Rich) Date: Tue, 22 Sep 2009 11:55:16 +1200 Subject: [Freeswitch-users] CudaTel Communications Server Version 1.0 Released In-Reply-To: <13ca621c0909211613r6aa5c683g1a777b37b3d32f39@mail.gmail.com> References: <87f2f3b90909211232o14b1a8b5w7369b47c8ab48274@mail.gmail.com> <13ca621c0909211613r6aa5c683g1a777b37b3d32f39@mail.gmail.com> Message-ID: <200909221155.16510.hads@nice.net.nz> On Tue, 22 Sep 2009 11:13:13 Gavin Henry wrote: > URL??? > > On 21/09/2009, Michael Collins wrote: > > We invite you to visit the CudaTel website or > > call 989-720-4000 for more information or to request evaluation units. -- https://nicegear.co.nz VoIP and Open Source Hardware From msc at freeswitch.org Mon Sep 21 17:34:42 2009 From: msc at freeswitch.org (Michael Collins) Date: Mon, 21 Sep 2009 17:34:42 -0700 Subject: [Freeswitch-users] CudaTel Communications Server Version 1.0 Released In-Reply-To: <6b65470d0909211619p79bfd1fep70b9e739d16efcc1@mail.gmail.com> References: <87f2f3b90909211232o14b1a8b5w7369b47c8ab48274@mail.gmail.com> <13ca621c0909211613r6aa5c683g1a777b37b3d32f39@mail.gmail.com> <6b65470d0909211619p79bfd1fep70b9e739d16efcc1@mail.gmail.com> Message-ID: <87f2f3b90909211734k2faa40a9i19328b415eca8908@mail.gmail.com> On Mon, Sep 21, 2009 at 4:19 PM, William Suffill wrote: > "We invite you to visit the CudaTel / website or > call 989-720-4000 for more information or to request evaluation > units." > > Hehe, thanks for pointing that out. Also, I said "TMD hardware option" when I really meant "TDM hardware option" :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/8cf5bb5b/attachment-0001.html From nicolas at medularis.com Mon Sep 21 17:41:19 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Mon, 21 Sep 2009 20:41:19 -0400 Subject: [Freeswitch-users] ALLOTTED_TIMEOUT hangup cause? Message-ID: <1b46b4e80909211741n2de3191fvb64a61eab3a3259e@mail.gmail.com> Hi, Today, while trying to bridge some calls I started to get a ALLOTTED_TIMEOUT hangup cause on the second leg. I looked for info on the Wiki and Google, but I couldn't find a detailed explanation. Does anybody know what does it mean exactly? Thanks! Nicolas -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090921/98afd586/attachment.html From codecomplete at free.fr Tue Sep 22 00:07:51 2009 From: codecomplete at free.fr (Fred-145) Date: Tue, 22 Sep 2009 00:07:51 -0700 (PDT) Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <25530400.post@talk.nabble.com> References: <25520404.post@talk.nabble.com> <25530400.post@talk.nabble.com> Message-ID: <25530601.post@talk.nabble.com> Thanks for the link to PORTech's MV-370. For those interested, it can be had for a retail price of ?150/?165 (before VAT). www.discountphonesystems.co.uk/acatalog/MV-370.html -- View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530601.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From dmitry.bely at gmail.com Tue Sep 22 00:31:28 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Tue, 22 Sep 2009 11:31:28 +0400 Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <25530601.post@talk.nabble.com> References: <25520404.post@talk.nabble.com> <25530400.post@talk.nabble.com> <25530601.post@talk.nabble.com> Message-ID: <90823c940909220031q33a463b6h5cc4935179d47774@mail.gmail.com> What about this one? http://www.gempro.com.tw/gp-710.htm On Tue, Sep 22, 2009 at 11:07 AM, Fred-145 wrote: > > Thanks for the link to PORTech's MV-370. For those interested, it can be had > for a retail price of ?150/?165 (before VAT). > > www.discountphonesystems.co.uk/acatalog/MV-370.html > -- > View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530601.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org - Dmitry Bely From edpimentl at gmail.com Tue Sep 22 01:02:56 2009 From: edpimentl at gmail.com (EdPimentl) Date: Tue, 22 Sep 2009 04:02:56 -0400 Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <25530601.post@talk.nabble.com> References: <25520404.post@talk.nabble.com> <25530400.post@talk.nabble.com> <25530601.post@talk.nabble.com> Message-ID: <9dc4a1670909220102i1376001aj17e2e0d3c73cfd14@mail.gmail.com> I used Portech now for 3 years and have been very happy with their product. -E -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/5c4edc49/attachment.html From sranil at gmail.com Tue Sep 22 02:19:37 2009 From: sranil at gmail.com (Anil Kumar S. R.) Date: Tue, 22 Sep 2009 14:49:37 +0530 Subject: [Freeswitch-users] Displaying matched extension during a call In-Reply-To: <87f2f3b90909211230v4a9c6eefxb28ff5d8fe02a969@mail.gmail.com> References: <1b2118200909211127i19998a9dm8d92078e04d21ac1@mail.gmail.com> <87f2f3b90909211230v4a9c6eefxb28ff5d8fe02a969@mail.gmail.com> Message-ID: <1b2118200909220219h4983ca5fx7504ea052ce9dca5@mail.gmail.com> Thanks Michael for your links. Till now I was working on the command line which we get on executing the 'freeswitch' command. I didn't know abt the 'fs_cli' command. The fs_cli gives lot more information that will be helpful for novice user like me. Thanks, Anil 2009/9/22 Michael Collins > > > On Mon, Sep 21, 2009 at 11:27 AM, Anil Kumar S. R. wrote: > >> Hi All, >> >> I am new to Freeswitch. So please bear with me if I ask any silly >> questions. >> >> * Can anyone of you please tell me how to display the extension name which >> has matched an incoming/outgoing call. >> * And can you please elaborate what does '' >> mean. >> * Suppose we have set a variable in the extension of the dialplan XML. Is >> there anyway we can display this variable on CLI for our debugging purposes. >> >> > Anil, > > Here are few links to get started: > Handy tutorial: http://www.linuxpromagazine.com/Issues/2009/106/TALK-SOFT > Chan vars: http://wiki.freeswitch.org/wiki/Channel_Variables > Log app: http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_log > > Note: the info app dumps all sorts of information to the console and is a > great way to learn about many of the channel variables that FS has. The log > app will make it easy for you to pinpoint just a single channel variable: > > > Have fun! > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anil Kumar S. R. http://sranil.googlepages.com/ "The best way to succeed in this world is to act on the advice you give to others." -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/bdeb7bd9/attachment.html From lakindia89 at gmail.com Tue Sep 22 02:30:36 2009 From: lakindia89 at gmail.com (lakshmanan) Date: Tue, 22 Sep 2009 02:30:36 -0700 (PDT) Subject: [Freeswitch-users] Mod_perl $session in not hangup Message-ID: <25530646.post@talk.nabble.com> Hi all, I've the following mod_perl program to execute when I call to an extension (say 777). I use twinkle as a soft phone, to make calls. #!/usr/bin/perl use strict; use freeswitch; our $session; $session->answer(); if($session->ready()) { my $uuid=$session->getVariable("uuid"); freeswitch::consoleLog("INFO","UUID is $uuid\n"); freeswitch::consoleLog("INFO","Session is answered\n"); $session->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav"); my $dtmf = $session->getDigits(4,"", 5000); freeswitch::consoleLog("INFO","I received $dtmf\n"); $session->hangup("NORMAL_CLEARING"); sleep(5); # Some other statements. } return 1; Everything is fine. After executing $session->hangup, I got NORMAL_CLEARING in my freeswitch console. But in my soft phone, still the channel is active for 5 seconds. The call got ended only after the 5 seconds sleep. But if I create my own session like my $session=new freeswitch::Session("user/1000"); and I say $session->hangup(), it got terminated. I wanted to know why there is such difference?? or am I wrong?? Please clarify me. -- View this message in context: http://www.nabble.com/Mod_perl-%24session-in-not-hangup-tp25530646p25530646.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From lyncker at lyth.de Tue Sep 22 03:56:39 2009 From: lyncker at lyth.de (Filip Lyncker) Date: Tue, 22 Sep 2009 12:56:39 +0200 Subject: [Freeswitch-users] Some Newbie questions about dialplan and local Sip registration In-Reply-To: <65d96fc80909181104v2dc2a9ag268a51cd45c28e6a@mail.gmail.com> References: <4AB39788.3000907@lyth.de> <65d96fc80909181104v2dc2a9ag268a51cd45c28e6a@mail.gmail.com> Message-ID: <4AB8AD67.7040201@lyth.de> Hi Tihomir, Thanks for your help , I added the Asteriskparameters as you described below, but I still get the same timeout error: 2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry to 270 seconds. 2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk Registration Failed with status Request Timeout [408]. failure #9 Now, my gateway entry looks like the following : What can be still wrong here? Regards, Filip Tihomir Culjaga schrieb: > hi Filip, > > > for calling a user... please read this first: > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User > for making a GW register into e.g. asterisk please use this: > > > > > > > > > > > > > this should be enough to register the GW... after that please read > this: > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways > > > in your case it will be something like this: > > > expression="^(NUMBER_TO_SEND_TO_ASTERISK)$"> > > > > > > > > > > > > > On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker > wrote: > > Hi List, > > for the first experiments with freeswitch I downloaded the Windows > installation. > Now Im trying to get my 2 Sipphones get connected to. Later I want > connect the freeswitch to my asterisk gateway. > > I find the examples pretty complex therfore Im trying to build up a > simple solution to understand the functions from the scratch .. > > my current problem is , that I cant route my local sips to each > other ( > registration seems to work now). > the next is , that freeshwitch is not able to connect to asterisk. > but I > will describe this later. > > I installed in the Directory a xml file ( called 22.xml) with the > following content : > > > > > > > > > > > > > > > value="22"> > > > > > > > > > > > > > > value="24"> > > > > > > This seems to be ok now. Now I want to dial from 22 to 24 , > wherefore I > configured this dialplan : > > > > > > > > > > > wich doesnt work , mybe b/c the user/${dialed_extension} I dont > know... > Freeswitch says: > [INFO] switch_core_state_machine.c:136 No Route, Aborting > [NOTICE] switch_core_state_machine.c:137 Hangup > sofia/internal/24 at 192.168.1.34 > [CS_ROUTING] [NO_ROUTE_DESTINATION] > [NOTICE] switch_core_session.c:1086 Session 17 > (sofia/internal/24 at 192.168.1.34 ) Ended > [NOTICE] switch_core_session.c:1088 Close Channel > sofia/internal/24 at 192.168.1.34 [CS_DESTROY] > > Im sure , for you guys this cant be a big deal;) > > > Next Point is my Asterisk registration , mybe you can help me out here > to .. : > > In the sip-profiles/external I installed the my_asterisk.xml with that > content : > > > > > > > > > > > > Freeswitch allways complains a timeout error : > [ERR] sofia_reg.c:1460 asterisk Registration Failed with status > Request > Timeout [408]. failure #17 > [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting retry > to 540 seconds. > > it seems that It cant connect ( I also tried out to explicit set the > port to 5060 b/c I read something about 5080 .. : name="sip-port" > value="5060"> but this didnt help) > In my Asterisk I set in the sip.conf the entry 28 with the pw test > .... > > > If someone could help me with my first steps I would be verrry > thankful ;)) > > cheers > > > Filip > > -- > _________________________________ > Filip Lyncker, Dipl.-Inform. (FH) > > > Lyncker & Theis GmbH > Wilhelmstr. 16 > 65185 Wiesbaden > Germany > > Fon +49 611/9006951 > Fax +49 611/9406125 > > > Handelsregister: HRB 23156 Amtsgericht Wiesbaden > Steuernummer: 4023897051 > USt-IdNr.: DE255806399 > > Gesch?ftsf?hrer: > Filip Lyncker, > Armin Theis > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- _________________________________ Filip Lyncker, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Gesch?ftsf?hrer: Filip Lyncker, Armin Theis From nagalenoj at gmail.com Tue Sep 22 04:11:08 2009 From: nagalenoj at gmail.com (Nagalenoj) Date: Tue, 22 Sep 2009 04:11:08 -0700 (PDT) Subject: [Freeswitch-users] Mod_perl or ESL In-Reply-To: <25520023.post@talk.nabble.com> References: <25520023.post@talk.nabble.com> Message-ID: <25530677.post@talk.nabble.com> I need to handle some hundreds of call. So, which one can I opt? Nagalenoj wrote: > > Dear friends, > I want to know which is the better way to do route calls and control > calls. I've did a experiment which can be done in both ways, Mod_perl and > ESL. I don't know which one is better to take. > When I see some earlier posts, It is given like Mod_perl has some > limitations and I don't know what kind of limitations they are., > Can someone say which is better to use and how it is better? > -- View this message in context: http://www.nabble.com/Mod_perl-or-ESL-tp25520023p25530677.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From janvb at live.com Tue Sep 22 04:35:34 2009 From: janvb at live.com (Jan Berger) Date: Tue, 22 Sep 2009 13:35:34 +0200 Subject: [Freeswitch-users] FreeSWITCH - UTRAN/UTRA In-Reply-To: References: <94790b850909211213j52aae598v861dd8ab2e9e7983@mail.gmail.com> Message-ID: hi all, Has anyone connected FreeSWITCH to UTRAN? If so, how was that done? Jan _________________________________________________________________ With Windows Live, you can organize, edit, and share your photos. http://www.microsoft.com/middleeast/windows/windowslive/products/photo-gallery-edit.aspx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/2df3e04a/attachment.html From lyncker at lyth.de Tue Sep 22 04:52:20 2009 From: lyncker at lyth.de (Filip Lyncker) Date: Tue, 22 Sep 2009 13:52:20 +0200 Subject: [Freeswitch-users] Unable to set internal call to registered sip user Message-ID: <4AB8BA74.9040008@lyth.de> Dear List, I read the documentation, but Im still confused about how to dial a internal registered sip user. I configured the both sip phones in the directory in my local.xml file : It seems, that they can connect to the freeswitch. I configured the dialplan like following : ... If I call from the sip user 24 to 22 , freeswitch logs the following and gives an busy tone immediately: freeswitch at Bigfish> 2009-09-22 13:50:29.367114 [NOTICE] switch_channel.c:602 New Channel sofia/internal/24 at 192.168.1.34 [decc119c-a973-6b4c-bf11-ec251c653cda] 2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing 24->22 in context default 2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user [@192.168.1.34] 2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] 2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed. Cause: SUBSCRIBER_ABSENT 2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup sofia/internal/24 at 192.168.1.34 [CS_EXECUTE] [SUBSCRIBER_ABSENT] 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session 13 (sofia/internal/24 at 192.168.1.34) Ended 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/24 at 192.168.1.34 [CS_DESTROY] thanks again for your help ... regards, Filip -- _________________________________ Filip Lyncker, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Gesch?ftsf?hrer: Filip Lyncker, Armin Theis From tculjaga at gmail.com Tue Sep 22 05:30:29 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 22 Sep 2009 14:30:29 +0200 Subject: [Freeswitch-users] Unable to set internal call to registered sip user In-Reply-To: <4AB8BA74.9040008@lyth.de> References: <4AB8BA74.9040008@lyth.de> Message-ID: <65d96fc80909220530v66e569a5n622c8163ba98d3ae@mail.gmail.com> and this is not enough for you? T. On Tue, Sep 22, 2009 at 1:52 PM, Filip Lyncker wrote: > Dear List, > > I read the documentation, but Im still confused about how to dial a > internal registered sip user. > > I configured the both sip phones in the directory in my local.xml file : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > It seems, that they can connect to the freeswitch. > > I configured the dialplan like following : > > > > > > > data="user/${dialed_extension}@${domain_name}"> > > > > > ... > > > If I call from the sip user 24 to 22 , freeswitch logs the following and > gives an busy tone immediately: > > freeswitch at Bigfish> 2009-09-22 13:50:29.367114 [NOTICE] > switch_channel.c:602 New Channel sofia/internal/24 at 192.168.1.34 > [decc119c-a973-6b4c-bf11-ec251c653cda] > 2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing > 24->22 in context default > 2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user > [@192.168.1.34] > 2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot > create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] > 2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed. > Cause: SUBSCRIBER_ABSENT > 2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup > sofia/internal/24 at 192.168.1.34 [CS_EXECUTE] [SUBSCRIBER_ABSENT] > 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session > 13 (sofia/internal/24 at 192.168.1.34) Ended > 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/24 at 192.168.1.34 [CS_DESTROY] > > thanks again for your help ... > > > regards, > > Filip > > > -- > _________________________________ > Filip Lyncker, Dipl.-Inform. (FH) > > > Lyncker & Theis GmbH > Wilhelmstr. 16 > 65185 Wiesbaden > Germany > > Fon +49 611/9006951 > Fax +49 611/9406125 > > > Handelsregister: HRB 23156 Amtsgericht Wiesbaden > Steuernummer: 4023897051 > USt-IdNr.: DE255806399 > > Gesch?ftsf?hrer: > Filip Lyncker, > Armin Theis > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/10de486d/attachment.html From tculjaga at gmail.com Tue Sep 22 05:34:27 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 22 Sep 2009 14:34:27 +0200 Subject: [Freeswitch-users] Some Newbie questions about dialplan and local Sip registration In-Reply-To: <4AB8AD67.7040201@lyth.de> References: <4AB39788.3000907@lyth.de> <65d96fc80909181104v2dc2a9ag268a51cd45c28e6a@mail.gmail.com> <4AB8AD67.7040201@lyth.de> Message-ID: <65d96fc80909220534x2482b1bcvbc1c17ff45e0fad8@mail.gmail.com> hmmm .. can you register using x-lite or some other softphone with the same credentials? can you paste a siptrace of the failed registration? BTW: Make sure nothing is already registered with this credentials when you try with FS T. On Tue, Sep 22, 2009 at 12:56 PM, Filip Lyncker wrote: > Hi Tihomir, > > Thanks for your help , I added the Asteriskparameters as you described > below, but I still get the same timeout error: > 2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed > Registration, setting retry to 270 seconds. > 2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk Registration > Failed with status Request Timeout [408]. failure #9 > > Now, my gateway entry looks like the following : > > > > > > > > > > > > > > > What can be still wrong here? > > Regards, > > Filip > > > > Tihomir Culjaga schrieb: > > hi Filip, > > > > > > for calling a user... please read this first: > > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User > > for making a GW register into e.g. asterisk please use this: > > > > > > > > > > > > > > > > > > > > > > > > > > this should be enough to register the GW... after that please read > > this: > > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways > > > > > > in your case it will be something like this: > > > > > > > expression="^(NUMBER_TO_SEND_TO_ASTERISK)$"> > > > > > > > > > > > > > > > > > > > > > > > > > > On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker > > wrote: > > > > Hi List, > > > > for the first experiments with freeswitch I downloaded the Windows > > installation. > > Now Im trying to get my 2 Sipphones get connected to. Later I want > > connect the freeswitch to my asterisk gateway. > > > > I find the examples pretty complex therfore Im trying to build up a > > simple solution to understand the functions from the scratch .. > > > > my current problem is , that I cant route my local sips to each > > other ( > > registration seems to work now). > > the next is , that freeshwitch is not able to connect to asterisk. > > but I > > will describe this later. > > > > I installed in the Directory a xml file ( called 22.xml) with the > > following content : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="22"> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="24"> > > > > > > > > > > > > This seems to be ok now. Now I want to dial from 22 to 24 , > > wherefore I > > configured this dialplan : > > > > > > > > > > > > > > > > > > > > > > wich doesnt work , mybe b/c the user/${dialed_extension} I dont > > know... > > Freeswitch says: > > [INFO] switch_core_state_machine.c:136 No Route, Aborting > > [NOTICE] switch_core_state_machine.c:137 Hangup > > sofia/internal/24 at 192.168.1.34 > > [CS_ROUTING] [NO_ROUTE_DESTINATION] > > [NOTICE] switch_core_session.c:1086 Session 17 > > (sofia/internal/24 at 192.168.1.34 ) Ended > > [NOTICE] switch_core_session.c:1088 Close Channel > > sofia/internal/24 at 192.168.1.34 [CS_DESTROY] > > > > Im sure , for you guys this cant be a big deal;) > > > > > > Next Point is my Asterisk registration , mybe you can help me out > here > > to .. : > > > > In the sip-profiles/external I installed the my_asterisk.xml with > that > > content : > > > > > > > > > > > > > > > > > > > > > > > > Freeswitch allways complains a timeout error : > > [ERR] sofia_reg.c:1460 asterisk Registration Failed with status > > Request > > Timeout [408]. failure #17 > > [WARNING] sofia_reg.c:364 asterisk Failed Registration, setting > retry > > to 540 seconds. > > > > it seems that It cant connect ( I also tried out to explicit set the > > port to 5060 b/c I read something about 5080 .. : > name="sip-port" > > value="5060"> but this didnt help) > > In my Asterisk I set in the sip.conf the entry 28 with the pw test > > .... > > > > > > If someone could help me with my first steps I would be verrry > > thankful ;)) > > > > cheers > > > > > > Filip > > > > -- > > _________________________________ > > Filip Lyncker, Dipl.-Inform. (FH) > > > > > > Lyncker & Theis GmbH > > Wilhelmstr. 16 > > 65185 Wiesbaden > > Germany > > > > Fon +49 611/9006951 > > Fax +49 611/9406125 > > > > > > Handelsregister: HRB 23156 Amtsgericht Wiesbaden > > Steuernummer: 4023897051 > > USt-IdNr.: DE255806399 > > > > Gesch?ftsf?hrer: > > Filip Lyncker, > > Armin Theis > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > _________________________________ > Filip Lyncker, Dipl.-Inform. (FH) > > > Lyncker & Theis GmbH > Wilhelmstr. 16 > 65185 Wiesbaden > Germany > > Fon +49 611/9006951 > Fax +49 611/9406125 > > > Handelsregister: HRB 23156 Amtsgericht Wiesbaden > Steuernummer: 4023897051 > USt-IdNr.: DE255806399 > > Gesch?ftsf?hrer: > Filip Lyncker, > Armin Theis > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/aaf06a8f/attachment-0001.html From tculjaga at gmail.com Tue Sep 22 05:36:42 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 22 Sep 2009 14:36:42 +0200 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> Message-ID: <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> hi Brian, well, there is no coredump at all... and when i start FS with gdb it doesn't crash :P I need to do some more testing and will come back to you. T. On Tue, Sep 22, 2009 at 1:22 AM, Brian West wrote: > This looks like gcc is segfaulting can you provide me a complete backtrace > of the core file that dumps from FreeSWITCH? > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > It sounds like you might have bad ram or bad hardware... gcc crashing is > usually a sign something is really wrong with your machine. > > /b > > On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: > > but without luck... > > ode1:/opt/freeswitch-trunk# > node1:/opt/freeswitch-trunk# sudo make > make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' > make all-am > make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' > g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF > .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc > g++: Internal error: Segmentation fault (program cc1plus) > Please submit a full bug report. > See for instructions. > make[2]: *** [pcrecpp_unittest.o] Error 1 > make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' > make: *** [libs/pcre/libpcre.la] Error 2 > node1:/opt/freeswitch-trunk# > node1:/opt/freeswitch-trunk# > > > Of course I'm using the latest trunk... > > Can anyone help? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/01baf75a/attachment.html From lyncker at lyth.de Tue Sep 22 06:55:09 2009 From: lyncker at lyth.de (Filip Lyncker) Date: Tue, 22 Sep 2009 15:55:09 +0200 Subject: [Freeswitch-users] Unable to set internal call to registered sip user In-Reply-To: <65d96fc80909220530v66e569a5n622c8163ba98d3ae@mail.gmail.com> References: <4AB8BA74.9040008@lyth.de> <65d96fc80909220530v66e569a5n622c8163ba98d3ae@mail.gmail.com> Message-ID: <4AB8D73D.1030801@lyth.de> ok , i tried several things : but all this doesnt work sorry mybe I dont see something apparent , but I dont have a clue... Tihomir Culjaga schrieb: > and this is not enough for you? > > > data="user/${dialed_extension}@${domain_name}"/> > > > > T. > > > On Tue, Sep 22, 2009 at 1:52 PM, Filip Lyncker > wrote: > > Dear List, > > I read the documentation, but Im still confused about how to dial a > internal registered sip user. > > I configured the both sip phones in the directory in my local.xml > file : > > > > > > > > > > > > > > > value="22"> > > > > > > > > > > > > > > value="24"> > > > > > > It seems, that they can connect to the freeswitch. > > I configured the dialplan like following : > > > > > > > data="user/${dialed_extension}@${domain_name}"> > > > > > ... > > > If I call from the sip user 24 to 22 , freeswitch logs the > following and > gives an busy tone immediately: > > freeswitch at Bigfish> 2009-09-22 13:50:29.367114 [NOTICE] > switch_channel.c:602 New Channel sofia/internal/24 at 192.168.1.34 > > [decc119c-a973-6b4c-bf11-ec251c653cda] > 2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing > 24->22 in context default > 2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find > user > [@192.168.1.34 ] > 2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot > create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] > 2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed. > Cause: SUBSCRIBER_ABSENT > 2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup > sofia/internal/24 at 192.168.1.34 > [CS_EXECUTE] [SUBSCRIBER_ABSENT] > 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session > 13 (sofia/internal/24 at 192.168.1.34 ) Ended > 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/24 at 192.168.1.34 > [CS_DESTROY] > > thanks again for your help ... > > > regards, > > Filip > > > -- > _________________________________ > Filip Lyncker, Dipl.-Inform. (FH) > > > Lyncker & Theis GmbH > Wilhelmstr. 16 > 65185 Wiesbaden > Germany > > Fon +49 611/9006951 > Fax +49 611/9406125 > > > Handelsregister: HRB 23156 Amtsgericht Wiesbaden > Steuernummer: 4023897051 > USt-IdNr.: DE255806399 > > Gesch?ftsf?hrer: > Filip Lyncker, > Armin Theis > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- _________________________________ Filip Lyncker, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Gesch?ftsf?hrer: Filip Lyncker, Armin Theis From brian at freeswitch.org Tue Sep 22 06:20:57 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Sep 2009 08:20:57 -0500 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> Message-ID: FreeSWITCH compiles with debug symbols by default but you showed an output where GCC was segfaulting so you have bad ram or bad hardware and I suspect that is your problem. /b On Sep 22, 2009, at 7:36 AM, Tihomir Culjaga wrote: > hi Brian, > > well, there is no coredump at all... and when i start FS with gdb it > doesn't crash :P > I need to do some more testing and will come back to you. > > T. From klejch+freeswitch at netbox.cz Tue Sep 22 06:06:19 2009 From: klejch+freeswitch at netbox.cz (Vladimir Klejch) Date: Tue, 22 Sep 2009 15:06:19 +0200 (CEST) Subject: [Freeswitch-users] Unable to set internal call to registered sip user In-Reply-To: <4AB8BA74.9040008@lyth.de> References: <4AB8BA74.9040008@lyth.de> Message-ID: Hi in dialplan i see: -> check on variable destination_number and later -> bridge to variable dialed_extension , other then checked destination_number or $1 from regexp try: or By Kleo On Tue, 22 Sep 2009, Filip Lyncker wrote: > Dear List, > > I read the documentation, but Im still confused about how to dial a > internal registered sip user. > > I configured the both sip phones in the directory in my local.xml file : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > It seems, that they can connect to the freeswitch. > > I configured the dialplan like following : > > > > > > > data="user/${dialed_extension}@${domain_name}"> > > > > > ... > > > If I call from the sip user 24 to 22 , freeswitch logs the following and > gives an busy tone immediately: > > freeswitch at Bigfish> 2009-09-22 13:50:29.367114 [NOTICE] > switch_channel.c:602 New Channel sofia/internal/24 at 192.168.1.34 > [decc119c-a973-6b4c-bf11-ec251c653cda] > 2009-09-22 13:50:29.372973 [INFO] mod_dialplan_xml.c:315 Processing > 24->22 in context default > 2009-09-22 13:50:29.372973 [WARNING] mod_dptools.c:2365 Can't find user > [@192.168.1.34] > 2009-09-22 13:50:29.372973 [ERR] switch_ivr_originate.c:1510 Cannot > create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT] > 2009-09-22 13:50:29.372973 [INFO] mod_dptools.c:2093 Originate Failed. > Cause: SUBSCRIBER_ABSENT > 2009-09-22 13:50:29.372973 [NOTICE] mod_dptools.c:2125 Hangup > sofia/internal/24 at 192.168.1.34 [CS_EXECUTE] [SUBSCRIBER_ABSENT] > 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1086 Session > 13 (sofia/internal/24 at 192.168.1.34) Ended > 2009-09-22 13:50:29.390550 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/24 at 192.168.1.34 [CS_DESTROY] > > thanks again for your help ... > > > regards, > > Filip > > > -- _____________________________________________________________ | You have moved the mouse. # | Windows must be restarted for the changes to take effect. # | # ##############################################################/ ~~ ~~ ~~ ~~ ~~ ~~ ~~ Vladimir `KLEO' Klejch Kleo'at'netbox.cz ... ... ... ... From lyncker at lyth.de Tue Sep 22 07:29:29 2009 From: lyncker at lyth.de (Filip Lyncker) Date: Tue, 22 Sep 2009 16:29:29 +0200 Subject: [Freeswitch-users] Some Newbie questions about dialplan and local Sip registration In-Reply-To: <65d96fc80909220534x2482b1bcvbc1c17ff45e0fad8@mail.gmail.com> References: <4AB39788.3000907@lyth.de> <65d96fc80909181104v2dc2a9ag268a51cd45c28e6a@mail.gmail.com> <4AB8AD67.7040201@lyth.de> <65d96fc80909220534x2482b1bcvbc1c17ff45e0fad8@mail.gmail.com> Message-ID: <4AB8DF49.8080902@lyth.de> now i registered from my x-lite client without anyproblems. but I think i got it now, my tcpdump says the following : IP 192.168.1.119.5060 > 93.210.212.xxx.5080: SIP, length: 465 wich is the external IP of my network ! must have somthing todo with NAT / Masquerade options... how can I avoid this ? thanks for your help ... regards, filip Tihomir Culjaga schrieb: > hmmm .. can you register using x-lite or some other softphone with the > same credentials? > > can you paste a siptrace of the failed registration? > > > BTW: Make sure nothing is already registered with this credentials > when you try with FS > > T. > > On Tue, Sep 22, 2009 at 12:56 PM, Filip Lyncker > wrote: > > Hi Tihomir, > > Thanks for your help , I added the Asteriskparameters as you described > below, but I still get the same timeout error: > 2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed > Registration, setting retry to 270 seconds. > 2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk > Registration > Failed with status Request Timeout [408]. failure #9 > > Now, my gateway entry looks like the following : > > > > > > > > > > > > > > > What can be still wrong here? > > Regards, > > Filip > > > > Tihomir Culjaga schrieb: > > hi Filip, > > > > > > for calling a user... please read this first: > > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User > > for making a GW register into e.g. asterisk please use this: > > > > > > > > > > > > > > > > > > > > > > > > > > this should be enough to register the GW... after that please read > > this: > > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways > > > > > > in your case it will be something like this: > > > > > > > expression="^(NUMBER_TO_SEND_TO_ASTERISK)$"> > > > > > > > > > > > > > > > > > > > > > > > > > > On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker > > >> wrote: > > > > Hi List, > > > > for the first experiments with freeswitch I downloaded the > Windows > > installation. > > Now Im trying to get my 2 Sipphones get connected to. Later > I want > > connect the freeswitch to my asterisk gateway. > > > > I find the examples pretty complex therfore Im trying to > build up a > > simple solution to understand the functions from the scratch .. > > > > my current problem is , that I cant route my local sips to each > > other ( > > registration seems to work now). > > the next is , that freeshwitch is not able to connect to > asterisk. > > but I > > will describe this later. > > > > I installed in the Directory a xml file ( called 22.xml) > with the > > following content : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="22"> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="24"> > > > > > > > > > > > > This seems to be ok now. Now I want to dial from 22 to 24 , > > wherefore I > > configured this dialplan : > > > > > > > > > > > > data="user/${dialed_extension}"/> > > > > > > > > > > wich doesnt work , mybe b/c the user/${dialed_extension} I dont > > know... > > Freeswitch says: > > [INFO] switch_core_state_machine.c:136 No Route, Aborting > > [NOTICE] switch_core_state_machine.c:137 Hangup > > sofia/internal/24 at 192.168.1.34 > > > > [CS_ROUTING] [NO_ROUTE_DESTINATION] > > [NOTICE] switch_core_session.c:1086 Session 17 > > (sofia/internal/24 at 192.168.1.34 > >) Ended > > [NOTICE] switch_core_session.c:1088 Close Channel > > sofia/internal/24 at 192.168.1.34 > > [CS_DESTROY] > > > > Im sure , for you guys this cant be a big deal;) > > > > > > Next Point is my Asterisk registration , mybe you can help > me out here > > to .. : > > > > In the sip-profiles/external I installed the my_asterisk.xml > with that > > content : > > > > > > > > > > > > > > > > > > > > > > > > Freeswitch allways complains a timeout error : > > [ERR] sofia_reg.c:1460 asterisk Registration Failed with status > > Request > > Timeout [408]. failure #17 > > [WARNING] sofia_reg.c:364 asterisk Failed Registration, > setting retry > > to 540 seconds. > > > > it seems that It cant connect ( I also tried out to explicit > set the > > port to 5060 b/c I read something about 5080 .. : > name="sip-port" > > value="5060"> but this didnt help) > > In my Asterisk I set in the sip.conf the entry 28 with the > pw test > > .... > > > > > > If someone could help me with my first steps I would be verrry > > thankful ;)) > > > > cheers > > > > > > Filip > > > > -- > > _________________________________ > > Filip Lyncker, Dipl.-Inform. (FH) > > > > > > Lyncker & Theis GmbH > > Wilhelmstr. 16 > > 65185 Wiesbaden > > Germany > > > > Fon +49 611/9006951 > > Fax +49 611/9406125 > > > > > > Handelsregister: HRB 23156 Amtsgericht Wiesbaden > > Steuernummer: 4023897051 > > USt-IdNr.: DE255806399 > > > > Gesch?ftsf?hrer: > > Filip Lyncker, > > Armin Theis > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > _________________________________ > Filip Lyncker, Dipl.-Inform. (FH) > > > Lyncker & Theis GmbH > Wilhelmstr. 16 > 65185 Wiesbaden > Germany > > Fon +49 611/9006951 > Fax +49 611/9406125 > > > Handelsregister: HRB 23156 Amtsgericht Wiesbaden > Steuernummer: 4023897051 > USt-IdNr.: DE255806399 > > Gesch?ftsf?hrer: > Filip Lyncker, > Armin Theis > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- _________________________________ Filip Lyncker, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Gesch?ftsf?hrer: Filip Lyncker, Armin Theis From itamar at ispbrasil.com.br Sun Sep 20 14:18:13 2009 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Sun, 20 Sep 2009 18:18:13 -0300 Subject: [Freeswitch-users] Affordable GSM gateway for one cellphone? In-Reply-To: <25530130.post@talk.nabble.com> References: <25520404.post@talk.nabble.com> <883102f54ceed3b94dd3c6362573e280.squirrel@correo.nodo50.org> <65d96fc80909190824k50c03defwf4d4cf2af931a1dd@mail.gmail.com> <25530130.post@talk.nabble.com> Message-ID: portech also seems to be good. On Sun, Sep 20, 2009 at 5:57 PM, Fred-145 wrote: > > Thanks Tihomir for the link. > > >From what I read, it appears that EdgePBX's FX02G is a full-fledged Asterisk > server with a GSM module and an FXS module. Did you reflash its NAND to run > Freeswitch? > > At $300, I guess customers will rather take a subscription with a VoIP > provided and use their GSM gateway, but I'm interested in knowing whether > the FX02G can be used as a PSTN/GSM gateway, possibly with FreeSwitch > running on that unit as well. > > Thank you. > -- > View this message in context: http://www.nabble.com/Affordable-GSM-gateway-for-one-cellphone--tp25520404p25530130.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ------------ Itamar Reis Peixoto e-mail/msn: itamar at ispbrasil.com.br sip: itamar at ispbrasil.com.br skype: itamarjp icq: 81053601 +55 11 4063 5033 +55 34 3221 8599 From francisv.list at gmail.com Mon Sep 21 21:17:15 2009 From: francisv.list at gmail.com (Francis Vidal) Date: Tue, 22 Sep 2009 12:17:15 +0800 Subject: [Freeswitch-users] Can this be done in FreeSWITCH? Message-ID: Hi all, Consider the following scenario: Calling party --> DID provider --> Cisco AS5300 --> POTS provider --> Called party The Calling party calls a number provided by the DID provider. This is then processed by the AS5300 facing the POTS provider to do the following number translation: ANI = 1 + DNIS (the ANI assumes the identify of the DNIS prefixed with "1"). The Cisco AS5300 then sends a "prefix" which is actually the number of the Called party in their system (of the POTS provider). However, the Cisco AS5300 has a finite limit on the number of translations (approx. 128-300 translations). Can the number translation be done on FreeSWITCH instead? Calling party --> DID provider --> FreeSWITCH --> Cisco AS5300 --> POTS provider --> Called party This can also evolve into: Calling party --> DID provider --> FreeSWITCH --> Cisco AS5300[1] --> POTS provider --> Called party \ / +-----> Cisco AS5300[2] --->+ If we wanted to increase the number of ports the POTS provider. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/91aa4272/attachment.html From anthony.minessale at gmail.com Tue Sep 22 07:35:21 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Sep 2009 09:35:21 -0500 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> Message-ID: <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> see this from your own log? make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc g++: Internal error: Segmentation fault (program cc1plus) Please submit a full bug report. See for instructions. make[2]: *** [pcrecpp_unittest.o] Error 1 make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make[1]: *** [all] Error 2 make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' make: *** [libs/pcre/libpcre.la] Error 2 This is a FATAL error to have on your machine. It's failing during the build. This is your compiler crashing while trying to build the software. This is very bad. You most likely have a hardware failure and need to replace the machine or at the very least all of the memory chips. On Tue, Sep 22, 2009 at 7:36 AM, Tihomir Culjaga wrote: > hi Brian, > > well, there is no coredump at all... and when i start FS with gdb it > doesn't crash :P > I need to do some more testing and will come back to you. > > T. > > On Tue, Sep 22, 2009 at 1:22 AM, Brian West wrote: > >> This looks like gcc is segfaulting can you provide me a complete backtrace >> of the core file that dumps from FreeSWITCH? >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> It sounds like you might have bad ram or bad hardware... gcc crashing is >> usually a sign something is really wrong with your machine. >> >> /b >> >> On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: >> >> but without luck... >> >> ode1:/opt/freeswitch-trunk# >> node1:/opt/freeswitch-trunk# sudo make >> make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' >> make all-am >> make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' >> g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF >> .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc >> g++: Internal error: Segmentation fault (program cc1plus) >> Please submit a full bug report. >> See for instructions. >> make[2]: *** [pcrecpp_unittest.o] Error 1 >> make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >> make[1]: *** [all] Error 2 >> make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >> make: *** [libs/pcre/libpcre.la] Error 2 >> node1:/opt/freeswitch-trunk# >> node1:/opt/freeswitch-trunk# >> >> >> Of course I'm using the latest trunk... >> >> Can anyone help? >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/8ec67a04/attachment.html From anthony.minessale at gmail.com Tue Sep 22 07:39:28 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Sep 2009 09:39:28 -0500 Subject: [Freeswitch-users] Mod_perl $session in not hangup In-Reply-To: <25530646.post@talk.nabble.com> References: <25530646.post@talk.nabble.com> Message-ID: <191c3a030909220739p66502dc3yc207f791bb158f8c@mail.gmail.com> The reason is you cannot complete the hangup until the script exits. On the bright side, if you update to latest trunk it will probably work more how you want it to because a recent change will make this possible. On Tue, Sep 22, 2009 at 4:30 AM, lakshmanan wrote: > > Hi all, I've the following mod_perl program to execute when I call to an > extension (say 777). > I use twinkle as a soft phone, to make calls. > > #!/usr/bin/perl > use strict; > use freeswitch; > our $session; > $session->answer(); > if($session->ready()) > { > my $uuid=$session->getVariable("uuid"); > freeswitch::consoleLog("INFO","UUID is $uuid\n"); > > freeswitch::consoleLog("INFO","Session is answered\n"); > > > $session->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav"); > my $dtmf = $session->getDigits(4,"", 5000); > freeswitch::consoleLog("INFO","I received $dtmf\n"); > $session->hangup("NORMAL_CLEARING"); > sleep(5); > # Some other statements. > } > return 1; > > Everything is fine. > After executing $session->hangup, I got NORMAL_CLEARING in my freeswitch > console. But in my soft phone, still the channel is active for 5 seconds. > The call got ended only after the 5 seconds sleep. > > But if I create my own session like > my $session=new > freeswitch::Session("user/1000"); > and I say $session->hangup(), it got terminated. > > I wanted to know why there is such difference?? or am I wrong?? > Please clarify me. > > > -- > View this message in context: > http://www.nabble.com/Mod_perl-%24session-in-not-hangup-tp25530646p25530646.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/576c8596/attachment.html From anthony.minessale at gmail.com Tue Sep 22 07:42:58 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Sep 2009 09:42:58 -0500 Subject: [Freeswitch-users] Mod_perl or ESL In-Reply-To: <25530677.post@talk.nabble.com> References: <25520023.post@talk.nabble.com> <25530677.post@talk.nabble.com> Message-ID: <191c3a030909220742v1d39aa74lb51b1c8e12953d65@mail.gmail.com> either one will work. The drawback of mod_perl is that the code executes inline so run the risk of a mistake in your perl code making FreeSWITCH become less stable. The drawback of ESL is you are opening a socket connection for each call. On Tue, Sep 22, 2009 at 6:11 AM, Nagalenoj wrote: > > I need to handle some hundreds of call. So, which one can I opt? > > > Nagalenoj wrote: > > > > Dear friends, > > I want to know which is the better way to do route calls and control > > calls. I've did a experiment which can be done in both ways, Mod_perl and > > ESL. I don't know which one is better to take. > > When I see some earlier posts, It is given like Mod_perl has some > > limitations and I don't know what kind of limitations they are., > > Can someone say which is better to use and how it is better? > > > > -- > View this message in context: > http://www.nabble.com/Mod_perl-or-ESL-tp25520023p25530677.html > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/e5964d7e/attachment-0001.html From lyncker at lyth.de Tue Sep 22 08:17:08 2009 From: lyncker at lyth.de (Filip Lyncker) Date: Tue, 22 Sep 2009 17:17:08 +0200 Subject: [Freeswitch-users] Some Newbie questions about dialplan and local Sip registration In-Reply-To: <65d96fc80909220534x2482b1bcvbc1c17ff45e0fad8@mail.gmail.com> References: <4AB39788.3000907@lyth.de> <65d96fc80909181104v2dc2a9ag268a51cd45c28e6a@mail.gmail.com> <4AB8AD67.7040201@lyth.de> <65d96fc80909220534x2482b1bcvbc1c17ff45e0fad8@mail.gmail.com> Message-ID: <4AB8EA74.3060702@lyth.de> Ok *solved* .... I set in my sip.conf (asterisk) now nat=true, b/c the asterisk ansered the packets sent from lan_ip to the external_ip. now it works, but its not the perfect solution because FS seems to send the packets with an nat envelope or flag. How can i avoid this? the next thing is the dialplan, wich doesnt work at all for me ! ( see my other post with sip registrares) ... if I call now a number , the following entry should route it to my asterisk-gw : but it doesnt and FS says : freeswitch at Bigfish> 2009-09-22 17:10:16.776629 [NOTICE] switch_channel.c:602 New Channel sofia/internal/22 at 192.168.1.34 [733236b0-be36-0049-8ace-a2903921fd81] 2009-09-22 17:10:16.781511 [INFO] mod_dialplan_xml.c:315 Processing 22->01776721280 in context default 2009-09-22 17:10:16.800065 [NOTICE] switch_ivr.c:1349 Transfer sofia/internal/22 at 192.168.1.34 to enum[01776721280 at default] 2009-09-22 17:10:26.800401 [INFO] switch_core_state_machine.c:136 No Route, Aborting 2009-09-22 17:10:26.800401 [NOTICE] switch_core_state_machine.c:137 Hangup sofia/internal/22 at 192.168.1.34 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2009-09-22 17:10:26.800401 [NOTICE] switch_core_session.c:1086 Session 3 (sofia/internal/22 at 192.168.1.34) Ended 2009-09-22 17:10:26.800401 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/22 at 192.168.1.34 [CS_DESTROY] what's wrong with my dialplan ? thanks again for help, regards filip Tihomir Culjaga schrieb: > hmmm .. can you register using x-lite or some other softphone with the > same credentials? > > can you paste a siptrace of the failed registration? > > > BTW: Make sure nothing is already registered with this credentials > when you try with FS > > T. > > On Tue, Sep 22, 2009 at 12:56 PM, Filip Lyncker > wrote: > > Hi Tihomir, > > Thanks for your help , I added the Asteriskparameters as you described > below, but I still get the same timeout error: > 2009-09-22 12:50:52.261103 [WARNING] sofia_reg.c:364 asterisk Failed > Registration, setting retry to 270 seconds. > 2009-09-22 12:50:54.324447 [ERR] sofia_reg.c:1460 asterisk > Registration > Failed with status Request Timeout [408]. failure #9 > > Now, my gateway entry looks like the following : > > > > > > > > > > > > > > > What can be still wrong here? > > Regards, > > Filip > > > > Tihomir Culjaga schrieb: > > hi Filip, > > > > > > for calling a user... please read this first: > > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_A_Registered_User > > for making a GW register into e.g. asterisk please use this: > > > > > > > > > > > > > > > > > > > > > > > > > > this should be enough to register the GW... after that please read > > this: > > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Dialing_through_gateways > > > > > > in your case it will be something like this: > > > > > > > expression="^(NUMBER_TO_SEND_TO_ASTERISK)$"> > > > > > > > > > > > > > > > > > > > > > > > > > > On Fri, Sep 18, 2009 at 4:22 PM, Filip Lyncker > > >> wrote: > > > > Hi List, > > > > for the first experiments with freeswitch I downloaded the > Windows > > installation. > > Now Im trying to get my 2 Sipphones get connected to. Later > I want > > connect the freeswitch to my asterisk gateway. > > > > I find the examples pretty complex therfore Im trying to > build up a > > simple solution to understand the functions from the scratch .. > > > > my current problem is , that I cant route my local sips to each > > other ( > > registration seems to work now). > > the next is , that freeshwitch is not able to connect to > asterisk. > > but I > > will describe this later. > > > > I installed in the Directory a xml file ( called 22.xml) > with the > > following content : > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="22"> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="24"> > > > > > > > > > > > > This seems to be ok now. Now I want to dial from 22 to 24 , > > wherefore I > > configured this dialplan : > > > > > > > > > > > > data="user/${dialed_extension}"/> > > > > > > > > > > wich doesnt work , mybe b/c the user/${dialed_extension} I dont > > know... > > Freeswitch says: > > [INFO] switch_core_state_machine.c:136 No Route, Aborting > > [NOTICE] switch_core_state_machine.c:137 Hangup > > sofia/internal/24 at 192.168.1.34 > > > > [CS_ROUTING] [NO_ROUTE_DESTINATION] > > [NOTICE] switch_core_session.c:1086 Session 17 > > (sofia/internal/24 at 192.168.1.34 > >) Ended > > [NOTICE] switch_core_session.c:1088 Close Channel > > sofia/internal/24 at 192.168.1.34 > > [CS_DESTROY] > > > > Im sure , for you guys this cant be a big deal;) > > > > > > Next Point is my Asterisk registration , mybe you can help > me out here > > to .. : > > > > In the sip-profiles/external I installed the my_asterisk.xml > with that > > content : > > > > > > > > > > > > > > > > > > > > > > > > Freeswitch allways complains a timeout error : > > [ERR] sofia_reg.c:1460 asterisk Registration Failed with status > > Request > > Timeout [408]. failure #17 > > [WARNING] sofia_reg.c:364 asterisk Failed Registration, > setting retry > > to 540 seconds. > > > > it seems that It cant connect ( I also tried out to explicit > set the > > port to 5060 b/c I read something about 5080 .. : > name="sip-port" > > value="5060"> but this didnt help) > > In my Asterisk I set in the sip.conf the entry 28 with the > pw test > > .... > > > > > > If someone could help me with my first steps I would be verrry > > thankful ;)) > > > > cheers > > > > > > Filip > > > > -- > > _________________________________ > > Filip Lyncker, Dipl.-Inform. (FH) > > > > > > Lyncker & Theis GmbH > > Wilhelmstr. 16 > > 65185 Wiesbaden > > Germany > > > > Fon +49 611/9006951 > > Fax +49 611/9406125 > > > > > > Handelsregister: HRB 23156 Amtsgericht Wiesbaden > > Steuernummer: 4023897051 > > USt-IdNr.: DE255806399 > > > > Gesch?ftsf?hrer: > > Filip Lyncker, > > Armin Theis > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > _________________________________ > Filip Lyncker, Dipl.-Inform. (FH) > > > Lyncker & Theis GmbH > Wilhelmstr. 16 > 65185 Wiesbaden > Germany > > Fon +49 611/9006951 > Fax +49 611/9406125 > > > Handelsregister: HRB 23156 Amtsgericht Wiesbaden > Steuernummer: 4023897051 > USt-IdNr.: DE255806399 > > Gesch?ftsf?hrer: > Filip Lyncker, > Armin Theis > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- _________________________________ Filip Lyncker, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 4023897051 USt-IdNr.: DE255806399 Gesch?ftsf?hrer: Filip Lyncker, Armin Theis From nicolas at medularis.com Tue Sep 22 09:24:52 2009 From: nicolas at medularis.com (Nicolas Brenner) Date: Tue, 22 Sep 2009 12:24:52 -0400 Subject: [Freeswitch-users] ALLOTTED_TIMEOUT hangup cause? In-Reply-To: <1b46b4e80909211741n2de3191fvb64a61eab3a3259e@mail.gmail.com> References: <1b46b4e80909211741n2de3191fvb64a61eab3a3259e@mail.gmail.com> Message-ID: <1b46b4e80909220924m6bdeb893o2a02e826bf7d6396@mail.gmail.com> Did a little more digging, ALLOTTED_TIMEOUT has an error code of 602 according to the Wiki (http://wiki.freeswitch.org/wiki/Hangup_causes) nevertheless that code is not covered in RFC 4497 ( http://tools.ietf.org/html/rfc4497) On Mon, Sep 21, 2009 at 8:41 PM, Nicolas Brenner wrote: > Hi, > > Today, while trying to bridge some calls I started to get a > ALLOTTED_TIMEOUT hangup cause on the second leg. I looked for info on the > Wiki and Google, but I couldn't find a detailed explanation. Does anybody > know what does it mean exactly? > > Thanks! > > Nicolas > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/258d5798/attachment.html From tculjaga at gmail.com Tue Sep 22 09:29:45 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 22 Sep 2009 18:29:45 +0200 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> Message-ID: <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> Hi Anthony, it is not the machine ... and yep there was some memory related issue ... but this was caused by my module .... So, to summarize.. i had two issues: 1. FS crashing without any notice (at 5 CPS) 2. Unable to recompile FS with gdb support The first issue was actually related to "-hp" switch i was using in my startup script. With it, FS was crashing without any notice (even on low traffic) and regardless if i load my custom modules or not. The second issue was related to many FS crashes having my module loaded... I found it later and fixed that. So, after the machine cleanup I rebuild FS with gdb support without any issues. Of course i sow this log .. but i didn't realize for a while... and after that i was fighting with crashes caused by "-hp" ... also, it was quite late as well ended up at 3 AM :P Anyhow, the poit is; FS works well with my custom module. It just finished 2 mil. calls (with media) at 100 CPS having ~1600 simultaneous calls... well, thats something :P. T. On Tue, Sep 22, 2009 at 4:35 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > see this from your own log? > > make[2]: Entering directory `/opt/freeswitch-trunk/libs/ > pcre' > g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF > .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc > g++: Internal error: Segmentation fault (program cc1plus) > Please submit a full bug report. > See for instructions. > make[2]: *** [pcrecpp_unittest.o] Error 1 > make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' > make: *** [libs/pcre/libpcre.la] Error 2 > > > This is a FATAL error to have on your machine. > It's failing during the build. This is your compiler crashing while trying > to build the software. > This is very bad. > You most likely have a hardware failure and need to replace the machine or > at the very least all of the memory chips. > > > > On Tue, Sep 22, 2009 at 7:36 AM, Tihomir Culjaga wrote: > >> hi Brian, >> >> well, there is no coredump at all... and when i start FS with gdb it >> doesn't crash :P >> I need to do some more testing and will come back to you. >> >> T. >> >> On Tue, Sep 22, 2009 at 1:22 AM, Brian West wrote: >> >>> This looks like gcc is segfaulting can you provide me a complete >>> backtrace of the core file that dumps from FreeSWITCH? >>> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>> >>> It sounds like you might have bad ram or bad hardware... gcc crashing is >>> usually a sign something is really wrong with your machine. >>> >>> /b >>> >>> On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: >>> >>> but without luck... >>> >>> ode1:/opt/freeswitch-trunk# >>> node1:/opt/freeswitch-trunk# sudo make >>> make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' >>> make all-am >>> make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' >>> g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF >>> .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc >>> g++: Internal error: Segmentation fault (program cc1plus) >>> Please submit a full bug report. >>> See for instructions. >>> make[2]: *** [pcrecpp_unittest.o] Error 1 >>> make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>> make[1]: *** [all] Error 2 >>> make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>> make: *** [libs/pcre/libpcre.la] Error 2 >>> node1:/opt/freeswitch-trunk# >>> node1:/opt/freeswitch-trunk# >>> >>> >>> Of course I'm using the latest trunk... >>> >>> Can anyone help? >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/515ecdc4/attachment-0001.html From msc at freeswitch.org Tue Sep 22 09:34:18 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Sep 2009 09:34:18 -0700 Subject: [Freeswitch-users] Can this be done in FreeSWITCH? In-Reply-To: References: Message-ID: <87f2f3b90909220934o60305a52xa1abfa39d5d67154@mail.gmail.com> How is the DID transported? SIP, PRI, analog DID trunks? -MC On Mon, Sep 21, 2009 at 9:17 PM, Francis Vidal wrote: > Hi all, > Consider the following scenario: Calling party --> DID provider --> Cisco > AS5300 --> POTS provider --> Called party > > The Calling party calls a number provided by the DID provider. This is then > processed by the AS5300 facing the POTS provider to do the following number > translation: ANI = 1 + DNIS (the ANI assumes the identify of the DNIS > prefixed with "1"). The Cisco AS5300 then sends a "prefix" which is actually > the number of the Called party in their system (of the POTS provider). > However, the Cisco AS5300 has a finite limit on the number of translations > (approx. 128-300 translations). Can the number translation be done on > FreeSWITCH instead? > > Calling party --> DID provider --> FreeSWITCH --> Cisco AS5300 --> POTS > provider --> Called party > > This can also evolve into: > > Calling party --> DID provider --> FreeSWITCH --> Cisco AS5300[1] --> > POTS provider --> Called party > \ > / > +-----> > Cisco AS5300[2] --->+ > > If we wanted to increase the number of ports the POTS provider. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/77825922/attachment.html From brian at freeswitch.org Tue Sep 22 09:41:05 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Sep 2009 11:41:05 -0500 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> Message-ID: <88EB1A08-BF4C-481D-9811-9EC1107BA19B@freeswitch.org> The issue is you clearly show GCC crashing trying to compile freeswitch which is BAD that indicates a larger problem with the hardware or memory. Its physical issues not logical ones. /b On Sep 22, 2009, at 11:29 AM, Tihomir Culjaga wrote: > Hi Anthony, > > it is not the machine ... and yep there was some memory related > issue ... but this was caused by my module .... From tculjaga at gmail.com Tue Sep 22 09:49:29 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 22 Sep 2009 18:49:29 +0200 Subject: [Freeswitch-users] Can this be done in FreeSWITCH? In-Reply-To: References: Message-ID: <65d96fc80909220949v5de8fc54ia0f1932c0def74b2@mail.gmail.com> so, you say ... CallingParty => AS5300 A: aNum B: didNum AS5300 => PSTN A: 1 + didNum B: prefix (actually the PSTN subscriber's number) well, without a doubt... you can manipulate whatever number you want ... you just need to find the best way to do it. This depends of the number of DIDs you would like to host. You can do a DB lookup to retrieve the prefix / Subscriber Number... or you can do it inline in your dialplan. It really depends of how much you need to scale. T. On Tue, Sep 22, 2009 at 6:17 AM, Francis Vidal wrote: > Hi all, > Consider the following scenario: Calling party --> DID provider --> Cisco > AS5300 --> POTS provider --> Called party > > The Calling party calls a number provided by the DID provider. This is then > processed by the AS5300 facing the POTS provider to do the following number > translation: ANI = 1 + DNIS (the ANI assumes the identify of the DNIS > prefixed with "1"). The Cisco AS5300 then sends a "prefix" which is actually > the number of the Called party in their system (of the POTS provider). > However, the Cisco AS5300 has a finite limit on the number of translations > (approx. 128-300 translations). Can the number translation be done on > FreeSWITCH instead? > > Calling party --> DID provider --> FreeSWITCH --> Cisco AS5300 --> POTS > provider --> Called party > > This can also evolve into: > > Calling party --> DID provider --> FreeSWITCH --> Cisco AS5300[1] --> > POTS provider --> Called party > \ > / > +-----> > Cisco AS5300[2] --->+ > > If we wanted to increase the number of ports the POTS provider. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/9b0e26ae/attachment.html From tculjaga at gmail.com Tue Sep 22 09:51:24 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 22 Sep 2009 18:51:24 +0200 Subject: [Freeswitch-users] Can this be done in FreeSWITCH? In-Reply-To: <65d96fc80909220949v5de8fc54ia0f1932c0def74b2@mail.gmail.com> References: <65d96fc80909220949v5de8fc54ia0f1932c0def74b2@mail.gmail.com> Message-ID: <65d96fc80909220951w335c9246j9a3f562b5941d1a4@mail.gmail.com> well .. it is AS .. it can be SIP or H323 ... well if it is hooked to a PGW it is MGCP but i doubt... so it is either SIP or H323. i will put a nickel for H323 :P T. On Tue, Sep 22, 2009 at 6:49 PM, Tihomir Culjaga wrote: > so, you say ... > > CallingParty => AS5300 > > A: aNum > B: didNum > > > AS5300 => PSTN > > A: 1 + didNum > B: prefix (actually the PSTN subscriber's number) > > > well, without a doubt... you can manipulate whatever number you want ... > you just need to find the best way to do it. This depends of the number of > DIDs you would like to host. You can do a DB lookup to retrieve the prefix / > Subscriber Number... or you can do it inline in your dialplan. It really > depends of how much you need to scale. > > > T. > > > > > > On Tue, Sep 22, 2009 at 6:17 AM, Francis Vidal wrote: > >> Hi all, >> Consider the following scenario: Calling party --> DID provider --> Cisco >> AS5300 --> POTS provider --> Called party >> >> The Calling party calls a number provided by the DID provider. This is >> then processed by the AS5300 facing the POTS provider to do the following >> number translation: ANI = 1 + DNIS (the ANI assumes the identify of the DNIS >> prefixed with "1"). The Cisco AS5300 then sends a "prefix" which is actually >> the number of the Called party in their system (of the POTS provider). >> However, the Cisco AS5300 has a finite limit on the number of translations >> (approx. 128-300 translations). Can the number translation be done on >> FreeSWITCH instead? >> >> Calling party --> DID provider --> FreeSWITCH --> Cisco AS5300 --> >> POTS provider --> Called party >> >> This can also evolve into: >> >> Calling party --> DID provider --> FreeSWITCH --> Cisco AS5300[1] --> >> POTS provider --> Called party >> \ >> / >> +-----> >> Cisco AS5300[2] --->+ >> >> If we wanted to increase the number of ports the POTS provider. >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/f1c5e7c8/attachment.html From diego.viola at gmail.com Tue Sep 22 10:10:37 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 22 Sep 2009 17:10:37 +0000 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> Message-ID: <86a32abc0909221010o18be55c6m72c2674ea4ca5078@mail.gmail.com> Why don't you try to do the same on another machine to see if you get the same results? I think it's hardware related as Anthony and Brian pointed out. Diego On Tue, Sep 22, 2009 at 4:29 PM, Tihomir Culjaga wrote: > Hi Anthony, > > it is not the machine ... and yep there was some memory related issue ... > but this was caused by my module .... > > So, to summarize.. i had two issues: > > > 1. FS crashing without any notice (at 5 CPS) > 2. Unable to recompile FS with gdb support > > > > The first issue was actually related to "-hp" switch i was using in my > startup script. With it, FS was crashing without any notice (even on low > traffic) and regardless if i load my custom modules or not. > The second issue was related to many FS crashes having my module loaded... > I found it later and fixed that. > > > So, after the machine cleanup I rebuild FS with gdb support without any > issues. > Of course i sow this log .. but i didn't realize for a while... and after > that i was fighting with crashes caused by "-hp" ... also, it was quite late > as well ended up at 3 AM :P > > > > Anyhow, the poit is; FS works well with my custom module. It just finished > 2 mil. calls (with media) at 100 CPS having ~1600 simultaneous calls... > well, thats something :P. > > > > T. > > > > On Tue, Sep 22, 2009 at 4:35 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> see this from your own log? >> >> make[2]: Entering directory `/opt/freeswitch-trunk/libs/ >> pcre' >> g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF >> .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc >> g++: Internal error: Segmentation fault (program cc1plus) >> Please submit a full bug report. >> See for instructions. >> make[2]: *** [pcrecpp_unittest.o] Error 1 >> make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >> make[1]: *** [all] Error 2 >> make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >> make: *** [libs/pcre/libpcre.la] Error 2 >> >> >> This is a FATAL error to have on your machine. >> It's failing during the build. This is your compiler crashing while >> trying to build the software. >> This is very bad. >> You most likely have a hardware failure and need to replace the machine or >> at the very least all of the memory chips. >> >> >> >> On Tue, Sep 22, 2009 at 7:36 AM, Tihomir Culjaga wrote: >> >>> hi Brian, >>> >>> well, there is no coredump at all... and when i start FS with gdb it >>> doesn't crash :P >>> I need to do some more testing and will come back to you. >>> >>> T. >>> >>> On Tue, Sep 22, 2009 at 1:22 AM, Brian West wrote: >>> >>>> This looks like gcc is segfaulting can you provide me a complete >>>> backtrace of the core file that dumps from FreeSWITCH? >>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>>> >>>> It sounds like you might have bad ram or bad hardware... gcc crashing is >>>> usually a sign something is really wrong with your machine. >>>> >>>> /b >>>> >>>> On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: >>>> >>>> but without luck... >>>> >>>> ode1:/opt/freeswitch-trunk# >>>> node1:/opt/freeswitch-trunk# sudo make >>>> make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' >>>> make all-am >>>> make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' >>>> g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF >>>> .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc >>>> g++: Internal error: Segmentation fault (program cc1plus) >>>> Please submit a full bug report. >>>> See for instructions. >>>> make[2]: *** [pcrecpp_unittest.o] Error 1 >>>> make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>>> make[1]: *** [all] Error 2 >>>> make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>>> make: *** [libs/pcre/libpcre.la] Error 2 >>>> node1:/opt/freeswitch-trunk# >>>> node1:/opt/freeswitch-trunk# >>>> >>>> >>>> Of course I'm using the latest trunk... >>>> >>>> Can anyone help? >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at 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_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/fac9dfdf/attachment-0001.html From tculjaga at gmail.com Tue Sep 22 10:46:38 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 22 Sep 2009 19:46:38 +0200 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <88EB1A08-BF4C-481D-9811-9EC1107BA19B@freeswitch.org> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> <88EB1A08-BF4C-481D-9811-9EC1107BA19B@freeswitch.org> Message-ID: <65d96fc80909221046r2e222dfagefc9afcb18545457@mail.gmail.com> hmmm, how to track that down? this is gonna be tricky... i have another machine but quite different i can try on that as well and we will see.... T. On Tue, Sep 22, 2009 at 6:41 PM, Brian West wrote: > The issue is you clearly show GCC crashing trying to compile > freeswitch which is BAD that indicates a larger problem with the > hardware or memory. Its physical issues not logical ones. > > /b > > On Sep 22, 2009, at 11:29 AM, Tihomir Culjaga wrote: > > > Hi Anthony, > > > > it is not the machine ... and yep there was some memory related > > issue ... but this was caused by my module .... > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/375736d7/attachment.html From mike at jerris.com Tue Sep 22 11:03:19 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Sep 2009 14:03:19 -0400 Subject: [Freeswitch-users] CudaTel Communications Server Version 1.0 Released In-Reply-To: <13ca621c0909211613r6aa5c683g1a777b37b3d32f39@mail.gmail.com> References: <87f2f3b90909211232o14b1a8b5w7369b47c8ab48274@mail.gmail.com> <13ca621c0909211613r6aa5c683g1a777b37b3d32f39@mail.gmail.com> Message-ID: <03601B75-D3E1-4ACF-B1F5-C2A83F05B551@jerris.com> It is down at the bottom of the email Mike On Sep 21, 2009, at 7:13 PM, Gavin Henry wrote: > URL??? > > On 21/09/2009, Michael Collins wrote: >> At ClueCon 2009 we had an exciting announcement: Barracuda Networks >> and the >> FreeSWITCH team have been working together to create a new PBX >> appliance. >> Dubbed the CudaTel Communications Server, this new communications >> platform >> is both feature-rich and easy-to-use. We are pleased to announce that >> version 1.0 of the CudaTel Communcations Server has been released! >> >> The feature list for this affordable system is impressive: >> Automatic phone provisioning >> Multi-party conferencing >> Group calling >> SIP phone and provider support >> Automated attendant >> Voicemail >> TMD hardware option >> High definition codec support (G.722, G.722.1, G.722.1c) >> Call recording >> Active Directory and LDAP integration >> Encrypted VoIP support >> >> Many more features are included, all of which are controlled by an >> intuitive >> Web-based interface. >> >> We invite you to visit the CudaTel >> website or call >> 989-720-4000 for more information or to request evaluation units. >> >> -The FreeSWITCH Team >> From diego.viola at gmail.com Tue Sep 22 11:18:36 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 22 Sep 2009 18:18:36 +0000 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <65d96fc80909221046r2e222dfagefc9afcb18545457@mail.gmail.com> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> <88EB1A08-BF4C-481D-9811-9EC1107BA19B@freeswitch.org> <65d96fc80909221046r2e222dfagefc9afcb18545457@mail.gmail.com> Message-ID: <86a32abc0909221118i6213f0c8q2971b25d9513795b@mail.gmail.com> Why don't you try changing RAM? Or run memtest86 or try another machine? Or... On Tue, Sep 22, 2009 at 5:46 PM, Tihomir Culjaga wrote: > hmmm, how to track that down? > this is gonna be tricky... > > > i have another machine but quite different i can try on that as well and we > will see.... > > T. > > > On Tue, Sep 22, 2009 at 6:41 PM, Brian West wrote: > >> The issue is you clearly show GCC crashing trying to compile >> freeswitch which is BAD that indicates a larger problem with the >> hardware or memory. Its physical issues not logical ones. >> >> /b >> >> On Sep 22, 2009, at 11:29 AM, Tihomir Culjaga wrote: >> >> > Hi Anthony, >> > >> > it is not the machine ... and yep there was some memory related >> > issue ... but this was caused by my module .... >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/4fb10f89/attachment.html From anthony.minessale at gmail.com Tue Sep 22 11:42:59 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Sep 2009 13:42:59 -0500 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> Message-ID: <191c3a030909221142o4e3251c7k8db2769640f9100e@mail.gmail.com> One of the things that -hp does is call "memlockall" which disables swapping which uses more memory which makes hitting a land mine in your ram chip much more likely. On the other hand: Since you are talking about "with" and "without" gcc support I am going to guess you are on Solaris which you probably should have mentioned before. it's possible that some of the more aggressive things activated by -hp is not possible on that platform. If so we either have to identify that and disable it or disable hp completely for Solaris. Either way, gcc randomly crashing is never ok and is a symptom of a pretty serious issue. Are you using 2 separate fresh checkouts for both suncc and gcc builds because it's not possible to switch the same source tree once it's already configured for one of them. On Tue, Sep 22, 2009 at 11:29 AM, Tihomir Culjaga wrote: > Hi Anthony, > > it is not the machine ... and yep there was some memory related issue ... > but this was caused by my module .... > > So, to summarize.. i had two issues: > > > 1. FS crashing without any notice (at 5 CPS) > 2. Unable to recompile FS with gdb support > > > > The first issue was actually related to "-hp" switch i was using in my > startup script. With it, FS was crashing without any notice (even on low > traffic) and regardless if i load my custom modules or not. > The second issue was related to many FS crashes having my module loaded... > I found it later and fixed that. > > > So, after the machine cleanup I rebuild FS with gdb support without any > issues. > Of course i sow this log .. but i didn't realize for a while... and after > that i was fighting with crashes caused by "-hp" ... also, it was quite late > as well ended up at 3 AM :P > > > > Anyhow, the poit is; FS works well with my custom module. It just finished > 2 mil. calls (with media) at 100 CPS having ~1600 simultaneous calls... > well, thats something :P. > > > > T. > > > > On Tue, Sep 22, 2009 at 4:35 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> see this from your own log? >> >> make[2]: Entering directory `/opt/freeswitch-trunk/libs/ >> pcre' >> g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF >> .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc >> g++: Internal error: Segmentation fault (program cc1plus) >> Please submit a full bug report. >> See for instructions. >> make[2]: *** [pcrecpp_unittest.o] Error 1 >> make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >> make[1]: *** [all] Error 2 >> make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >> make: *** [libs/pcre/libpcre.la] Error 2 >> >> >> This is a FATAL error to have on your machine. >> It's failing during the build. This is your compiler crashing while >> trying to build the software. >> This is very bad. >> You most likely have a hardware failure and need to replace the machine or >> at the very least all of the memory chips. >> >> >> >> On Tue, Sep 22, 2009 at 7:36 AM, Tihomir Culjaga wrote: >> >>> hi Brian, >>> >>> well, there is no coredump at all... and when i start FS with gdb it >>> doesn't crash :P >>> I need to do some more testing and will come back to you. >>> >>> T. >>> >>> On Tue, Sep 22, 2009 at 1:22 AM, Brian West wrote: >>> >>>> This looks like gcc is segfaulting can you provide me a complete >>>> backtrace of the core file that dumps from FreeSWITCH? >>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>>> >>>> It sounds like you might have bad ram or bad hardware... gcc crashing is >>>> usually a sign something is really wrong with your machine. >>>> >>>> /b >>>> >>>> On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: >>>> >>>> but without luck... >>>> >>>> ode1:/opt/freeswitch-trunk# >>>> node1:/opt/freeswitch-trunk# sudo make >>>> make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' >>>> make all-am >>>> make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' >>>> g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF >>>> .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc >>>> g++: Internal error: Segmentation fault (program cc1plus) >>>> Please submit a full bug report. >>>> See for instructions. >>>> make[2]: *** [pcrecpp_unittest.o] Error 1 >>>> make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>>> make[1]: *** [all] Error 2 >>>> make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>>> make: *** [libs/pcre/libpcre.la] Error 2 >>>> node1:/opt/freeswitch-trunk# >>>> node1:/opt/freeswitch-trunk# >>>> >>>> >>>> Of course I'm using the latest trunk... >>>> >>>> Can anyone help? >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at 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_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/531c8738/attachment-0001.html From anthony.minessale at gmail.com Tue Sep 22 11:36:17 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 22 Sep 2009 13:36:17 -0500 Subject: [Freeswitch-users] CudaTel Communications Server Version 1.0 Released In-Reply-To: <200909221155.16510.hads@nice.net.nz> References: <87f2f3b90909211232o14b1a8b5w7369b47c8ab48274@mail.gmail.com> <13ca621c0909211613r6aa5c683g1a777b37b3d32f39@mail.gmail.com> <200909221155.16510.hads@nice.net.nz> Message-ID: <191c3a030909221136m1e49b861o94da3e98447d6300@mail.gmail.com> Nice link, Are you offering to become a reseller? On Mon, Sep 21, 2009 at 6:55 PM, Hadley Rich wrote: > On Tue, 22 Sep 2009 11:13:13 Gavin Henry wrote: > > URL??? > > > > On 21/09/2009, Michael Collins wrote: > > > We invite you to visit the CudaTel website > or > > > call 989-720-4000 for more information or to request evaluation units. > -- > https://nicegear.co.nz > VoIP and Open Source Hardware > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/e6a49d05/attachment.html From hads at nice.net.nz Tue Sep 22 12:01:18 2009 From: hads at nice.net.nz (Hadley Rich) Date: Wed, 23 Sep 2009 07:01:18 +1200 Subject: [Freeswitch-users] CudaTel Communications Server Version 1.0 Released In-Reply-To: <191c3a030909221136m1e49b861o94da3e98447d6300@mail.gmail.com> References: <87f2f3b90909211232o14b1a8b5w7369b47c8ab48274@mail.gmail.com> <200909221155.16510.hads@nice.net.nz> <191c3a030909221136m1e49b861o94da3e98447d6300@mail.gmail.com> Message-ID: <200909230701.18446.hads@nice.net.nz> On Wed, 23 Sep 2009 06:36:17 Anthony Minessale wrote: > Nice link, > Are you offering to become a reseller? Heh, I was actually trying to quote the original link to the person that asked for it rather than spam with my sig. That said, we'd love to be a reseller for our little part of the world. hads -- https://nicegear.co.nz VoIP and Open Source Hardware From tculjaga at gmail.com Tue Sep 22 12:04:25 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 22 Sep 2009 21:04:25 +0200 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <191c3a030909221142o4e3251c7k8db2769640f9100e@mail.gmail.com> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> <191c3a030909221142o4e3251c7k8db2769640f9100e@mail.gmail.com> Message-ID: <65d96fc80909221204u18e0662drfdd64c3c7122b4a1@mail.gmail.com> Hi, Nope, I'm still on Debian 5.0... in transit to CentOS 5.3 but it needs to wait a bit. i was talking about gdb, not gcc and was trying to recompile FS with debug symbols on: CFLAGS="-g -ggdb" MOD_CFLAGS="-g -ggdb". yes, I understand that gcc segfault most probably means only one thing... HW isues. This is sometihng that I'm going to check tomorrow.... running memtest to see what i get. Also, I will repeat the same test with a new block of RAM. Maybe i didn't explain myself well... apologize. T. On Tue, Sep 22, 2009 at 8:42 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > One of the things that -hp does is call "memlockall" which disables > swapping which uses more memory which makes hitting a land mine in your ram > chip much more likely. > > On the other hand: > > Since you are talking about "with" and "without" gcc support I am going to > guess you are on Solaris which you probably should have mentioned before. > it's possible that some of the more aggressive things activated by -hp is > not possible on that platform. If so we either have to identify that and > disable it or disable hp completely for Solaris. > > Either way, gcc randomly crashing is never ok and is a symptom of a pretty > serious issue. > > Are you using 2 separate fresh checkouts for both suncc and gcc builds > because it's not possible to switch the same source tree once it's already > configured for one of them. > > > > On Tue, Sep 22, 2009 at 11:29 AM, Tihomir Culjaga wrote: > >> Hi Anthony, >> >> it is not the machine ... and yep there was some memory related issue ... >> but this was caused by my module .... >> >> So, to summarize.. i had two issues: >> >> >> 1. FS crashing without any notice (at 5 CPS) >> 2. Unable to recompile FS with gdb support >> >> >> >> The first issue was actually related to "-hp" switch i was using in my >> startup script. With it, FS was crashing without any notice (even on low >> traffic) and regardless if i load my custom modules or not. >> The second issue was related to many FS crashes having my module loaded... >> I found it later and fixed that. >> >> >> So, after the machine cleanup I rebuild FS with gdb support without any >> issues. >> Of course i sow this log .. but i didn't realize for a while... and after >> that i was fighting with crashes caused by "-hp" ... also, it was quite late >> as well ended up at 3 AM :P >> >> >> >> Anyhow, the poit is; FS works well with my custom module. It just finished >> 2 mil. calls (with media) at 100 CPS having ~1600 simultaneous calls... >> well, thats something :P. >> >> >> >> T. >> >> >> >> On Tue, Sep 22, 2009 at 4:35 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> see this from your own log? >>> >>> make[2]: Entering directory `/opt/freeswitch-trunk/libs/ >>> pcre' >>> g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF >>> .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc >>> g++: Internal error: Segmentation fault (program cc1plus) >>> Please submit a full bug report. >>> See for instructions. >>> make[2]: *** [pcrecpp_unittest.o] Error 1 >>> make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>> make[1]: *** [all] Error 2 >>> make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>> make: *** [libs/pcre/libpcre.la] Error 2 >>> >>> >>> This is a FATAL error to have on your machine. >>> It's failing during the build. This is your compiler crashing while >>> trying to build the software. >>> This is very bad. >>> You most likely have a hardware failure and need to replace the machine >>> or at the very least all of the memory chips. >>> >>> >>> >>> On Tue, Sep 22, 2009 at 7:36 AM, Tihomir Culjaga wrote: >>> >>>> hi Brian, >>>> >>>> well, there is no coredump at all... and when i start FS with gdb it >>>> doesn't crash :P >>>> I need to do some more testing and will come back to you. >>>> >>>> T. >>>> >>>> On Tue, Sep 22, 2009 at 1:22 AM, Brian West wrote: >>>> >>>>> This looks like gcc is segfaulting can you provide me a complete >>>>> backtrace of the core file that dumps from FreeSWITCH? >>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>>>> >>>>> It sounds like you might have bad ram or bad hardware... gcc crashing >>>>> is usually a sign something is really wrong with your machine. >>>>> >>>>> /b >>>>> >>>>> On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: >>>>> >>>>> but without luck... >>>>> >>>>> ode1:/opt/freeswitch-trunk# >>>>> node1:/opt/freeswitch-trunk# sudo make >>>>> make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' >>>>> make all-am >>>>> make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' >>>>> g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF >>>>> .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc >>>>> g++: Internal error: Segmentation fault (program cc1plus) >>>>> Please submit a full bug report. >>>>> See for instructions. >>>>> make[2]: *** [pcrecpp_unittest.o] Error 1 >>>>> make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>>>> make[1]: *** [all] Error 2 >>>>> make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>>>> make: *** [libs/pcre/libpcre.la] Error 2 >>>>> node1:/opt/freeswitch-trunk# >>>>> node1:/opt/freeswitch-trunk# >>>>> >>>>> >>>>> Of course I'm using the latest trunk... >>>>> >>>>> Can anyone help? >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at 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 at 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_minessale at hotmail.com >>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>> IRC: irc.freenode.net #freeswitch >>> >>> FreeSWITCH Developer Conference >>> sip:888 at conference.freeswitch.org >>> iax:guest at conference.freeswitch.org/888 >>> googletalk:conf+888 at conference.freeswitch.org >>> pstn:213-799-1400 >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/43226718/attachment-0001.html From mike at jerris.com Tue Sep 22 12:04:33 2009 From: mike at jerris.com (Michael Jerris) Date: Tue, 22 Sep 2009 15:04:33 -0400 Subject: [Freeswitch-users] Custom Variables In-Reply-To: <40292.1252445272@a2unlimited.com> References: <40292.1252445272@a2unlimited.com> Message-ID: This should defiantly be in there, please double check if its in a different name, and if not, please post a bug to jira.freeswitch.org. Mike On Sep 8, 2009, at 5:27 PM, Tina Martinez wrote: > Using the verbose-events definitely improved my ability to see the > custom > variables, but now I noticed that the "Member-ID" variable does not > appear in the > DTMF event. > > Would this be related, or did I screw something else up? > > - T From ruda at ruda.com.br Tue Sep 22 12:12:25 2009 From: ruda at ruda.com.br (=?ISO-8859-1?Q?Rud=E1_Cunha?=) Date: Tue, 22 Sep 2009 16:12:25 -0300 Subject: [Freeswitch-users] SUBSCRIBE and NOTIFY Message-ID: <7600794b0909221212x73e23d49va08eb85c355f6c9f@mail.gmail.com> I'm having to configure FreeSWITCH. Baxei version 1.0.4 and I am accessing with the users 1000 and 1001. I register, make the connection. But I'm trying to see to see who is connected (SUBSCRIBE and NOTIFY). But sometimes you work, sometimes you do I connect the other User does not receive the information that I connected (I sent the (SUBSCRIBE and NOTIFY)). That is, sometimes you and I'm connected to another user I'm not, sometimes you work. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/73fa3145/attachment.html From ruda at ruda.com.br Tue Sep 22 12:20:20 2009 From: ruda at ruda.com.br (=?ISO-8859-1?Q?Rud=E1_Cunha?=) Date: Tue, 22 Sep 2009 16:20:20 -0300 Subject: [Freeswitch-users] SUBSCRIBE and NOTIFY In-Reply-To: <7600794b0909221212x73e23d49va08eb85c355f6c9f@mail.gmail.com> References: <7600794b0909221212x73e23d49va08eb85c355f6c9f@mail.gmail.com> Message-ID: <7600794b0909221220y6a43357fs75caeec9c205e0f5@mail.gmail.com> I'm having to configure FreeSWITCH. Download version 1.0.4 and I am accessing with the users 1000 and 1001. I register, make the connection. But I'm trying to see to see who is connected (SUBSCRIBE and NOTIFY). But sometimes you work, sometimes you do I connect the other User does not receive the information that I connected (I sent the (SUBSCRIBE and NOTIFY)). That is, sometimes you and I'm connected to another user I'm not, sometimes you work. ---------- Forwarded message ---------- From: Rud? Cunha Date: 2009/9/22 Subject: SUBSCRIBE and NOTIFY To: freeswitch-users at lists.freeswitch.org I'm having to configure FreeSWITCH. Baxei version 1.0.4 and I am accessing with the users 1000 and 1001. I register, make the connection. But I'm trying to see to see who is connected (SUBSCRIBE and NOTIFY). But sometimes you work, sometimes you do I connect the other User does not receive the information that I connected (I sent the (SUBSCRIBE and NOTIFY)). That is, sometimes you and I'm connected to another user I'm not, sometimes you work. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/923172f6/attachment.html From diego.viola at gmail.com Tue Sep 22 12:32:05 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 22 Sep 2009 19:32:05 +0000 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <65d96fc80909221204u18e0662drfdd64c3c7122b4a1@mail.gmail.com> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> <191c3a030909221142o4e3251c7k8db2769640f9100e@mail.gmail.com> <65d96fc80909221204u18e0662drfdd64c3c7122b4a1@mail.gmail.com> Message-ID: <86a32abc0909221232o6e8a96b3w1e30a28616385fd3@mail.gmail.com> Doesn't FS already compiles with debug symbols by default? On Tue, Sep 22, 2009 at 7:04 PM, Tihomir Culjaga wrote: > Hi, > > Nope, I'm still on Debian 5.0... in transit to CentOS 5.3 but it needs to > wait a bit. > i was talking about gdb, not gcc and was trying to recompile FS with debug > symbols on: CFLAGS="-g -ggdb" MOD_CFLAGS="-g -ggdb". > > yes, I understand that gcc segfault most probably means only one thing... > HW isues. This is sometihng that I'm going to check tomorrow.... running > memtest to see what i get. Also, I will repeat the same test with a new > block of RAM. > > > Maybe i didn't explain myself well... apologize. > > T. > > > > > On Tue, Sep 22, 2009 at 8:42 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> One of the things that -hp does is call "memlockall" which disables >> swapping which uses more memory which makes hitting a land mine in your ram >> chip much more likely. >> >> On the other hand: >> >> Since you are talking about "with" and "without" gcc support I am going to >> guess you are on Solaris which you probably should have mentioned before. >> it's possible that some of the more aggressive things activated by -hp is >> not possible on that platform. If so we either have to identify that and >> disable it or disable hp completely for Solaris. >> >> Either way, gcc randomly crashing is never ok and is a symptom of a pretty >> serious issue. >> >> Are you using 2 separate fresh checkouts for both suncc and gcc builds >> because it's not possible to switch the same source tree once it's already >> configured for one of them. >> >> >> >> On Tue, Sep 22, 2009 at 11:29 AM, Tihomir Culjaga wrote: >> >>> Hi Anthony, >>> >>> it is not the machine ... and yep there was some memory related issue ... >>> but this was caused by my module .... >>> >>> So, to summarize.. i had two issues: >>> >>> >>> 1. FS crashing without any notice (at 5 CPS) >>> 2. Unable to recompile FS with gdb support >>> >>> >>> >>> The first issue was actually related to "-hp" switch i was using in my >>> startup script. With it, FS was crashing without any notice (even on low >>> traffic) and regardless if i load my custom modules or not. >>> The second issue was related to many FS crashes having my module >>> loaded... I found it later and fixed that. >>> >>> >>> So, after the machine cleanup I rebuild FS with gdb support without any >>> issues. >>> Of course i sow this log .. but i didn't realize for a while... and after >>> that i was fighting with crashes caused by "-hp" ... also, it was quite late >>> as well ended up at 3 AM :P >>> >>> >>> >>> Anyhow, the poit is; FS works well with my custom module. It just >>> finished 2 mil. calls (with media) at 100 CPS having ~1600 simultaneous >>> calls... well, thats something :P. >>> >>> >>> >>> T. >>> >>> >>> >>> On Tue, Sep 22, 2009 at 4:35 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> see this from your own log? >>>> >>>> make[2]: Entering directory `/opt/freeswitch-trunk/libs/ >>>> pcre' >>>> g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF >>>> .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc >>>> g++: Internal error: Segmentation fault (program cc1plus) >>>> Please submit a full bug report. >>>> See for instructions. >>>> make[2]: *** [pcrecpp_unittest.o] Error 1 >>>> make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>>> make[1]: *** [all] Error 2 >>>> make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>>> make: *** [libs/pcre/libpcre.la] Error 2 >>>> >>>> >>>> This is a FATAL error to have on your machine. >>>> It's failing during the build. This is your compiler crashing while >>>> trying to build the software. >>>> This is very bad. >>>> You most likely have a hardware failure and need to replace the machine >>>> or at the very least all of the memory chips. >>>> >>>> >>>> >>>> On Tue, Sep 22, 2009 at 7:36 AM, Tihomir Culjaga wrote: >>>> >>>>> hi Brian, >>>>> >>>>> well, there is no coredump at all... and when i start FS with gdb it >>>>> doesn't crash :P >>>>> I need to do some more testing and will come back to you. >>>>> >>>>> T. >>>>> >>>>> On Tue, Sep 22, 2009 at 1:22 AM, Brian West wrote: >>>>> >>>>>> This looks like gcc is segfaulting can you provide me a complete >>>>>> backtrace of the core file that dumps from FreeSWITCH? >>>>>> http://wiki.freeswitch.org/wiki/Reporting_Bugs >>>>>> >>>>>> It sounds like you might have bad ram or bad hardware... gcc crashing >>>>>> is usually a sign something is really wrong with your machine. >>>>>> >>>>>> /b >>>>>> >>>>>> On Sep 21, 2009, at 5:46 PM, Tihomir Culjaga wrote: >>>>>> >>>>>> but without luck... >>>>>> >>>>>> ode1:/opt/freeswitch-trunk# >>>>>> node1:/opt/freeswitch-trunk# sudo make >>>>>> make[1]: Entering directory `/opt/freeswitch-trunk/libs/pcre' >>>>>> make all-am >>>>>> make[2]: Entering directory `/opt/freeswitch-trunk/libs/pcre' >>>>>> g++ -DHAVE_CONFIG_H -I. -O2 -MT pcrecpp_unittest.o -MD -MP -MF >>>>>> .deps/pcrecpp_unittest.Tpo -c -o pcrecpp_unittest.o pcrecpp_unittest.cc >>>>>> g++: Internal error: Segmentation fault (program cc1plus) >>>>>> Please submit a full bug report. >>>>>> See for instructions. >>>>>> make[2]: *** [pcrecpp_unittest.o] Error 1 >>>>>> make[2]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>>>>> make[1]: *** [all] Error 2 >>>>>> make[1]: Leaving directory `/opt/freeswitch-trunk/libs/pcre' >>>>>> make: *** [libs/pcre/libpcre.la] Error 2 >>>>>> node1:/opt/freeswitch-trunk# >>>>>> node1:/opt/freeswitch-trunk# >>>>>> >>>>>> >>>>>> Of course I'm using the latest trunk... >>>>>> >>>>>> Can anyone help? >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 at 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_minessale at hotmail.com >>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >>>> IRC: irc.freenode.net #freeswitch >>>> >>>> FreeSWITCH Developer Conference >>>> sip:888 at conference.freeswitch.org >>>> iax:guest at conference.freeswitch.org/888 >>>> googletalk:conf+888 at conference.freeswitch.org >>>> pstn:213-799-1400 >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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 at 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_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> iax:guest at conference.freeswitch.org/888 >> googletalk:conf+888 at conference.freeswitch.org >> pstn:213-799-1400 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/b70587f3/attachment-0001.html From brian at freeswitch.org Tue Sep 22 13:00:23 2009 From: brian at freeswitch.org (Brian West) Date: Tue, 22 Sep 2009 15:00:23 -0500 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <86a32abc0909221232o6e8a96b3w1e30a28616385fd3@mail.gmail.com> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> <191c3a030909221142o4e3251c7k8db2769640f9100e@mail.gmail.com> <65d96fc80909221204u18e0662drfdd64c3c7122b4a1@mail.gmail.com> <86a32abc0909221232o6e8a96b3w1e30a28616385fd3@mail.gmail.com> Message-ID: yes On Sep 22, 2009, at 2:32 PM, Diego Viola wrote: > Doesn't FS already compiles with debug symbols by default? From diego.viola at gmail.com Tue Sep 22 13:11:41 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 22 Sep 2009 20:11:41 +0000 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> <191c3a030909221142o4e3251c7k8db2769640f9100e@mail.gmail.com> <65d96fc80909221204u18e0662drfdd64c3c7122b4a1@mail.gmail.com> <86a32abc0909221232o6e8a96b3w1e30a28616385fd3@mail.gmail.com> Message-ID: <86a32abc0909221311t45a584b4mb49180c168c00c9a@mail.gmail.com> Then why is Tihomir trying to compile with debug symbols? On Tue, Sep 22, 2009 at 8:00 PM, Brian West wrote: > yes > > On Sep 22, 2009, at 2:32 PM, Diego Viola wrote: > > > Doesn't FS already compiles with debug symbols by default? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/e9c39a72/attachment.html From diego.viola at gmail.com Tue Sep 22 13:12:56 2009 From: diego.viola at gmail.com (Diego Viola) Date: Tue, 22 Sep 2009 20:12:56 +0000 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <86a32abc0909221311t45a584b4mb49180c168c00c9a@mail.gmail.com> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> <191c3a030909221142o4e3251c7k8db2769640f9100e@mail.gmail.com> <65d96fc80909221204u18e0662drfdd64c3c7122b4a1@mail.gmail.com> <86a32abc0909221232o6e8a96b3w1e30a28616385fd3@mail.gmail.com> <86a32abc0909221311t45a584b4mb49180c168c00c9a@mail.gmail.com> Message-ID: <86a32abc0909221312x63396448r47183b0486e31c7b@mail.gmail.com> He's doing an extra effort... just compile it as you would normally and you will have the debug symbols. On Tue, Sep 22, 2009 at 8:11 PM, Diego Viola wrote: > Then why is Tihomir trying to compile with debug symbols? > > > On Tue, Sep 22, 2009 at 8:00 PM, Brian West wrote: > >> yes >> >> On Sep 22, 2009, at 2:32 PM, Diego Viola wrote: >> >> > Doesn't FS already compiles with debug symbols by default? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/2b5deefb/attachment.html From tculjaga at gmail.com Tue Sep 22 13:33:43 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 22 Sep 2009 22:33:43 +0200 Subject: [Freeswitch-users] recompile with gdb In-Reply-To: <86a32abc0909221312x63396448r47183b0486e31c7b@mail.gmail.com> References: <65d96fc80909211546v33f6878fh800e2d4797d78b3@mail.gmail.com> <65d96fc80909220536s5b4b6b43v4cd6b5fc7ec28a4c@mail.gmail.com> <191c3a030909220735q7dbeaa36o54b3332a424104fd@mail.gmail.com> <65d96fc80909220929s42049ea3g4e63905292ceb140@mail.gmail.com> <191c3a030909221142o4e3251c7k8db2769640f9100e@mail.gmail.com> <65d96fc80909221204u18e0662drfdd64c3c7122b4a1@mail.gmail.com> <86a32abc0909221232o6e8a96b3w1e30a28616385fd3@mail.gmail.com> <86a32abc0909221311t45a584b4mb49180c168c00c9a@mail.gmail.com> <86a32abc0909221312x63396448r47183b0486e31c7b@mail.gmail.com> Message-ID: <65d96fc80909221333n6b4dc7d0ha371adf04b5b49e@mail.gmail.com> well ... shame on me :P thx anyway... T. On Tue, Sep 22, 2009 at 10:12 PM, Diego Viola wrote: > He's doing an extra effort... just compile it as you would normally and you > will have the debug symbols. > > > On Tue, Sep 22, 2009 at 8:11 PM, Diego Viola wrote: > >> Then why is Tihomir trying to compile with debug symbols? >> >> >> On Tue, Sep 22, 2009 at 8:00 PM, Brian West wrote: >> >>> yes >>> >>> On Sep 22, 2009, at 2:32 PM, Diego Viola wrote: >>> >>> > Doesn't FS already compiles with debug symbols by default? >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/33bbb436/attachment.html From msc at freeswitch.org Tue Sep 22 14:42:26 2009 From: msc at freeswitch.org (Michael Collins) Date: Tue, 22 Sep 2009 14:42:26 -0700 Subject: [Freeswitch-users] SUBSCRIBE and NOTIFY In-Reply-To: <7600794b0909221220y6a43357fs75caeec9c205e0f5@mail.gmail.com> References: <7600794b0909221212x73e23d49va08eb85c355f6c9f@mail.gmail.com> <7600794b0909221220y6a43357fs75caeec9c205e0f5@mail.gmail.com> Message-ID: <87f2f3b90909221442j59db5f6cr6bb8c596036d73ef@mail.gmail.com> On Tue, Sep 22, 2009 at 12:20 PM, Rud? Cunha wrote: > I'm having to configure FreeSWITCH. > > Download version 1.0.4 and I am accessing with the users 1000 and 1001. > > I register, make the connection. But I'm trying to see to see who is > connected (SUBSCRIBE and NOTIFY). But sometimes you work, sometimes you do I > connect the other User does not receive the information that I connected (I > sent the (SUBSCRIBE and NOTIFY)). That is, sometimes you and I'm connected > to another user I'm not, sometimes you work. > What are you using to see who is connected? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090922/39ec60e7/attachment.html From francisv.list at gmail.com Tue Sep 22 16:30:37 2009 From: francisv.list at gmail.com (Francis Vidal) Date: Wed, 23 Sep 2009 07:30:37 +0800 Subject: [Freeswitch-users] Can this be done in FreeSWITCH? In-Reply-To: <87f2f3b90909220934o60305a52xa1abfa39d5d67154@mail.gmail.com> References: <87f2f3b90909220934o60305a52xa1abfa39d5d67154@mail.gmail.com> Message-ID: Hi Michael, The DID is transported via SIP to the router. From the router to the POTS provider, it's PRI. On Wed, Sep 23, 2009 at 12:34 AM, Michael Collins wrote: > How is the DID transported? SIP, PRI, analog DID trunks? > -MC > > On Mon, Sep 21, 2009 at 9:17 PM, Francis Vidal wrote: > >> Hi all, >> Consider the following scenario: Calling party --> DID provider --> Cisco >> AS5300 --> POTS provider --> Called party >> >> The Calling party calls a number provided by the DID provider. This is >> then processed by the AS5300 facing the POTS provider to do the following >> number translation: ANI = 1 + DNIS (the ANI assumes the identify of the DNIS >> prefixed with "1"). The Cisco AS5300 then sends a "prefix" which is actually >> the number of the Called party in their system (of the POTS provider). >> However, the Cisco AS5300 has a finite limit on the number of translations >> (approx. 128-300 translations). Can the number translation be done on >> FreeSWITCH instead? >> >> Calling party --> DID provider --> FreeSWITCH --> Cisco AS5300 --> >> POTS provider --> Called party >> >> This can also evolve into: >> >> Calling party --> DID provider --> FreeSWITCH --> Cisco AS5300[1] --> >> POTS provider --> Called party >> \ >> / >> +-----> >> Cisco AS5300[2] --->+ >> >> If we wanted to increase the number of ports the POTS provider. >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/7e4f43f5/attachment.html From francisv.list at gmail.com Tue Sep 22 16:32:32 2009 From: francisv.list at gmail.com (Francis Vidal) Date: Wed, 23 Sep 2009 07:32:32 +0800 Subject: [Freeswitch-users] Can this be done in FreeSWITCH? In-Reply-To: <65d96fc80909220949v5de8fc54ia0f1932c0def74b2@mail.gmail.com> References: <65d96fc80909220949v5de8fc54ia0f1932c0def74b2@mail.gmail.com> Message-ID: Yes, this is the desired outcome. I was planning of using FreeSWITCH + MySQL to do this. How do I do this "inline"? On Wed, Sep 23, 2009 at 12:49 AM, Tihomir Culjaga wrote: > so, you say ... > > CallingParty => AS5300 > > A: aNum > B: didNum > > > AS5300 => PSTN > > A: 1 + didNum > B: prefix (actually the PSTN subscriber's number) > > > well, without a doubt... you can manipulate whatever number you want ... > you just need to find the best way to do it. This depends of the number of > DIDs you would like to host. You can do a DB lookup to retrieve the prefix / > Subscriber Number... or you can do it inline in your dialplan. It really > depends of how much you need to scale. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/7fa9ec60/attachment-0001.html From diego.viola at gmail.com Tue Sep 22 21:57:03 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 23 Sep 2009 04:57:03 +0000 Subject: [Freeswitch-users] Multitenancy Message-ID: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> Hello all, How do I configure multi tenant in FS? For example, I want some users to be able to register only with their own domain. Ie: Users: 1000-1010 Domain: foo.org Users: 2000-2010 Domain: bar.org But 1000-1010 shouldn't work with bar.org or 2000-2010 shouldn't work with foo.org. Any ideas how to do that? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/caf5ddf9/attachment.html From diego.viola at gmail.com Tue Sep 22 22:11:45 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 23 Sep 2009 05:11:45 +0000 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> Message-ID: <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> I want user 1000-1010 to belong to foo.org and 2000-2020 to belong to bar.org, and I want both of those domains to have their own dialplan/context. On Wed, Sep 23, 2009 at 4:57 AM, Diego Viola wrote: > Hello all, > > How do I configure multi tenant in FS? > > For example, I want some users to be able to register only with their own > domain. > > Ie: > > Users: 1000-1010 > Domain: foo.org > > Users: 2000-2010 > Domain: bar.org > > But 1000-1010 shouldn't work with bar.org or 2000-2010 shouldn't work with > foo.org. > > Any ideas how to do that? > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/316fd6f2/attachment.html From brian at freeswitch.org Tue Sep 22 22:17:57 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Sep 2009 00:17:57 -0500 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> Message-ID: <249BB60C-79F6-4A3A-8E3B-0E821D4D9046@freeswitch.org> Then setup two domains in your directory and setup proper DNS its really just that simple. /b On Sep 23, 2009, at 12:11 AM, Diego Viola wrote: > I want user 1000-1010 to belong to foo.org and 2000-2020 to belong > to bar.org, and I want both of those domains to have their own > dialplan/context. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/aaf2202d/attachment.html From diego.viola at gmail.com Tue Sep 22 22:22:13 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 23 Sep 2009 05:22:13 +0000 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> Message-ID: <86a32abc0909222222n4af8afbx97ee603cd3b97971@mail.gmail.com> I know I could create different domains on the directory but how do I tell a user to belong to a specific domain? On Wed, Sep 23, 2009 at 5:11 AM, Diego Viola wrote: > I want user 1000-1010 to belong to foo.org and 2000-2020 to belong to > bar.org, and I want both of those domains to have their own > dialplan/context. > > > On Wed, Sep 23, 2009 at 4:57 AM, Diego Viola wrote: > >> Hello all, >> >> How do I configure multi tenant in FS? >> >> For example, I want some users to be able to register only with their own >> domain. >> >> Ie: >> >> Users: 1000-1010 >> Domain: foo.org >> >> Users: 2000-2010 >> Domain: bar.org >> >> But 1000-1010 shouldn't work with bar.org or 2000-2010 shouldn't work >> with foo.org. >> >> Any ideas how to do that? >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/15bd726b/attachment.html From diego.viola at gmail.com Tue Sep 22 22:23:27 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 23 Sep 2009 05:23:27 +0000 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909222222n4af8afbx97ee603cd3b97971@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <86a32abc0909222222n4af8afbx97ee603cd3b97971@mail.gmail.com> Message-ID: <86a32abc0909222223o2d2db1bdr930938ec4804e162@mail.gmail.com> Oh nvm I think I got it =D On Wed, Sep 23, 2009 at 5:22 AM, Diego Viola wrote: > I know I could create different domains on the directory but how do I tell > a user to belong to a specific domain? > > > On Wed, Sep 23, 2009 at 5:11 AM, Diego Viola wrote: > >> I want user 1000-1010 to belong to foo.org and 2000-2020 to belong to >> bar.org, and I want both of those domains to have their own >> dialplan/context. >> >> >> On Wed, Sep 23, 2009 at 4:57 AM, Diego Viola wrote: >> >>> Hello all, >>> >>> How do I configure multi tenant in FS? >>> >>> For example, I want some users to be able to register only with their own >>> domain. >>> >>> Ie: >>> >>> Users: 1000-1010 >>> Domain: foo.org >>> >>> Users: 2000-2010 >>> Domain: bar.org >>> >>> But 1000-1010 shouldn't work with bar.org or 2000-2010 shouldn't work >>> with foo.org. >>> >>> Any ideas how to do that? >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/411d3386/attachment.html From siniypin at gmail.com Tue Sep 22 22:25:41 2009 From: siniypin at gmail.com (RobertT) Date: Wed, 23 Sep 2009 09:25:41 +0400 Subject: [Freeswitch-users] mod_conference performance (Brian West) Message-ID: <2160023e0909222225i4c86180ct288a2a79b0bff1c9@mail.gmail.com> It was one big conference. Robert -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/3b7502f0/attachment.html From lakindia89 at gmail.com Tue Sep 22 22:33:09 2009 From: lakindia89 at gmail.com (lakshmanan ganapathy) Date: Wed, 23 Sep 2009 11:03:09 +0530 Subject: [Freeswitch-users] Mod_perl $session in not hangup In-Reply-To: <191c3a030909220739p66502dc3yc207f791bb158f8c@mail.gmail.com> References: <25530646.post@talk.nabble.com> <191c3a030909220739p66502dc3yc207f791bb158f8c@mail.gmail.com> Message-ID: <7d79b3930909222233p11504f17jeee7b4f923fafea9@mail.gmail.com> Thanks for your replay. I don't know what is latest trunk. Is it latest version? I'm using freeswitch 1.0.4. On Tue, Sep 22, 2009 at 8:09 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > The reason is you cannot complete the hangup until the script exits. > On the bright side, if you update to latest trunk it will probably work > more how you want it to > because a recent change will make this possible. > > > On Tue, Sep 22, 2009 at 4:30 AM, lakshmanan wrote: > >> >> Hi all, I've the following mod_perl program to execute when I call to an >> extension (say 777). >> I use twinkle as a soft phone, to make calls. >> >> #!/usr/bin/perl >> use strict; >> use freeswitch; >> our $session; >> $session->answer(); >> if($session->ready()) >> { >> my $uuid=$session->getVariable("uuid"); >> freeswitch::consoleLog("INFO","UUID is $uuid\n"); >> >> freeswitch::consoleLog("INFO","Session is answered\n"); >> >> >> $session->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav"); >> my $dtmf = $session->getDigits(4,"", 5000); >> freeswitch::consoleLog("INFO","I received $dtmf\n"); >> $session->hangup("NORMAL_CLEARING"); >> sleep(5); >> # Some other statements. >> } >> return 1; >> >> Everything is fine. >> After executing $session->hangup, I got NORMAL_CLEARING in my freeswitch >> console. But in my soft phone, still the channel is active for 5 seconds. >> The call got ended only after the 5 seconds sleep. >> >> But if I create my own session like >> my $session=new >> freeswitch::Session("user/1000"); >> and I say $session->hangup(), it got terminated. >> >> I wanted to know why there is such difference?? or am I wrong?? >> Please clarify me. >> >> >> -- >> View this message in context: >> http://www.nabble.com/Mod_perl-%24session-in-not-hangup-tp25530646p25530646.html >> Sent from the Freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/18625330/attachment-0001.html From brian at freeswitch.org Tue Sep 22 22:42:36 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Sep 2009 00:42:36 -0500 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909222222n4af8afbx97ee603cd3b97971@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <86a32abc0909222222n4af8afbx97ee603cd3b97971@mail.gmail.com> Message-ID: You don't have to think about it with proper DNS it all just magically happens. /b On Sep 23, 2009, at 12:22 AM, Diego Viola wrote: > I know I could create different domains on the directory but how do > I tell a user to belong to a specific domain? From diego.viola at gmail.com Tue Sep 22 23:03:46 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 23 Sep 2009 06:03:46 +0000 Subject: [Freeswitch-users] Multitenancy In-Reply-To: References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <86a32abc0909222222n4af8afbx97ee603cd3b97971@mail.gmail.com> Message-ID: <86a32abc0909222303u55291bfelc9d4265fc6fd3361@mail.gmail.com> Nice, so I just rename the "default" to foo.org and bar.org and I put the users I want inside them? On Wed, Sep 23, 2009 at 5:42 AM, Brian West wrote: > You don't have to think about it with proper DNS it all just magically > happens. > > /b > > On Sep 23, 2009, at 12:22 AM, Diego Viola wrote: > > > I know I could create different domains on the directory but how do > > I tell a user to belong to a specific domain? > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/2bc3b97d/attachment.html From diego.viola at gmail.com Tue Sep 22 23:04:02 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 23 Sep 2009 06:04:02 +0000 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909222303u55291bfelc9d4265fc6fd3361@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <86a32abc0909222222n4af8afbx97ee603cd3b97971@mail.gmail.com> <86a32abc0909222303u55291bfelc9d4265fc6fd3361@mail.gmail.com> Message-ID: <86a32abc0909222304r7874e006gec03e743909535a8@mail.gmail.com> s/rename/copy/ On Wed, Sep 23, 2009 at 6:03 AM, Diego Viola wrote: > Nice, so I just rename the "default" to foo.org and bar.org and I put the > users I want inside them? > > > On Wed, Sep 23, 2009 at 5:42 AM, Brian West wrote: > >> You don't have to think about it with proper DNS it all just magically >> happens. >> >> /b >> >> On Sep 23, 2009, at 12:22 AM, Diego Viola wrote: >> >> > I know I could create different domains on the directory but how do >> > I tell a user to belong to a specific domain? >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/3d2d6d6a/attachment.html From jason at jasonjgw.net Tue Sep 22 23:05:25 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 23 Sep 2009 16:05:25 +1000 Subject: [Freeswitch-users] Mod_perl $session in not hangup In-Reply-To: <7d79b3930909222233p11504f17jeee7b4f923fafea9@mail.gmail.com> References: <25530646.post@talk.nabble.com> <191c3a030909220739p66502dc3yc207f791bb158f8c@mail.gmail.com> <7d79b3930909222233p11504f17jeee7b4f923fafea9@mail.gmail.com> Message-ID: <20090923060525.GA8185@jdc.jasonjgw.net> lakshmanan ganapathy wrote: > Thanks for your replay. I don't know what is latest trunk. Is it latest > version? I'm using freeswitch 1.0.4. It's the latest version from the svn repository. Use svn checkout, then compile it as documented on the wiki. From craig at vastpark.com Tue Sep 22 22:55:55 2009 From: craig at vastpark.com (Craig Presti) Date: Wed, 23 Sep 2009 15:55:55 +1000 Subject: [Freeswitch-users] Conference with pin dialplan Message-ID: <014001ca3c12$858bccd0$90a36670$@com> Hi all, I've written a C# module for FS that creates conference dialplans on the fly. From my limited understanding the easiest way to do this is by writing XML to the directory: conf/dialplan/default/ - so this is the approach I've taken. After some suggestions in IRC to remove errant flags, I'm still having difficulty making the conference require pin entry. My generated XML looks like this: If I dial that extension I get music but no pin challenge, the next person that dials the same. If i hardcode in conference.conf.xml (default profile) a pin of: ...then it works as expected, and pin access is required. Is it possible I've overridden pin settings elsewhere in the FS config? Does ACL have any relationship to conference pin requirements? Thanks! Craig -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/c0cfcf46/attachment.html From brian at freeswitch.org Tue Sep 22 23:15:31 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Sep 2009 01:15:31 -0500 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909222303u55291bfelc9d4265fc6fd3361@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <86a32abc0909222222n4af8afbx97ee603cd3b97971@mail.gmail.com> <86a32abc0909222303u55291bfelc9d4265fc6fd3361@mail.gmail.com> Message-ID: <0BBC3ECC-D68C-4010-8EBC-AAA0A21235A7@freeswitch.org> make sure you look at default.xml and set the domain in the files for foo.org.xml and bar.org.xml /b On Sep 23, 2009, at 1:03 AM, Diego Viola wrote: > Nice, so I just rename the "default" to foo.org and bar.org and I > put the users I want inside them? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/74046bb9/attachment.html From tculjaga at gmail.com Tue Sep 22 23:35:24 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 23 Sep 2009 08:35:24 +0200 Subject: [Freeswitch-users] Can this be done in FreeSWITCH? In-Reply-To: References: <65d96fc80909220949v5de8fc54ia0f1932c0def74b2@mail.gmail.com> Message-ID: <65d96fc80909222335n38cfa195udbfb230c7fb22ece@mail.gmail.com> when i said inline ... i just meant to define some variables in your DP ... this is not a solution for you ... it is rather a proof of concept instead. you need to do a DB lookup (sqlite or mysql). T. On Wed, Sep 23, 2009 at 1:32 AM, Francis Vidal wrote: > Yes, this is the desired outcome. I was planning of using FreeSWITCH + > MySQL to do this. How do I do this "inline"? > > > On Wed, Sep 23, 2009 at 12:49 AM, Tihomir Culjaga wrote: > >> so, you say ... >> >> CallingParty => AS5300 >> >> A: aNum >> B: didNum >> >> >> AS5300 => PSTN >> >> A: 1 + didNum >> B: prefix (actually the PSTN subscriber's number) >> >> >> well, without a doubt... you can manipulate whatever number you want ... >> you just need to find the best way to do it. This depends of the number of >> DIDs you would like to host. You can do a DB lookup to retrieve the prefix / >> Subscriber Number... or you can do it inline in your dialplan. It really >> depends of how much you need to scale. >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/c92603e0/attachment-0001.html From jason at jasonjgw.net Tue Sep 22 23:46:45 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 23 Sep 2009 16:46:45 +1000 Subject: [Freeswitch-users] Conference with pin dialplan In-Reply-To: <014001ca3c12$858bccd0$90a36670$@com> References: <014001ca3c12$858bccd0$90a36670$@com> Message-ID: <20090923064645.GA12159@jdc.jasonjgw.net> Craig Presti wrote: > I've written a C# module for FS that creates conference dialplans on the > fly. From my limited understanding the easiest way to do this is by writing > XML to the directory: conf/dialplan/default/ - so this is the approach I've > taken. mod_curl might be easier if you have a Web server handy, or you could just write code that attaches to the socket interface, or gets invoked from the dial plan, and creates the conferences. See the wiki for further details. From mike at jerris.com Tue Sep 22 23:59:52 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 02:59:52 -0400 Subject: [Freeswitch-users] Attended transfer - no audio In-Reply-To: <698401620909101514h15214d47r78130a81e3a81c5d@mail.gmail.com> References: <698401620909101514h15214d47r78130a81e3a81c5d@mail.gmail.com> Message-ID: Did you ever resolve this issue? If not, please make sure you open a bug on jira.freeswitch.org with as much detail to reproduce this as possible. Mike On Sep 10, 2009, at 6:14 PM, Jan Kubr wrote: > Hi, > we have a Freeswitch server on a public IP and a few phones behind > NAT. The phones are configured to use STUN and can register and call > each other fine. > The problem is that after attended transfer (using the mechanism the > phones provide - REFER) is finished, the two parties can't hear each > other. This problem doesn't occur when the phones are in the same > subnet as Freeswitch. > > I know this isn't enough information to solve the problem, but do you > have any hints on how to debug this? Are there any specific Freeswitch > settings that could help us? > > Thanks, > Jan From sranil at gmail.com Wed Sep 23 00:30:47 2009 From: sranil at gmail.com (Anil Kumar S. R.) Date: Wed, 23 Sep 2009 13:00:47 +0530 Subject: [Freeswitch-users] Gateways in Freeswitch Message-ID: <1b2118200909230030k139218a2g879224bbb0c839cf@mail.gmail.com> Hi All, Can anybody please tell me what are the gateways in Freeswitch ? Thanks, -- Anil Kumar S. R. http://sranil.googlepages.com/ "The best way to succeed in this world is to act on the advice you give to others." -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/5203c26e/attachment.html From msc at freeswitch.org Wed Sep 23 00:46:21 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Sep 2009 00:46:21 -0700 Subject: [Freeswitch-users] Gateways in Freeswitch In-Reply-To: <1b2118200909230030k139218a2g879224bbb0c839cf@mail.gmail.com> References: <1b2118200909230030k139218a2g879224bbb0c839cf@mail.gmail.com> Message-ID: <87f2f3b90909230046v762d81aw3541ae25b33911b5@mail.gmail.com> On Wed, Sep 23, 2009 at 12:30 AM, Anil Kumar S. R. wrote: > Hi All, > > Can anybody please tell me what are the gateways in Freeswitch ? > A gateway is simply a means of doing an outbound registration to another server. Once the gateway is created you may send calls out on it. A typical use for a gateway is for you to have your FS box register to your SIP provider. Once you have your FS is registered to your provider you can then send and receive calls. More information and examples is available on the wiki: http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Dialing_out_via_Gateway -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/2ebf645d/attachment.html From tculjaga at gmail.com Wed Sep 23 00:49:02 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 23 Sep 2009 09:49:02 +0200 Subject: [Freeswitch-users] Gateways in Freeswitch In-Reply-To: <1b2118200909230030k139218a2g879224bbb0c839cf@mail.gmail.com> References: <1b2118200909230030k139218a2g879224bbb0c839cf@mail.gmail.com> Message-ID: <65d96fc80909230049i1a0b510ev88e2be6bb2ad4ca7@mail.gmail.com> endpoints that you are sending/receiving calls to/from .... It is useful to have a separate configuration (other than dialplan) when you need to specify credentials for GW to register somewhere, to specify domain, etc, etc ... T. On Wed, Sep 23, 2009 at 9:30 AM, Anil Kumar S. R. wrote: > Hi All, > > Can anybody please tell me what are the gateways in Freeswitch ? > > Thanks, > -- > Anil Kumar S. R. > http://sranil.googlepages.com/ > > "The best way to succeed in this world is to act on the advice you give to > others." > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/89eb21a4/attachment.html From mike at jerris.com Wed Sep 23 00:54:33 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 03:54:33 -0400 Subject: [Freeswitch-users] fax detection In-Reply-To: <65d96fc80909171525r72cbc381u6a4565269fe02512@mail.gmail.com> References: <65d96fc80909150845i5b287bc2w5a04537c836b1b48@mail.gmail.com> <87f2f3b90909171351y401b2782offc5522b9fc0681c@mail.gmail.com> <65d96fc80909171525r72cbc381u6a4565269fe02512@mail.gmail.com> Message-ID: <124A7A1E-A4F6-4229-91A7-B1CFF7A96E8E@jerris.com> You have access to the full sdp in channel vars, so you can condition on those with regex. Mike On Sep 17, 2009, at 6:25 PM, Tihomir Culjaga wrote: > Hi Michael, thanks for your response. > > i think it will be enough to check the call capability... we always > know the call is fax. We just need to apply the correct protocol :P > > > let's suppose you have 2 incoming calls: > SDP containing G711, gsm, T.38 caps > SDP containing G711, gsm caps > > the caps will be known on within INVITE message and FS can act > accordingly. > If there is no T.38 support within SDP, start fax application > (SpanDSP). > If there is T.38 suport within the SDP, route the call to some > predefined gateway meant for T.38 fax receiving. > In both cases, the fax should be received :P. > > So, any chance to route the call according to T.38 caps within SDP > message? > > > > Now, for sending faxes there is some challenge.... we still have 2 > calls: > > we start sending a fax inband (SpanDSP) as this is the only thing we > know... and: > if we receive 200 OK with SDP containing G711, anyCaps => continue > with InBand Fax > if we receive 200 OK with DSP containing anyCompressedCaps, T.38 => > drop the call without sending ACK to 200 OK ... and move the fax to > be sent into a different directory. > well, as i said, while receiving faxes will work 100%, sending is > tricky... but it might work. > > what do you guys think? > > T. > > > > > On Thu, Sep 17, 2009 at 10:51 PM, Michael Collins > wrote: > > > On Tue, Sep 15, 2009 at 8:45 AM, Tihomir Culjaga > wrote: > Hi, > > is there any way to route fax calls according to the call capability? > > I mean .. if the fax call supports T.38 i'd like to route it to a T. > 38 capable gateway. All other fax calls (meaning inband) should be > handled by FS/SpanDSP. > Of course, I know that every fax call starts as a voice call and > upon fax tone detection additional capabilities are being negotiated > (T.38 or G711). Can it be done in early media, before the call is > even answered? > > I don't claim to be an expert in all this, especially T.38, but if I > understand correctly, in both cases the call needs to be answered > first. I'm pretty sure that the sending fax machine won't start > emitting the 1100Hz tone until the receiving end answers. Also, with > T.38 doesn't the call have to come up and then T.38 gets negotiated? > (I don't know, I have only read about it.) > -MC > > > So, here the goal is to have a T.38 capable GW handling T.38 calls > while SpanDSP handling T.30... > > > Any chance to do that with FS? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/874ef2f6/attachment.html From mike at jerris.com Wed Sep 23 01:06:28 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 04:06:28 -0400 Subject: [Freeswitch-users] memory leak - outbound socket In-Reply-To: <4AB106B6.7090305@xpirio.com> References: <4AB0F65E.4060600@xpirio.com> <4AB106B6.7090305@xpirio.com> Message-ID: This should now be resolved in svn trunk. Mike On Sep 16, 2009, at 11:39 AM, Christian L?schenkohl wrote: > as a good fs user - of course i am :-) - i made a jira on this > MODAPP-336 to be precise > > i hope this helps to solve my problem > > br > > On 2009-09-16 17:05, Rupa Schomaker wrote: >> Either: >> >> 1) Provide a simple self-contained example that demonstrates the leak >> >> or >> >> 2) Run your application with FreeSWITCH under valgrind and provide >> the >> final output. To run freeswitch under valgrind: >> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs#Collection_Information_With_Valgrind_.28Linux.2FUnix.29 >> >> You should not have to run with high load to capture the behavior. >> Try with just 5 (in series) and then stop freeswitch. >> >> >> 2009/9/16 Christian L?schenkohl: >>> hello >>> >>> version : 1.0.4 std. tarball >>> >>> - the wiki example for php outbound socket connection leaks memory >>> without the async option >>> - the memory used is never given back >>> - async isn't that usefull for us - we want to query databases, >>> set variables and so on >>> no wait statements are possible >>> >>> >>> >>> >>> >>> <<<<---- no async !!!! >>> >>> >>> >>> the script is on the site >>> http://wiki.freeswitch.org/wiki/PHP_ESL >>> >>> ------------------------------- >>> >>> what can i do? >>> on our production server we use outbound socket connection and the >>> 4 gig of memory are >>> eaten up in less than a day >>> >>> br From mike at jerris.com Wed Sep 23 01:14:59 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 04:14:59 -0400 Subject: [Freeswitch-users] Build problems with Shoutcast module under Debian In-Reply-To: <20090918020726.GA3242@jdc.jasonjgw.net> References: <20090918020726.GA3242@jdc.jasonjgw.net> Message-ID: What issues are there with libtool 2 under debian? Libtool 2 issues that I am aware of were all sorted out quite some time ago. Mike On Sep 17, 2009, at 10:07 PM, Jason White wrote: > While trying to build FreeSWITCH rev. 14913, compilation failed with > the > following. > > the operating system is Debian Sid. Ogg development files are > installed, but > libogg.la does not exist anywhere. I'm still using libtool 1.5.26, > because the > build problems with FreeSWITCH and libtool 2 under Debian haven't been > resolved. From mike at jerris.com Wed Sep 23 01:20:07 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 04:20:07 -0400 Subject: [Freeswitch-users] FreeSWITCH 64bit compilation error (Solaris 10) In-Reply-To: <4ab3d091.9453f10a.0891.2fa4@mx.google.com> References: <4ab3d091.9453f10a.0891.2fa4@mx.google.com> Message-ID: <2C944C29-80D7-4F4F-ABD4-EC5202FD08B7@jerris.com> Try taking a list at the info here: http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Solaris You need to be passing any necessary cflags in on configure Mike On Sep 18, 2009, at 2:26 PM, email lists wrote: > Forwarding the issue below to see if anyone is familiar with this > issue, and/or what our next steps should be. > > Thanks, > Vladimir > > > Looks like a problem with a Makefile not honoring CFLAGS,etc. > Perhaps you can report this to the dev team. Other components built > fine but this damn spidermonkey is buggering. > > # file /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/ > libfreeswitch*.o | head > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > g711.o: ELF 64-bit LSB relocatable AMD64 Version 1 > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > getgateway.o: ELF 64-bit LSB relocatable AMD64 Version 1 [CMOV] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > igd_desc_parse.o: ELF 64-bit LSB relocatable AMD64 Version 1 > [CMOV] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > libteletone_detect.o: ELF 64-bit LSB relocatable AMD64 Version 1 > [SSE2 SSE AMD_3DNow CMOV FPU] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > libteletone_generate.o: ELF 64-bit LSB relocatable AMD64 Version 1 > [SSE2 SSE AMD_3DNow CMOV] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > minisoap.o: ELF 64-bit LSB relocatable AMD64 Version 1 > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > minissdpc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > miniupnpc.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE > AMD_3DNow] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > miniwget.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] > /builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libfreeswitch_la- > minixml.o: ELF 64-bit LSB relocatable AMD64 Version 1 [SSE] > > # file /opt/freeradius-client/sbin/radacct > /opt/freeradius-client/sbin/radacct: ELF 64-bit LSB executable > AMD64 Version 1 [SSE2 SSE FXSR AMD_3DNow CMOV FPU], dynamically > linked, not stripped > > Build output: > Making all in nua > LTCOMPILE nua.lo > LTCOMPILE nua_common.lo > LTCOMPILE nua_stack.lo > LTCOMPILE nua_server.lo > LTCOMPILE nua_client.lo > LTCOMPILE nua_extension.lo > LTCOMPILE nua_dialog.lo > LTCOMPILE outbound.lo > LTCOMPILE nua_params.lo > LTCOMPILE nua_register.lo > LTCOMPILE nua_registrar.lo > LTCOMPILE nua_session.lo > LTCOMPILE nua_options.lo > LTCOMPILE nua_message.lo > LTCOMPILE nua_publish.lo > LTCOMPILE nua_subnotref.lo > LTCOMPILE nua_notifier.lo > LTCOMPILE nua_event_server.lo > LTCOMPILE nua_tag.lo > LTCOMPILE nua_tag_ref.lo > LINK libnua.la > LINK libsofia-sip-ua.la > libtool: link: warning: `-version-info/-version-number' is ignored > for convenience libraries > Making all in packages > Creating mod_sofia_la-mod_sofia.lo > mkdir .libs > Compiling mod_sofia.c ... > "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/ > libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable > bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) > Creating mod_sofia_la-sofia.lo > Compiling sofia.c ... > "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/ > libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable > bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) > "sofia.c", line 3522: warning: enum type mismatch: arg #2 > (E_ENUM_TYPE_MISMATCH_ARG) > Creating mod_sofia_la-sofia_glue.lo > Compiling sofia_glue.c ... > "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/ > libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable > bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) > Creating mod_sofia_la-sofia_presence.lo > Compiling sofia_presence.c ... > "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/ > libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable > bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) > Creating mod_sofia_la-sofia_reg.lo > Compiling sofia_reg.c ... > "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/ > libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable > bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) > Creating mod_sofia_la-sofia_sla.lo > Compiling sofia_sla.c ... > "/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/sofia-sip/ > libsofia-sip-ua/nua/nua_params.h", line 221: warning: nonportable > bit-field type (E_NONPORTABLE_BIT_FIELD_TYPE) > Creating mod_sofia.la > > making all mod_speex > Compiling mod_speex.c... > mkdir .libs > Compiling mod_speex.c ... > Creating mod_speex.so... > > making all mod_spidermonkey > cd config; /usr/sfw/bin/gmake -j1 export > ld: fatal: file now.o: wrong ELF class: ELFCLASS64 > ld: fatal: File processing errors. No output written to now > gmake[7]: *** [now] Error 1 > gmake[6]: *** [export] Error 2 > gmake[5]: *** [/builds/work/freeswitch-1.0.4/freeswitch-1.0.4/libs/ > js/libjs.la] Error 2 > gmake[4]: *** [all] Error 1 > gmake[3]: *** [mod_spidermonkey-all] Error 1 > gmake[2]: *** [all-recursive] Error 1 > Making all in build > +-------- FreeSWITCH Build Complete -----------+ > + FreeSWITCH has been successfully built. + > + Install by running: + > + + > + /usr/sfw/bin/gmake install + > +----------------------------------------------+ > gmake[1]: *** [all-recursive] Error 1 > gmake: *** [all] Error 2 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/6125193b/attachment-0001.html From mike at jerris.com Wed Sep 23 01:28:02 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 04:28:02 -0400 Subject: [Freeswitch-users] A real puzzler for you guys... (perpetual registration causes denial of service.) In-Reply-To: <20090919064901.GA2129@jdc.jasonjgw.net> References: <1BA91B48-688F-4FF1-8078-F1810185D4AB@ken-ton.com> <8A58CD3E-F0C3-4BAD-8DB9-3C9D3A22A70F@ken-ton.com> <191c3a030909140617v396063e7n86cd8fdb39d75fa9@mail.gmail.com> <0E10F348-B983-48F9-87DB-19BE52CEFD50@ken-ton.com> <20090917040742.GA24786@jdc.jasonjgw.net> <9AC967E2-2807-4FB7-BA8E-F5C128BCDE76@freeswitch.org> <9C617DD2-DDFD-4160-81EE-B0FD5E2892EA@ken-ton.com> <20090919064901.GA2129@jdc.jasonjgw.net> Message-ID: <606F5023-5C70-4C15-AF56-81F9104475B2@jerris.com> This is a well known and documented problem already fixed in upstream kernels. There was quite a bit of discussions about tests that came out that showed this issue. Opening another bug on it is not likely to help. Mike On Sep 19, 2009, at 2:49 AM, Jason White wrote: > Karl Vesterling wrote: >> No penguin is perfect... >> There's issues w/ 2.6.X - 2.6.27.X with respect to timing for things >> like packet shaping, which is a requirement for me. > > Two suggestions: > > 1. Your distribution's bug tracker. > > 2. http://ltp.sourceforge.net/ > (If they get test coverage of the relevant interfaces there will be > quicker > detection of problems and, we hope, prevention of regressions.) > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From jason at jasonjgw.net Wed Sep 23 01:36:24 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 23 Sep 2009 18:36:24 +1000 Subject: [Freeswitch-users] Debian, libtool2 and mod_portaudio In-Reply-To: References: <20090918020726.GA3242@jdc.jasonjgw.net> Message-ID: <20090923083624.GA25398@jdc.jasonjgw.net> Michael Jerris wrote: > What issues are there with libtool 2 under debian? Libtool 2 issues > that I am aware of were all sorted out quite some time ago. With libtool2, mod_portaudio fails to link to the Alsa sound library, hence fails to load due to unresolved symbols. I can't test this just now, due to the Ogg vorbis issue mentioned recently in this thread. From mike at jerris.com Wed Sep 23 02:03:03 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 05:03:03 -0400 Subject: [Freeswitch-users] Debian, libtool2 and mod_portaudio In-Reply-To: <20090923083624.GA25398@jdc.jasonjgw.net> References: <20090918020726.GA3242@jdc.jasonjgw.net> <20090923083624.GA25398@jdc.jasonjgw.net> Message-ID: <4B0246BA-C9D8-4C7A-A28B-91683E952FB0@jerris.com> if someone can contact we later in the day offlist with credentials for a box I can try to fix these issues. Mike On Sep 23, 2009, at 4:36 AM, Jason White wrote: > Michael Jerris wrote: >> What issues are there with libtool 2 under debian? Libtool 2 issues >> that I am aware of were all sorted out quite some time ago. > > With libtool2, mod_portaudio fails to link to the Alsa sound > library, hence > fails to load due to unresolved symbols. > > I can't test this just now, due to the Ogg vorbis issue mentioned > recently in > this thread. > From mike at jerris.com Wed Sep 23 02:08:37 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 05:08:37 -0400 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> Message-ID: <23B64F60-4BDC-4F98-8CA8-02D21F4828D7@jerris.com> A couple people have taken on major work on packages for ubuntu. Most of that work will translate directly back to debian, we should just need people to do testing of debian pacakges once their work is done. Also we had one more person step up to help with spec file work. I still need help with coders who can assist on language portability, with both the mod_say_* modules for all kinds of languages, translating the phrase files, and once those are done, looking for sponsors to get other language prompts recorded. Mike On Sep 15, 2009, at 10:58 AM, Brian West wrote: > The friday meetings are where we all collaborate on these group > efforts and discuss project direction, goals and areas where people > can help out more. > > Did we ever find someone to officially take over the Debian packages? > > /b > > > > On Sep 15, 2009, at 9:56 AM, Michael Gende wrote: > >> Answered my own question. Will email Diego off-list. >> >> On Tue, Sep 15, 2009 at 9:50 AM, Michael Gende >> wrote: >> Hey Demuel, et al, >> >> I agree that without coordination, there will be duplication of >> effort, etc. >> >> How does one "find the line to get in" to help with, in my case, >> documentation? I don't want to, with the best of intentions, simply >> create confusion. >> >> I'm new, so if I've simply not yet RTFM that answers my question >> above, some kind person might nudge me in the right direction. >> >> Back to registering with my SIP provider... >> >> Regards, >> >> Mike G. >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/2b0b2b5f/attachment.html From mike at jerris.com Wed Sep 23 02:10:37 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 05:10:37 -0400 Subject: [Freeswitch-users] Recording inbound call including DTMF - possible ? In-Reply-To: <25753ef00909130727geaa6002k80fbca40611d603@mail.gmail.com> References: <25753ef00909130142w68b12c5pf44f4bc40f2376a@mail.gmail.com> <25753ef00909130727geaa6002k80fbca40611d603@mail.gmail.com> Message-ID: Are you using freeswitch to detect the inband dtmf or are you getting both inband and some other method (rfc 2833?) of dtmf as well? Mike On Sep 13, 2009, at 10:27 AM, Morten Henckel wrote: > Hi > > I need to measure DTM digits duration and interdigit delay for > various phones in a two stage dialing scenario. I.e Phone dials DID > and after answer then the second number > > My set-up is: > > Phone->PSTN network->DID(inband DTMF) ->FS > > I ha ve FS to answer the call and record the call - all this is fine. > > However when i analyse the rdecording the Digits are being cut off > down to 10 msec "bursts" - I trust its FS that cust the DTMF in > order to avoid further propogation inband to second leg of the call. > > Is theer a way to avoid this ? I.e record the inbound call without > DTMF processing ? > > Thx > > Morten > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/dbb372ce/attachment.html From mike at jerris.com Wed Sep 23 02:11:44 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 05:11:44 -0400 Subject: [Freeswitch-users] mod_nibblebill In-Reply-To: References: Message-ID: If it is no where in the code I would assume it is not implemented, try catching up with pyite on irc to confirm. Mike On Sep 20, 2009, at 12:03 AM, Jo?o Mesquita wrote: > Guys, I have been testing mod_nibblebill lately and there are 2 > params that I could not make work. > > > > > > Looking at code, I could not find a single line that would actually > test those. > > Is this confirmed to be implemented? If not, this should be removed > from the configs so it won't get ppl lured. From mike at jerris.com Wed Sep 23 02:13:49 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 05:13:49 -0400 Subject: [Freeswitch-users] Vestec Speech Recognition Integration ? In-Reply-To: <98a86adf0909161421g5b51c434ofeb01f409b4a85f6@mail.gmail.com> References: <98a86adf0909161421g5b51c434ofeb01f409b4a85f6@mail.gmail.com> Message-ID: Its probably trivial to add MRCP for them now with the unimrcp lib. I would suggest making them aware of it. At this point that is the best way to go so we don't have to support a bunch of custom interfaces. Mike On Sep 16, 2009, at 5:21 PM, Gerry Hull wrote: > Has anyone integrated Vestec Speech Recognition with FreeSwitch? > It's $99/port... http://www.vestec.ca/ > > They have a C/C++ api, looks pretty simple. Alas, no MRCP until > 2010. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/a107c01e/attachment.html From mike at jerris.com Wed Sep 23 01:49:40 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 04:49:40 -0400 Subject: [Freeswitch-users] Compile error In-Reply-To: <881903.13614.qm@web53103.mail.re2.yahoo.com> References: <1253162070.3583.14.camel@localhost.localdomain> <20090917054213.GH30343@base.carmickle.com> <19FE43B3-7C34-49C5-8E10-DA66B66197BA@freeswitch.org> <881903.13614.qm@web53103.mail.re2.yahoo.com> Message-ID: <01D1DE01-71EE-4D17-961C-B5911CCBE814@jerris.com> Please catch up on irc to discuss this real time, this shouldn't be happening and bkw or I likely will need remote access to your box to figure out why it is doing this. Mike On Sep 21, 2009, at 1:06 PM, Luis Manuel Zuccolo wrote: > I' ve get the same error with a fresh tree > > Thanks in advance > > De: Brian West > Para: freeswitch-users at lists.freeswitch.org > Enviado: jueves 17 de septiembre de 2009, 10:12:36 > Asunto: Re: [Freeswitch-users] Compile error > > NO you must not. The issue has been fixed in svn already please > start with a fresh tree. > > /b > PS: end users should NEVER have to reswig. > > On Sep 17, 2009, at 12:42 AM, Frank Carmickle wrote: > >> On Thu, Sep 17, Luis M. Zuccolo wrote: >>> Hi: >>> >>> Since svn version 13523 to current I get this error: >>> >>> make[5]: swig: Command not found >> >> You must install swig. If your on debian apt-get install swig. If >> your not see http://www.swig.org/ >> >> HTH >> --FC > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/603f37fa/attachment-0001.html From jason at jasonjgw.net Wed Sep 23 02:23:49 2009 From: jason at jasonjgw.net (Jason White) Date: Wed, 23 Sep 2009 19:23:49 +1000 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <23B64F60-4BDC-4F98-8CA8-02D21F4828D7@jerris.com> References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> <23B64F60-4BDC-4F98-8CA8-02D21F4828D7@jerris.com> Message-ID: <20090923092348.GA1769@jdc.jasonjgw.net> Michael Jerris wrote: > A couple people have taken on major work on packages for ubuntu. > Most of that work will translate directly back to debian, we should > just need people to do testing of debian pacakges once their work is > done. I'm volunteering. From shaheryarkh at googlemail.com Wed Sep 23 03:55:58 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 23 Sep 2009 16:55:58 +0600 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: <20090923092348.GA1769@jdc.jasonjgw.net> References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> <23B64F60-4BDC-4F98-8CA8-02D21F4828D7@jerris.com> <20090923092348.GA1769@jdc.jasonjgw.net> Message-ID: I am not a debian fan nor expert in it but if you guys just want me do to dpkg -i freeswitch*.deb on my test box and report for success or failure then you can count me in as well. Thank you. On Wed, Sep 23, 2009 at 3:23 PM, Jason White wrote: > Michael Jerris wrote: > > A couple people have taken on major work on packages for ubuntu. > > Most of that work will translate directly back to debian, we should > > just need people to do testing of debian pacakges once their work is > > done. > > I'm volunteering. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- ________________________________________________________ | | | FATAL ERROR --- O X | |_______________________________________________________| | You have moved the mouse. | | Windows must be restarted for the changes to take effect. | | | ####################################/ Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/b1f56d28/attachment.html From demuel at thephinix.org Wed Sep 23 04:22:49 2009 From: demuel at thephinix.org (demuel at thephinix.org) Date: Wed, 23 Sep 2009 12:22:49 +0100 Subject: [Freeswitch-users] CALL FOR VOLUNTEERS: Assisting With FreeSWITCH Subprojects In-Reply-To: References: <87f2f3b90909140933t3747689csb676383a6419657a@mail.gmail.com> <86a32abc0909141432q4558af81ved54bb6991376cbc@mail.gmail.com> <0c3b6c3de1ef0f43b24cbd6a47d73cb6.squirrel@mail.thephinix.org> <23B64F60-4BDC-4F98-8CA8-02D21F4828D7@jerris.com> <20090923092348.GA1769@jdc.jasonjgw.net> Message-ID: <68746984bfea209078804b76a52a85fd.squirrel@www.thephinix.org> This is one of the reasons I hate to use debian aka lesbian libyan linux distribution. They keep changing their packaging system. If you have a lenny with you, doing an "apt-get" or "dpkg" will give you a whole lot of package dependencies and even introduced a new one ( can't remember the name of it) as well as it's not easy to deal with those. However, this can have a shortcut if the package installing manager will automatically fetch the dependent package thereby giving you a straight installation. nohup, engrxyz > I am not a debian fan nor expert in it but if you guys just want me do to > > dpkg -i freeswitch*.deb > > on my test box and report for success or failure then you can count me in as > well. > > Thank you. > > > On Wed, Sep 23, 2009 at 3:23 PM, Jason White wrote: > >> Michael Jerris wrote: >> > A couple people have taken on major work on packages for ubuntu. >> > Most of that work will translate directly back to debian, we should >> > just need people to do testing of debian pacakges once their work is >> > done. >> >> I'm volunteering. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > ________________________________________________________ > | > | > | FATAL ERROR --- > O X | > |_______________________________________________________| > | You have moved the mouse. > | > | Windows must be restarted for the changes to take effect. | > | > | > ####################################/ > > > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Prometheus001 at gmx.net Wed Sep 23 04:28:38 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 23 Sep 2009 13:28:38 +0200 Subject: [Freeswitch-users] Hangup: Always the same Q.850 cause code In-Reply-To: <191c3a030909180746t326e06aakef4e26a0945de195@mail.gmail.com> References: <4AB33C76.5050702@gmx.net> <191c3a030909180746t326e06aakef4e26a0945de195@mail.gmail.com> Message-ID: <4ABA0666.8090102@gmx.net> Hello Anthony, I did further testing on a second machine and found out the following: After The called party receives a "NO_ANSWER" and the calling party receives a "NORMAL_CLEARING" See the logs: Best regards Peter Logs: ============================ To called party: U 82.xxx.9xx.163:5080 -> 82.xxx.9xx.165:5060 CANCEL sip:0xxxxxxxxx299 at 21x.xx.xx.189:3273;line=fihb87zs SIP/2.0. Via: SIP/2.0/UDP 82.xxx.9xx.163:5080;rport;branch=z9hG4bKg5SZ7829tHDae. Route: . Max-Forwards: 68. From: "0xxxxxxxxx298" ;tag=1mFgvS7t9Krtj. To: . Call-ID: 9aab911f-22ce-122d-8686-001517956764. CSeq: 120732503 CANCEL. Reason: Q.850;cause=19;text="NO_ANSWER". Content-Length: 0. To calling party: U 82.xxx.9xx.163:5062 -> 82.xxx.9xx.165:5060 SIP/2.0 480 Temporarily Unavailable. Via: SIP/2.0/UDP 82.xxx.9xx.165;branch=z9hG4bKc08d.b2a0b296.0. Via: SIP/2.0/UDP 21x.xx.xx.189:2048;received=21x.xx.xx.189;branch=z9hG4bK-dnhr44fkakhd;rport=2048. From: "0xxxxxxxxx298" ;tag=nvxy9h3rsk. To: ;tag=5y6B9FS9ZeUZB. Call-ID: 3c49ea1f4563-8c3hia75cxuh. CSeq: 2 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14741M. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, refer. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. 2009-09-23 12:51:37.008546 [NOTICE] switch_ivr_originate.c:2025 Hangup sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 [CS_CONSUME_MEDIA] [NO_ANSWER] 2009-09-23 12:51:37.008546 [DEBUG] switch_channel.c:1715 Send signal sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 [KILL] 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 [BREAK] 2009-09-23 12:51:37.008546 [DEBUG] switch_ivr_originate.c:2169 Originate Resulted in Error Cause: 19 [NO_ANSWER] 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:398 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) Running State Change CS_HANGUP 2009-09-23 12:51:37.008546 [INFO] mod_dptools.c:2098 Originate Failed. Cause: NO_ANSWER 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:434 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State HANGUP EXECUTE sofia/internal/0xxxxxxxxx298 at mydomain.de set(sip_ignore_remote_cause=true) 2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:338 Channel sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 hanging up, cause: NO_ANSWER 2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:386 Sending CANCEL to sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:46 sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 Standard HANGUP, cause: NO_ANSWER 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:434 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State HANGUP going to sleep 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:479 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State Change CS_HANGUP -> CS_REPORTING 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 [BREAK] 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:398 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) Running State Change CS_REPORTING 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:616 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State REPORTING 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:53 sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 Standard REPORTING, cause: NO_ANSWER 2009-09-23 12:51:37.008546 [DEBUG] mod_dptools.c:748 sofia/internal/0xxxxxxxxx298 at mydomain.de SET [sip_ignore_remote_cause]=[true] 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:616 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State REPORTING going to sleep 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:411 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State Change CS_REPORTING -> CS_DESTROY 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 [BREAK] 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:1068 Session 129 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) Locked, Waiting on external entities 2009-09-23 12:51:37.008546 [NOTICE] switch_core_session.c:1086 Session 129 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) Ended 2009-09-23 12:51:37.008546 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 [CS_DESTROY] EXECUTE sofia/internal/0xxxxxxxxx298 at mydomain.de hangup() 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:559 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) Running State Change CS_DESTROY 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:568 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State DESTROY 2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:255 sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 SOFIA DESTROY 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:60 sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 Standard DESTROY 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:568 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State DESTROY going to sleep 2009-09-23 12:51:37.008546 [NOTICE] mod_dptools.c:633 Hangup sofia/internal/0xxxxxxxxx298 at mydomain.de [CS_EXECUTE] [NORMAL_CLEARING] 2009-09-23 12:51:37.008546 [DEBUG] switch_channel.c:1715 Send signal sofia/internal/0xxxxxxxxx298 at mydomain.de [KILL] 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/0xxxxxxxxx298 at mydomain.de [BREAK] 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/0xxxxxxxxx298 at mydomain.de) State EXECUTE going to sleep 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/0xxxxxxxxx298 at mydomain.de) Running State Change CS_HANGUP 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/0xxxxxxxxx298 at mydomain.de) State HANGUP 2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:338 Channel sofia/internal/0xxxxxxxxx298 at mydomain.de hanging up, cause: NORMAL_CLEARING 2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:397 Responding to INVITE with: 480 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:46 sofia/internal/0xxxxxxxxx298 at mydomain.de Standard HANGUP, cause: NORMAL_CLEARING 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/0xxxxxxxxx298 at mydomain.de) State HANGUP going to sleep 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:479 (sofia/internal/0xxxxxxxxx298 at mydomain.de) State Change CS_HANGUP -> CS_REPORTING 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/0xxxxxxxxx298 at mydomain.de [BREAK] 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/0xxxxxxxxx298 at mydomain.de) Running State Change CS_REPORTING 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:616 (sofia/internal/0xxxxxxxxx298 at mydomain.de) State REPORTING 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:53 sofia/internal/0xxxxxxxxx298 at mydomain.de Standard REPORTING, cause: NORMAL_CLEARING 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:616 (sofia/internal/0xxxxxxxxx298 at mydomain.de) State REPORTING going to sleep 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:411 (sofia/internal/0xxxxxxxxx298 at mydomain.de) State Change CS_REPORTING -> CS_DESTROY 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/0xxxxxxxxx298 at mydomain.de [BREAK] 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:1068 Session 128 (sofia/internal/0xxxxxxxxx298 at mydomain.de) Locked, Waiting on external entities 2009-09-23 12:51:37.008546 [NOTICE] switch_core_session.c:1086 Session 128 (sofia/internal/0xxxxxxxxx298 at mydomain.de) Ended 2009-09-23 12:51:37.008546 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/0xxxxxxxxx298 at mydomain.de [CS_DESTROY] 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:559 (sofia/internal/0xxxxxxxxx298 at mydomain.de) Running State Change CS_DESTROY 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:568 (sofia/internal/0xxxxxxxxx298 at mydomain.de) State DESTROY 2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:255 sofia/internal/0xxxxxxxxx298 at mydomain.de SOFIA DESTROY 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:60 sofia/internal/0xxxxxxxxx298 at mydomain.de Standard DESTROY 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:568 (sofia/internal/0xxxxxxxxx298 at mydomain.de) State DESTROY going to sleep Anthony Minessale schrieb: > Here is what I get when I test it. > you may want to look at your console for the blue Hangup lines and > confirm it's your call to hangup > > > > > > > > > > send 644 bytes to udp/[72.128.89.126]:42988 at 14:34:29.043915: > > ------------------------------------------------------------------------ > SIP/2.0 480 Temporarily Unavailable > Via: SIP/2.0/UDP > 10.0.1.8:50606;branch=z9hG4bK-d8754z-6077f64f00ed157e-1---d8754z-;rport=42988;received=72.128.89.126 > From: "tony" >;tag=352a2a46 > To: "7016" >;tag=HUye00UQZKySQ > Call-ID: N2Y1MWYwZjA2YzJlY2ZhY2VjYzRhNDZmMzczYWMwN2Q. > CSeq: 1 INVITE > User-Agent: The Guy In IRC IS WRONG > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Reason: Q.850;cause=19;text="NO_ANSWER" > Content-Length: 0 > > > > > > > > > > > send 630 bytes to udp/[72.128.89.126]:42988 at 14:35:31.286436: > > ------------------------------------------------------------------------ > SIP/2.0 486 Busy Here > Via: SIP/2.0/UDP > 10.0.1.8:50606;branch=z9hG4bK-d8754z-223ae00e1829097e-1---d8754z-;rport=42988;received=72.128.89.126 > From: "tony" >;tag=aa3b2b1d > To: "7016" >;tag=j4Q71UcUvvmcK > Call-ID: NDcyNmQyYjY5YWQwOTI3MjZiZWFlZDQyNDIyZjZlMDA. > CSeq: 1 INVITE > User-Agent: The Guy In IRC IS WRONG > Accept: application/sdp > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, refer > Reason: Q.850;cause=17;text="USER_BUSY" > Content-Length: 0 > > > > On Fri, Sep 18, 2009 at 2:53 AM, Peter P GMX > wrote: > > Hello , > > I try to hangup aa call with a certain cause code. > > If the user dials a number which we currently do not serve we send > > > which gives a > "SIP/2.0 480 Temporarily Unavailable." Message , which is fine. > > For the target number being busy or having another state, we use this. > data="sip_ignore_remote_cause=true"/> > > which gives a > "SIP/2.0 486 Busy Here."" Message , which is fine in case of Busy. > > However in both cases the SIP mssage has the following cause code: > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > which can lead to problems when forwarding to a PSTN Gateway. > > How can we achieve, that the cause code is in sync with the deiivered > message? > > We are on Trunk 14741M. > > Best regards > Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From peder at networkoblivion.com Wed Sep 23 05:24:56 2009 From: peder at networkoblivion.com (Peder) Date: Wed, 23 Sep 2009 07:24:56 -0500 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <249BB60C-79F6-4A3A-8E3B-0E821D4D9046@freeswitch.org> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <249BB60C-79F6-4A3A-8E3B-0E821D4D9046@freeswitch.org> Message-ID: <0f3c01ca3c48$dc802d20$95808760$@com> So if you do this, how do you call between contexts? Say you have 100 tenants on one box each with their own domain and they are all 4 digit for local dialing. If they call a 10 digit number like they are calling outbound and it is another tenant on the same box, they don't want to go out and back in, they just want to be bridged over to the other context. I imagine you need to create a file for each context and add every other context and did to it to check if they are on the same box, otherwise dial outbound to the PSTN. That doesn't scale very well though since if there are 100 tenants, and you add another, you need to modify the other 100 contexts to add the new DIDs. Or is there some built in way to do this easily? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West Sent: Wednesday, September 23, 2009 12:18 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Multitenancy Then setup two domains in your directory and setup proper DNS its really just that simple. /b On Sep 23, 2009, at 12:11 AM, Diego Viola wrote: I want user 1000-1010 to belong to foo.org and 2000-2020 to belong to bar.org , and I want both of those domains to have their own dialplan/context. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/9a9a743e/attachment.html From ruda at ruda.com.br Wed Sep 23 05:33:59 2009 From: ruda at ruda.com.br (=?ISO-8859-1?Q?Rud=E1_Cunha?=) Date: Wed, 23 Sep 2009 09:33:59 -0300 Subject: [Freeswitch-users] SUBSCRIBE and NOTIFY In-Reply-To: <87f2f3b90909221442j59db5f6cr6bb8c596036d73ef@mail.gmail.com> References: <7600794b0909221212x73e23d49va08eb85c355f6c9f@mail.gmail.com> <7600794b0909221220y6a43357fs75caeec9c205e0f5@mail.gmail.com> <87f2f3b90909221442j59db5f6cr6bb8c596036d73ef@mail.gmail.com> Message-ID: <7600794b0909230533i3b4815a1x528591ad892e46ce@mail.gmail.com> I upgraded to version 1.0.trunk. And still with the problem. I am using the soft phone X-Lite. I set it (1000) and I connect. Once connected to create the User 1001. Ai in another X-Lite I connect with the (1001) and create the User 1000. Ai closing the two X-Lite and open the 1000 (it appears that the 1001 ta offline). Then open the 1001 (it says that the 1000 ta connected). But in the 1000 X-Lite does not appear that the 1001 ta connected. 2009/9/22 Michael Collins > > > On Tue, Sep 22, 2009 at 12:20 PM, Rud? Cunha wrote: > >> I'm having to configure FreeSWITCH. >> >> Download version 1.0.4 and I am accessing with the users 1000 and 1001. >> >> I register, make the connection. But I'm trying to see to see who is >> connected (SUBSCRIBE and NOTIFY). But sometimes you work, sometimes you do I >> connect the other User does not receive the information that I connected (I >> sent the (SUBSCRIBE and NOTIFY)). That is, sometimes you and I'm connected >> to another user I'm not, sometimes you work. >> > > What are you using to see who is connected? > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/aac52f1d/attachment.html From a.afzali2003 at gmail.com Wed Sep 23 05:37:57 2009 From: a.afzali2003 at gmail.com (afshin afzali) Date: Wed, 23 Sep 2009 16:07:57 +0330 Subject: [Freeswitch-users] Using Sofia-SIP with ICE NAT Traversal Mechanism Message-ID: Hi, I know that FreeSWITCH uses libdingaling to talk to Jingle call parties. Also I know that Jingle Protocol uses ICE protocol to traverse NAT devices. So : does libdingaling use an open library such as libnice for ICE? Is it possible to use the ICE implementation in Sofia-SIP endpoint? If not, how could I integrate an open ICE library in Sofia-SIP? Regards, -- afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/b5d54941/attachment.html From shaheryarkh at googlemail.com Wed Sep 23 05:51:28 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 23 Sep 2009 18:51:28 +0600 Subject: [Freeswitch-users] Using Sofia-SIP with ICE NAT Traversal Mechanism In-Reply-To: References: Message-ID: No mod_dingaling does not use LibNICE. However, i have plans to integrate NICE with Sofia in mod_msn project, which is at the moment moving with very slow pace due to some trouble in reverse engineering MSNP-18 protocol (used in Windows Live Messenger 2009). Thank you. On Wed, Sep 23, 2009 at 6:37 PM, afshin afzali wrote: > Hi, > > I know that FreeSWITCH uses libdingaling to talk to Jingle call parties. > Also I know that Jingle Protocol uses ICE protocol to traverse NAT devices. > So : > does libdingaling use an open library such as libnice for ICE? > Is it possible to use the ICE implementation in Sofia-SIP endpoint? > If not, how could I integrate an open ICE library in Sofia-SIP? > > Regards, > -- afshin > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ________________________________________________________ | | | FATAL ERROR --- O X | |_______________________________________________________| | You have moved the mouse. | | Windows must be restarted for the changes to take effect. | | | ####################################/ Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/533cd530/attachment.html From a.afzali2003 at gmail.com Wed Sep 23 06:17:10 2009 From: a.afzali2003 at gmail.com (afshin afzali) Date: Wed, 23 Sep 2009 16:47:10 +0330 Subject: [Freeswitch-users] Using Sofia-SIP with ICE NAT Traversal Mechanism In-Reply-To: References: Message-ID: Don't you think is better to integrate LibNICE to FreeSWITCH's RTP stack than a module such as mod_msn / mod_dingaling ? -- afshin On Wed, Sep 23, 2009 at 4:21 PM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > No mod_dingaling does not use LibNICE. However, i have plans to integrate > NICE with Sofia in mod_msn project, which is at the moment moving with very > slow pace due to some trouble in reverse engineering MSNP-18 protocol (used > in Windows Live Messenger 2009). > > Thank you. > > > On Wed, Sep 23, 2009 at 6:37 PM, afshin afzali wrote: > >> Hi, >> >> I know that FreeSWITCH uses libdingaling to talk to Jingle call parties. >> Also I know that Jingle Protocol uses ICE protocol to traverse NAT devices. >> So : >> does libdingaling use an open library such as libnice for ICE? >> Is it possible to use the ICE implementation in Sofia-SIP endpoint? >> If not, how could I integrate an open ICE library in Sofia-SIP? >> >> Regards, >> -- afshin >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > ________________________________________________________ > | > | > | FATAL ERROR --- > O X | > |_______________________________________________________| > | You have moved the mouse. > | > | Windows must be restarted for the changes to take effect. | > | > | > ####################################/ > > > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/dde045d6/attachment-0001.html From shaheryarkh at googlemail.com Wed Sep 23 06:42:44 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 23 Sep 2009 19:42:44 +0600 Subject: [Freeswitch-users] Using Sofia-SIP with ICE NAT Traversal Mechanism In-Reply-To: References: Message-ID: Yup, that's a good idea but not in my project list right now. Thank you. On Wed, Sep 23, 2009 at 7:17 PM, afshin afzali wrote: > Don't you think is better to integrate LibNICE to FreeSWITCH's RTP stack > than a module such as mod_msn / mod_dingaling ? > > -- afshin > > On Wed, Sep 23, 2009 at 4:21 PM, Muhammad Shahzad < > shaheryarkh at googlemail.com> wrote: > >> No mod_dingaling does not use LibNICE. However, i have plans to integrate >> NICE with Sofia in mod_msn project, which is at the moment moving with very >> slow pace due to some trouble in reverse engineering MSNP-18 protocol (used >> in Windows Live Messenger 2009). >> >> Thank you. >> >> >> On Wed, Sep 23, 2009 at 6:37 PM, afshin afzali wrote: >> >>> Hi, >>> >>> I know that FreeSWITCH uses libdingaling to talk to Jingle call parties. >>> Also I know that Jingle Protocol uses ICE protocol to traverse NAT devices. >>> So : >>> does libdingaling use an open library such as libnice for ICE? >>> Is it possible to use the ICE implementation in Sofia-SIP endpoint? >>> If not, how could I integrate an open ICE library in Sofia-SIP? >>> >>> Regards, >>> -- afshin >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> ________________________________________________________ >> | >> | >> | FATAL ERROR >> --- O X | >> |_______________________________________________________| >> | You have moved the mouse. >> | >> | Windows must be restarted for the changes to take effect. | >> | >> | >> ####################################/ >> >> >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ________________________________________________________ | | | FATAL ERROR --- O X | |_______________________________________________________| | You have moved the mouse. | | Windows must be restarted for the changes to take effect. | | | ####################################/ Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/24b98bab/attachment.html From Prometheus001 at gmx.net Wed Sep 23 06:44:27 2009 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 23 Sep 2009 15:44:27 +0200 Subject: [Freeswitch-users] Hangup: Always the same Q.850 cause code In-Reply-To: <4ABA0666.8090102@gmx.net> References: <4AB33C76.5050702@gmx.net> <191c3a030909180746t326e06aakef4e26a0945de195@mail.gmail.com> <4ABA0666.8090102@gmx.net> Message-ID: <4ABA263B.3070202@gmx.net> Hello, I finally solved it by using Best regards Peter Peter P GMX schrieb: > Hello Anthony, > > I did further testing on a second machine and found out the following: > After > > > > The called party receives a "NO_ANSWER" > and the calling party receives a "NORMAL_CLEARING" > > See the logs: > > Best regards > Peter > > > Logs: > ============================ > To called party: > U 82.xxx.9xx.163:5080 -> 82.xxx.9xx.165:5060 > CANCEL sip:0xxxxxxxxx299 at 21x.xx.xx.189:3273;line=fihb87zs SIP/2.0. > Via: SIP/2.0/UDP 82.xxx.9xx.163:5080;rport;branch=z9hG4bKg5SZ7829tHDae. > Route: . > Max-Forwards: 68. > From: "0xxxxxxxxx298" ;tag=1mFgvS7t9Krtj. > To: . > Call-ID: 9aab911f-22ce-122d-8686-001517956764. > CSeq: 120732503 CANCEL. > Reason: Q.850;cause=19;text="NO_ANSWER". > Content-Length: 0. > > To calling party: > U 82.xxx.9xx.163:5062 -> 82.xxx.9xx.165:5060 > SIP/2.0 480 Temporarily Unavailable. > Via: SIP/2.0/UDP 82.xxx.9xx.165;branch=z9hG4bKc08d.b2a0b296.0. > Via: SIP/2.0/UDP > 21x.xx.xx.189:2048;received=21x.xx.xx.189;branch=z9hG4bK-dnhr44fkakhd;rport=2048. > From: "0xxxxxxxxx298" ;tag=nvxy9h3rsk. > To: ;tag=5y6B9FS9ZeUZB. > Call-ID: 3c49ea1f4563-8c3hia75cxuh. > CSeq: 2 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14741M. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY. > Supported: timer, precondition, path, replaces. > Allow-Events: talk, refer. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > > > 2009-09-23 12:51:37.008546 [NOTICE] switch_ivr_originate.c:2025 Hangup > sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 [CS_CONSUME_MEDIA] > [NO_ANSWER] > 2009-09-23 12:51:37.008546 [DEBUG] switch_channel.c:1715 Send signal > sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 [KILL] > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal > sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 [BREAK] > 2009-09-23 12:51:37.008546 [DEBUG] switch_ivr_originate.c:2169 Originate > Resulted in Error Cause: 19 [NO_ANSWER] > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:398 > (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) Running State Change > CS_HANGUP > 2009-09-23 12:51:37.008546 [INFO] mod_dptools.c:2098 Originate Failed. > Cause: NO_ANSWER > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:434 > (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State HANGUP > EXECUTE sofia/internal/0xxxxxxxxx298 at mydomain.de > set(sip_ignore_remote_cause=true) > 2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:338 Channel > sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 hanging up, cause: NO_ANSWER > 2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:386 Sending CANCEL to > sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:46 > sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 Standard HANGUP, cause: > NO_ANSWER > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:434 > (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State HANGUP going to > sleep > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:479 > (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State Change CS_HANGUP > -> CS_REPORTING > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal > sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 [BREAK] > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:398 > (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) Running State Change > CS_REPORTING > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:616 > (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State REPORTING > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:53 > sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 Standard REPORTING, > cause: NO_ANSWER > 2009-09-23 12:51:37.008546 [DEBUG] mod_dptools.c:748 > sofia/internal/0xxxxxxxxx298 at mydomain.de SET > [sip_ignore_remote_cause]=[true] > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:616 > (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State REPORTING going > to sleep > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:411 > (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State Change > CS_REPORTING -> CS_DESTROY > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal > sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 [BREAK] > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:1068 Session > 129 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) Locked, Waiting on > external entities > 2009-09-23 12:51:37.008546 [NOTICE] switch_core_session.c:1086 Session > 129 (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) Ended > 2009-09-23 12:51:37.008546 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 [CS_DESTROY] > EXECUTE sofia/internal/0xxxxxxxxx298 at mydomain.de hangup() > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:559 > (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) Running State Change > CS_DESTROY > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:568 > (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State DESTROY > 2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:255 > sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 SOFIA DESTROY > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:60 > sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273 Standard DESTROY > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:568 > (sofia/external/0xxxxxxxxx299 at 21x.xx.xx.189:3273) State DESTROY going to > sleep > 2009-09-23 12:51:37.008546 [NOTICE] mod_dptools.c:633 Hangup > sofia/internal/0xxxxxxxxx298 at mydomain.de [CS_EXECUTE] [NORMAL_CLEARING] > 2009-09-23 12:51:37.008546 [DEBUG] switch_channel.c:1715 Send signal > sofia/internal/0xxxxxxxxx298 at mydomain.de [KILL] > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/0xxxxxxxxx298 at mydomain.de [BREAK] > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:494 > (sofia/internal/0xxxxxxxxx298 at mydomain.de) State EXECUTE going to sleep > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/0xxxxxxxxx298 at mydomain.de) Running State Change CS_HANGUP > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/0xxxxxxxxx298 at mydomain.de) State HANGUP > 2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:338 Channel > sofia/internal/0xxxxxxxxx298 at mydomain.de hanging up, cause: NORMAL_CLEARING > 2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:397 Responding to INVITE > with: 480 > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:46 > sofia/internal/0xxxxxxxxx298 at mydomain.de Standard HANGUP, cause: > NORMAL_CLEARING > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:434 > (sofia/internal/0xxxxxxxxx298 at mydomain.de) State HANGUP going to sleep > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:479 > (sofia/internal/0xxxxxxxxx298 at mydomain.de) State Change CS_HANGUP -> > CS_REPORTING > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/0xxxxxxxxx298 at mydomain.de [BREAK] > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:398 > (sofia/internal/0xxxxxxxxx298 at mydomain.de) Running State Change CS_REPORTING > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:616 > (sofia/internal/0xxxxxxxxx298 at mydomain.de) State REPORTING > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:53 > sofia/internal/0xxxxxxxxx298 at mydomain.de Standard REPORTING, cause: > NORMAL_CLEARING > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:616 > (sofia/internal/0xxxxxxxxx298 at mydomain.de) State REPORTING going to sleep > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:411 > (sofia/internal/0xxxxxxxxx298 at mydomain.de) State Change CS_REPORTING -> > CS_DESTROY > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:932 Send signal > sofia/internal/0xxxxxxxxx298 at mydomain.de [BREAK] > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_session.c:1068 Session > 128 (sofia/internal/0xxxxxxxxx298 at mydomain.de) Locked, Waiting on > external entities > 2009-09-23 12:51:37.008546 [NOTICE] switch_core_session.c:1086 Session > 128 (sofia/internal/0xxxxxxxxx298 at mydomain.de) Ended > 2009-09-23 12:51:37.008546 [NOTICE] switch_core_session.c:1088 Close > Channel sofia/internal/0xxxxxxxxx298 at mydomain.de [CS_DESTROY] > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:559 > (sofia/internal/0xxxxxxxxx298 at mydomain.de) Running State Change CS_DESTROY > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:568 > (sofia/internal/0xxxxxxxxx298 at mydomain.de) State DESTROY > 2009-09-23 12:51:37.008546 [DEBUG] mod_sofia.c:255 > sofia/internal/0xxxxxxxxx298 at mydomain.de SOFIA DESTROY > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:60 > sofia/internal/0xxxxxxxxx298 at mydomain.de Standard DESTROY > 2009-09-23 12:51:37.008546 [DEBUG] switch_core_state_machine.c:568 > (sofia/internal/0xxxxxxxxx298 at mydomain.de) State DESTROY going to sleep > > > > > > > > Anthony Minessale schrieb: > >> Here is what I get when I test it. >> you may want to look at your console for the blue Hangup lines and >> confirm it's your call to hangup >> >> >> >> >> >> >> >> >> >> send 644 bytes to udp/[72.128.89.126]:42988 at 14:34:29.043915: >> >> ------------------------------------------------------------------------ >> SIP/2.0 480 Temporarily Unavailable >> Via: SIP/2.0/UDP >> 10.0.1.8:50606;branch=z9hG4bK-d8754z-6077f64f00ed157e-1---d8754z-;rport=42988;received=72.128.89.126 >> From: "tony"> >;tag=352a2a46 >> To: "7016" > >;tag=HUye00UQZKySQ >> Call-ID: N2Y1MWYwZjA2YzJlY2ZhY2VjYzRhNDZmMzczYWMwN2Q. >> CSeq: 1 INVITE >> User-Agent: The Guy In IRC IS WRONG >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Reason: Q.850;cause=19;text="NO_ANSWER" >> Content-Length: 0 >> >> >> >> >> >> >> >> >> >> >> send 630 bytes to udp/[72.128.89.126]:42988 at 14:35:31.286436: >> >> ------------------------------------------------------------------------ >> SIP/2.0 486 Busy Here >> Via: SIP/2.0/UDP >> 10.0.1.8:50606;branch=z9hG4bK-d8754z-223ae00e1829097e-1---d8754z-;rport=42988;received=72.128.89.126 >> From: "tony"> >;tag=aa3b2b1d >> To: "7016" > >;tag=j4Q71UcUvvmcK >> Call-ID: NDcyNmQyYjY5YWQwOTI3MjZiZWFlZDQyNDIyZjZlMDA. >> CSeq: 1 INVITE >> User-Agent: The Guy In IRC IS WRONG >> Accept: application/sdp >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Allow-Events: talk, refer >> Reason: Q.850;cause=17;text="USER_BUSY" >> Content-Length: 0 >> >> >> >> On Fri, Sep 18, 2009 at 2:53 AM, Peter P GMX > > wrote: >> >> Hello , >> >> I try to hangup aa call with a certain cause code. >> >> If the user dials a number which we currently do not serve we send >> >> >> which gives a >> "SIP/2.0 480 Temporarily Unavailable." Message , which is fine. >> >> For the target number being busy or having another state, we use this. >> > data="sip_ignore_remote_cause=true"/> >> >> which gives a >> "SIP/2.0 486 Busy Here."" Message , which is fine in case of Busy. >> >> However in both cases the SIP mssage has the following cause code: >> Reason: Q.850;cause=16;text="NORMAL_CLEARING". >> which can lead to problems when forwarding to a PSTN Gateway. >> >> How can we achieve, that the cause code is in sync with the deiivered >> message? >> >> We are on Trunk 14741M. >> >> Best regards >> Peter >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> >> iax:guest at conference.freeswitch.org/888 >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:213-799-1400 >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From rupa at rupa.com Wed Sep 23 06:47:19 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 23 Sep 2009 08:47:19 -0500 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <0f3c01ca3c48$dc802d20$95808760$@com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <249BB60C-79F6-4A3A-8E3B-0E821D4D9046@freeswitch.org> <0f3c01ca3c48$dc802d20$95808760$@com> Message-ID: Use something like mod_easyroute to consult a database of DIDs. If you host the DID it'll give you a route to dial out. If not, route out your gateway. Or load your lcr tables with your own DIDs. Consult mod_lcr and use it's dialstring. It'll prefer longer prefix matches, so you will always win with your own customers. On Wed, Sep 23, 2009 at 7:24 AM, Peder wrote: > So if you do this, how do you call between contexts??? Say you have 100 > tenants on one box each with their own domain and they are all 4 digit for > local dialing.? If they call a 10 digit number like they are calling > outbound and it is another tenant on the same box, they don?t want to go out > and back in, they just want to be bridged over to the other context.? I > imagine you need to create a file for each context and add every other > context and did to it to check if they are on the same box, otherwise dial > outbound to the PSTN.? That doesn?t scale very well though since if there > are 100 tenants, and you add another, you need to modify the other 100 > contexts to add the new DIDs.? Or is there some built in way to do this > easily? > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian > West > Sent: Wednesday, September 23, 2009 12:18 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Multitenancy > > > > Then setup two domains in your directory and setup proper DNS its really > just that simple. > > > > /b > > > > On Sep 23, 2009, at 12:11 AM, Diego Viola wrote: > > I want user 1000-1010 to belong to?foo.org?and 2000-2020 to belong > to?bar.org, and I want both of those domains to have their own > dialplan/context. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa From peder at networkoblivion.com Wed Sep 23 07:11:44 2009 From: peder at networkoblivion.com (Peder) Date: Wed, 23 Sep 2009 09:11:44 -0500 Subject: [Freeswitch-users] Multitenancy In-Reply-To: References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <249BB60C-79F6-4A3A-8E3B-0E821D4D9046@freeswitch.org> <0f3c01ca3c48$dc802d20$95808760$@com> Message-ID: <0f9b01ca3c57$c789ab30$569d0190$@com> That's a good idea. I thought about using a DB, but I was going to have to use a lua script to look stuff up. I didn't think about easyroute or lcr. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa Schomaker Sent: Wednesday, September 23, 2009 8:47 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Multitenancy Use something like mod_easyroute to consult a database of DIDs. If you host the DID it'll give you a route to dial out. If not, route out your gateway. Or load your lcr tables with your own DIDs. Consult mod_lcr and use it's dialstring. It'll prefer longer prefix matches, so you will always win with your own customers. On Wed, Sep 23, 2009 at 7:24 AM, Peder wrote: > So if you do this, how do you call between contexts??? Say you have 100 > tenants on one box each with their own domain and they are all 4 digit for > local dialing.? If they call a 10 digit number like they are calling > outbound and it is another tenant on the same box, they don?t want to go out > and back in, they just want to be bridged over to the other context.? I > imagine you need to create a file for each context and add every other > context and did to it to check if they are on the same box, otherwise dial > outbound to the PSTN.? That doesn?t scale very well though since if there > are 100 tenants, and you add another, you need to modify the other 100 > contexts to add the new DIDs.? Or is there some built in way to do this > easily? > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian > West > Sent: Wednesday, September 23, 2009 12:18 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Multitenancy > > > > Then setup two domains in your directory and setup proper DNS its really > just that simple. > > > > /b > > > > On Sep 23, 2009, at 12:11 AM, Diego Viola wrote: > > I want user 1000-1010 to belong to?foo.org?and 2000-2020 to belong > to?bar.org, and I want both of those domains to have their own > dialplan/context. > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -Rupa _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Wed Sep 23 07:14:21 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Sep 2009 09:14:21 -0500 Subject: [Freeswitch-users] Using Sofia-SIP with ICE NAT Traversal Mechanism In-Reply-To: References: Message-ID: <97C8AEB9-F4A1-467B-BFE9-70A122E51E2E@freeswitch.org> I'm not comfortable adding libnice into FreeSWITCH as it depends on glib and that would add bloat in my opinion... is there no other license compatible option? /b On Sep 23, 2009, at 8:42 AM, Muhammad Shahzad wrote: > Yup, that's a good idea but not in my project list right now. > > Thank you. > > > On Wed, Sep 23, 2009 at 7:17 PM, afshin afzali > wrote: > Don't you think is better to integrate LibNICE to FreeSWITCH's RTP > stack than a module such as mod_msn / mod_dingaling ? > > -- afshin > > On Wed, Sep 23, 2009 at 4:21 PM, Muhammad Shahzad > wrote: > No mod_dingaling does not use LibNICE. However, i have plans to > integrate NICE with Sofia in mod_msn project, which is at the moment > moving with very slow pace due to some trouble in reverse > engineering MSNP-18 protocol (used in Windows Live Messenger 2009). > > Thank you. > > > On Wed, Sep 23, 2009 at 6:37 PM, afshin afzali > wrote: > Hi, > > I know that FreeSWITCH uses libdingaling to talk to Jingle call > parties. Also I know that Jingle Protocol uses ICE protocol to > traverse NAT devices. So : > does libdingaling use an open library such as libnice for ICE? > Is it possible to use the ICE implementation in Sofia-SIP endpoint? > If not, how could I integrate an open ICE library in Sofia-SIP? > > Regards, > -- afshin > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > > -- > ________________________________________________________ > | > | > | FATAL > ERROR --- > O X | > |_______________________________________________________| > | You have moved the > mouse. | > | Windows must be restarted for the changes to take effect. | > | > | > ####################################/ > > > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > > > > -- > ________________________________________________________ > | > | > | FATAL > ERROR --- > O X | > |_______________________________________________________| > | You have moved the > mouse. | > | Windows must be restarted for the changes to take effect. | > | > | > ####################################/ > > > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/a811fa65/attachment.html From tculjaga at gmail.com Wed Sep 23 07:15:30 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Wed, 23 Sep 2009 16:15:30 +0200 Subject: [Freeswitch-users] Event-Name=CHANNEL_DATA - duplicate variables Message-ID: <65d96fc80909230715m43ff6774k3fb495277743425f@mail.gmail.com> Hello, the setup is like this: CALLING_USER(context:Public) => FS => ENDPOINT The CALLING_USER places a call towards FS. The call is answered and a welcome prompt is being played. After that, FS bridges the call towards an ENDPOINT. I've built a custom module where among other stuff i subscribe to events... switch_core_add_state_handler(&albatross_state_handler); switch_state_handler_table_t albatross_state_handler = { /* on_init */ process_init, /* on_routing */ NULL, /* Need to add a check here for anything in their account before routing */ /* on_execute */ NULL, /* Turn on heartbeat for this session and do an initial account check */ /* on_hangup */ process_hangup, /* On hangup - most important place to go bill */ /* on_exch_media */ NULL, /* on_soft_exec */ NULL, /* on_consume_med */ NULL, /* on_hibernate */ NULL, /* on_reset */ NULL }; So, according to the callflow specified above, on hangup i get two events ... (1st B-LEG followed by A-LEG). well .. it sems i have double variables on the A-LEG ?!?!?! .. whats wrong hewre ? variable_duration=0 variable_billsec=0 variable_progresssec=0 variable_answersec=0 variable_progress_mediasec=0 variable_flow_billsec=0 variable_mduration=0 variable_billmsec=0 variable_progressmsec=0 variable_answermsec=0 variable_progress_mediamsec=0 variable_flow_billmsec=0 variable_uduration=0 variable_billusec=0 variable_progressusec=0 variable_answerusec=0 variable_progress_mediausec=0 variable_flow_billusec=0 Dump:** 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Name=CHANNEL_DATA *<= B_LEG (Outbound call)* 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Core-UUID=ca97d306-a840-11de-8ae7-5585e27e6446 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 FreeSWITCH-Hostname=node1 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 FreeSWITCH-IPv4=192.168.102.81 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 FreeSWITCH-IPv6=::1 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Date-Local=2009-09-23 14:59:22 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Date-GMT=Wed, 23 Sep 2009 12:59:22 GMT 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Date-Timestamp=1253710762323094 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Calling-File=mod_albatross.c 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Calling-Function=process_hangup 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Calling-Line-Number=2366 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-State=CS_HANGUP 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-State-Number=10 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-Name=sofia/external/0914392122 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Unique-ID=de716bd0-a840-11de-8ae7-5585e27e6446 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Call-Direction=outbound 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Presence-Call-Direction=outbound 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Answer-State=answered 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-Read-Codec-Name=PCMA 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-Read-Codec-Rate=8000 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-Write-Codec-Name=PCMA 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-Write-Codec-Rate=8000 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Username=016659280 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Dialplan=XML 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Caller-ID-Name=016659280 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Caller-ID-Number=016659280 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Network-Addr=172.16.1.20 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Destination-Number=0914392122 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Unique-ID=de716bd0-a840-11de-8ae7-5585e27e6446 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Source=mod_sofia 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Context=public 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Channel-Name=sofia/external/0914392122 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Profile-Index=1 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Profile-Created-Time=1253710743103107 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Channel-Created-Time=1253710743103107 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Channel-Answered-Time=1253710748807099 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Channel-Progress-Time=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Channel-Progress-Media-Time=1253710745451100 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Channel-Hangup-Time=1253710762323094 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Channel-Transfer-Time=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Screen-Bit=true 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Privacy-Hide-Name=false 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Privacy-Hide-Number=false 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Username=016659280 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Dialplan=XML 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Caller-ID-Name=016659280 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Caller-ID-Number=016659280 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Network-Addr=172.16.1.20 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Destination-Number=012468601_Kviz 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Unique-ID=d2242912-a840-11de-8ae7-5585e27e6446 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Source=mod_sofia 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Context=public 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Channel-Name=sofia/external/016659280 at sip-priv.amis.hr 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Profile-Created-Time=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Channel-Created-Time=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Channel-Answered-Time=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Channel-Progress-Time=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Channel-Progress-Media-Time=1253710745451100 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Channel-Hangup-Time=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Channel-Transfer-Time=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Screen-Bit=true 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Privacy-Hide-Name=false 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Other-Leg-Privacy-Hide-Number=false 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_gateway_name=gw1 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_channel_name=sofia/external/0914392122 2009-09-23 14:59:22.323094 [NOTICE] mod_albatross.c:2320 *** HANGUP RECEIVED, BYE... 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_destination_url=sip:0914392122 at 172.16.1.20 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_is_outbound=true 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_h_P-Access-Network-Info=ADSL;dsl_location="NOA=4;APRI=1;ADD=3851";network-provided 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_cid_type=pid 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_max_forwards=30 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_originator_codec=PCMA at 8000h@20i 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_originator=d2242912-a840-11de-8ae7-5585e27e6446 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_execute_on_answer=updateQuizServiceStatus_ch in 012468601, in connected 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_switch_m_sdp=v=0 o=cp10 125371067255 125371067255 IN IP4 172.16.2.42 s=SIP Call c=IN IP4 172.16.2.42 t=0 0 m=audio 34474 RTP/AVP 8 0 18 125 101 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:125 CLEARMODE/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_service_number=012468601 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_dialed_number=0914392122 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_originate_early_media=true 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sofia_profile_name=external 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_call_id=b5cfd0c9-22e3-122d-2c98-001e684a25c5 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_switch_r_sdp=v=0 o=cp10 125371069278 125371069279 IN IP4 172.16.2.42 s=SIP Call c=IN IP4 172.16.2.42 t=0 0 m=audio 34972 RTP/AVP 8 0 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=ptime:20 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_remote_media_ip=172.16.2.42 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_remote_media_port=34972 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_read_codec=PCMA 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_read_rate=8000 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Name=CHANNEL_DATA *<= A-LEG (Inbound)* 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Core-UUID=ca97d306-a840-11de-8ae7-5585e27e6446 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 FreeSWITCH-Hostname=node1 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 FreeSWITCH-IPv4=192.168.102.81 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 FreeSWITCH-IPv6=::1 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Date-Local=2009-09-23 14:59:22 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Date-GMT=Wed, 23 Sep 2009 12:59:22 GMT 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Date-Timestamp=1253710762323094 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Calling-File=mod_albatross.c 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Calling-Function=process_hangup 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_write_codec=PCMA 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_write_rate=8000 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_local_media_ip=172.16.3.2 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_local_media_port=42684 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_bridge_channel=sofia/external/016659280 at sip-priv.amis.hr 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_bridge_uuid=d2242912-a840-11de-8ae7-5585e27e6446 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_signal_bond=d2242912-a840-11de-8ae7-5585e27e6446 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_reply_host=172.16.1.20 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_reply_port=5060 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_endpoint_disposition=ANSWER 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_current_application_data=in 012468601, in connected 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_current_application=updateQuizServiceStatus_ch 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_term_status=200 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_term_cause=16 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_user_agent=Cirpack/v4.42d (gw_sip) 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_hangup_disposition=recv_bye 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_hangup_cause=NORMAL_CLEARING 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_hangup_cause_q850=16 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_digits_dialed=none 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_start_stamp=2009-09-23 14:59:03 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_profile_start_stamp=2009-09-23 14:59:03 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_answer_stamp=2009-09-23 14:59:08 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progress_media_stamp=2009-09-23 14:59:05 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_end_stamp=2009-09-23 14:59:22 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_start_epoch=1253710743 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_start_uepoch=1253710743103107 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_profile_start_epoch=1253710743 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_profile_start_uepoch=1253710743103107 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_answer_epoch=1253710748 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_answer_uepoch=1253710748807099 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progress_epoch=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progress_uepoch=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progress_media_epoch=1253710745 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progress_media_uepoch=1253710745451100 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_end_epoch=1253710762 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_end_uepoch=1253710762323094 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_last_app=updateQuizServiceStatus_ch 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_last_arg=in 012468601, in connected 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Calling-Line-Number=2366 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_caller_id="016659280" <016659280> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_duration=19 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-State=CS_HANGUP 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_billsec=14 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-State-Number=10 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progresssec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-Name=sofia/external/016659280 at sip-priv.amis.hr 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_answersec=5 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progress_mediasec=2 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Unique-ID=d2242912-a840-11de-8ae7-5585e27e6446 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_flow_billsec=19 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Call-Direction=inbound 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_mduration=19220 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Presence-Call-Direction=inbound 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_billmsec=13516 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Answer-State=answered 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progressmsec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-Read-Codec-Name=PCMA 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_answermsec=5704 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-Read-Codec-Rate=8000 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progress_mediamsec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-Write-Codec-Name=PCMA 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_flow_billmsec=19220 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Channel-Write-Codec-Rate=8000 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_uduration=19219987 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_billusec=13515995 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Username=016659280 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progressusec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Dialplan=XML 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_answerusec=5703992 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progress_mediausec=2347993 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Caller-ID-Name=016659280 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_flow_billusec=19219987 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Caller-ID-Number=016659280 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_raw_bytes=144996 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_media_bytes=144996 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Network-Addr=172.16.1.20 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_packet_count=843 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Destination-Number=012468601_Kviz 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_media_packet_count=843 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Unique-ID=d2242912-a840-11de-8ae7-5585e27e6446 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_skip_packet_count=1 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Source=mod_sofia 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_jb_packet_count=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Context=public 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_dtmf_packet_count=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Channel-Name=sofia/external/016659280 at sip-priv.amis.hr 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_cng_packet_count=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Screen-Bit=true 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_flush_packet_count=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Privacy-Hide-Name=false 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_out_raw_bytes=143964 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Privacy-Hide-Number=false 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_out_media_bytes=143964 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_received_ip=172.16.1.20 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_out_packet_count=837 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_received_port=5060 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_out_media_packet_count=837 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_via_protocol=udp 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_out_skip_packet_count=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_from_params=user=phone 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_out_dtmf_packet_count=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_from_user=016659280 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_out_cng_packet_count=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_from_uri=016659280 at sip-priv.amis.hr 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_from_host=sip-priv.amis.hr 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_from_user_stripped=016659280 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_from_tag=11746-FX-007c2917-23e2091e3 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sofia_profile_name=external 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_P-Asserted-Identity=016659280 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_cid_type=pid 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2407 siId: (null) 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_req_params=user=phone 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2408 ani: 016659280 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_req_user=012468601 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2409 dnis: 0914392122 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_req_port=5060 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2410 dialed_number: 0914392122 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2412 start_stamp: 2009-09-23 14:59:03 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_req_uri= 012468601 at 172.16.3.2:5060 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2413 answer_stamp: 2009-09-23 14:59:08 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2414 progress_stamp: (null) 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_req_host=172.16.3.2 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_to_params=user=phone 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2415 progress_media_stamp: 2009-09-23 14:59:05 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_to_user=012468601 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2416 end_stamp: 2009-09-23 14:59:22 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_to_uri= 012468601 at 172.16.3.2 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2420 privacy_hide_number: false 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_to_host=172.16.3.2 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_contact_user=nobody 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_contact_port=5060 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_contact_uri=nobody at 172.16.1.20:5060 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_contact_host=172.16.1.20 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_channel_name=sofia/external/016659280 at sip-priv.amis.hr 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_call_id= 11746-QJ-007c2916-59db25232 at sip-priv.amis.hr 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_user_agent=Cirpack/v4.42d (gw_sip) 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_via_host=172.16.1.20 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_via_port=5060 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_max_forwards=31 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_h_P-Access-Network-Info=ADSL;dsl_location="NOA=4;APRI=1;ADD=3851";network-provided 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_switch_r_sdp=v=0 o=cp10 125371067255 125371067255 IN IP4 172.16.2.42 s=SIP Call c=IN IP4 172.16.2.42 t=0 0 m=audio 34474 RTP/AVP 8 0 18 125 101 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:125 CLEARMODE/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_remote_media_ip=172.16.2.42 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_remote_media_port=34474 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_read_codec=PCMA 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_read_rate=8000 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_write_codec=PCMA 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_write_rate=8000 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_service_type_id=1 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_local_media_ip=172.16.3.2 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_local_media_port=27686 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_price_prompt=3.66kn_novo_upozorenje.wav 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_dialed_number=012468601 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_bNum=012468601 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_service_status1=win 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_number_2_connect=0914392122 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_next_number_2_connect=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_next_number_2_display=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_service_instance=130 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_service_id=2 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_quiz_status=win 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_not_working_prompt=2/emisija trenutno nije u tijeku-za telefone.wav 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_service_win_prompt=2/bit cete spojeni u emisiju1.wav 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_service_loose_prompt=2/zovi ponovo.wav 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_outside_call=true 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_endpoint_disposition=ANSWER 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_playback_ms=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_playback_samples=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_bypass_media=false 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_hangup_after_bridge=true 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_nolocal:execute_on_answer=updateQuizServiceStatus_ch in 012468601, in connected 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_export_vars=nolocal:execute_on_answer 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_current_application_data=[service_number=012468601,dialed_number=0914392122]sofia/gateway/gw1/0914392122 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_current_application=bridge 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_switch_m_sdp=v=0 o=cp10 125371069278 125371069279 IN IP4 172.16.2.42 s=SIP Call c=IN IP4 172.16.2.42 t=0 0 m=audio 34972 RTP/AVP 8 0 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=rtpmap:0 PCMU/8000/1 a=ptime:20 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_originate_disposition=SUCCESS 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_bridge_channel=sofia/external/0914392122 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_bridge_uuid=de716bd0-a840-11de-8ae7-5585e27e6446 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_signal_bond=de716bd0-a840-11de-8ae7-5585e27e6446 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_hangup_phrase=OK 2009-09-23 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14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progress_mediasec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_flow_billsec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_mduration=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_billmsec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progressmsec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_answermsec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progress_mediamsec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_flow_billmsec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_uduration=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_billusec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progressusec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_answerusec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_progress_mediausec=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_flow_billusec=0* 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_hangup_disposition=send_bye 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_raw_bytes=341764 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_media_bytes=341764 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_packet_count=1987 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_media_packet_count=1987 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_skip_packet_count=10 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_jb_packet_count=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_dtmf_packet_count=0 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_rtp_audio_in_cng_packet_count=0 2009-09-23 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/58f136c1/attachment-0001.html From svetikvoip at gmail.com Wed Sep 23 07:34:21 2009 From: svetikvoip at gmail.com (Svetik VOIP) Date: Wed, 23 Sep 2009 10:34:21 -0400 Subject: [Freeswitch-users] No ring tone while recording incoming call. Please help. Message-ID: <94790b850909230734x1d791927ifb0a52a3b1d1d2f1@mail.gmail.com> Brian, Thank yo very much for your reply. I have tried to add transfer_ringback action, but it did not solve my problem. Destination phone is ringing, but the person who is calling does not hear ringing tone in hte handset. Is there anything in the logfile that can help you to identify the problem? Closest I can see is: 2009-09-22 17:18:05.444402 [DEBUG] switch_ivr_originate.c:1738 Raw Codec Activation Success L16 at 8000hz 1 channel 20ms 2009-09-22 17:18:05.444402 [DEBUG] switch_ivr_originate.c:1797 Play Ringback Tone [%(2000,4000,440.0,480.0)] 2009-09-22 17:18:05.447237 [DEBUG] switch_core_io.c:232 sofia/external/ 4163641113 at 67.205.74.164 receive message [TRANSCODING_NECESSARY] 2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/ sip:main at 192.168.0.121:5060 entering state [proceeding][180] 2009-09-22 17:18:05.463192 [NOTICE] sofia.c:3376 Ring-Ready sofia/internal/ sip:main at 192.168.0.121:5060! 2009-09-22 17:18:14.739182 [DEBUG] sofia.c:3312 Channel sofia/external/ 4163641113 at 67.205.74.164 entering state [terminated][487] 2009-09-22 17:18:14.739182 [NOTICE] sofia.c:3873 Hangup sofia/external/ 4163641113 at 67.205.74.164 [CS_EXECUTE] [ORIGINATOR_CANCEL] Thank you, Igor >set ringback before record_session and also set transfer_ringback >because record_session causes an pre-answer. > >/b > >On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote: > >> Hi, >> >> I have trouble recording incoming calls with FreeSwitch. >> >> I have followed the instruction from Misc. Dialplan Tools record >> session >> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session) >> It works well for outgoing calls, but I have the problem with >> incoming calls. >> >> The person who is calling does not hear ring tone, he hears just the >> silence until >> I pick up the phone. Everything else is working, we can talk, >> conversation is recorded. >> >> Here is a copy of my dialplan for incoming calls >> /usr/local/freeswitch/conf/dialplan/public/voipms.xml >> >> >> >> > expression="XXXXXXXXXX"> >> >> >> > data="RECORD_SOFTWARE=FreeSwitch"/> >> > data="RECORD_ARTIST=FreeSwitch"/> >> > data="RECORD_COMMENT=FreeSwitch"/> >> >> >> >> >> >> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/244b40e2/attachment.html From mike at jerris.com Wed Sep 23 07:40:31 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 10:40:31 -0400 Subject: [Freeswitch-users] Using Sofia-SIP with ICE NAT Traversal Mechanism In-Reply-To: <97C8AEB9-F4A1-467B-BFE9-70A122E51E2E@freeswitch.org> References: <97C8AEB9-F4A1-467B-BFE9-70A122E51E2E@freeswitch.org> Message-ID: <32A6F39D-08E2-4C45-A98C-1E6F44BCEC7F@jerris.com> We already have ice support in freeswitch, granted it is the slightly twisted ice from the old jingle, but this should not be difficult to fix. Knowing what I know about libnice architechture I can say almost without doubt that it will never fit well into freeeswitch. Is the basis of this question and you loooking for an ice library on the sofia list just to support ice in sip? If so, for both sip and msn the path of least resistance and probably the only way that would work would be to address this within our existing ice implementation. Mike On Sep 23, 2009, at 10:14 AM, Brian West wrote: > I'm not comfortable adding libnice into FreeSWITCH as it depends on > glib and that would add bloat in my opinion... is there no other > license compatible option? > > /b > > On Sep 23, 2009, at 8:42 AM, Muhammad Shahzad wrote: > >> Yup, that's a good idea but not in my project list right now. >> >> Thank you. >> >> >> On Wed, Sep 23, 2009 at 7:17 PM, afshin afzali > > wrote: >> Don't you think is better to integrate LibNICE to FreeSWITCH's RTP >> stack than a module such as mod_msn / mod_dingaling ? >> >> -- afshin >> >> On Wed, Sep 23, 2009 at 4:21 PM, Muhammad Shahzad > > wrote: >> No mod_dingaling does not use LibNICE. However, i have plans to >> integrate NICE with Sofia in mod_msn project, which is at the >> moment moving with very slow pace due to some trouble in reverse >> engineering MSNP-18 protocol (used in Windows Live Messenger 2009). >> >> Thank you. >> >> >> On Wed, Sep 23, 2009 at 6:37 PM, afshin afzali > > wrote: >> Hi, >> >> I know that FreeSWITCH uses libdingaling to talk to Jingle call >> parties. Also I know that Jingle Protocol uses ICE protocol to >> traverse NAT devices. So : >> does libdingaling use an open library such as libnice for ICE? >> Is it possible to use the ICE implementation in Sofia-SIP endpoint? >> If not, how could I integrate an open ICE library in Sofia-SIP? >> >> Regards, >> -- afshin >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> >> >> -- >> ________________________________________________________ >> | >> >> >> >> >> >> >> >> >> >> | >> | FATAL >> ERROR --- >> O X | >> |_______________________________________________________| >> | You have moved the >> mouse. | >> | Windows must be restarted for the changes to take effect. | >> | >> | >> ####################################/ >> >> >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> users >> http://www.freeswitch.org >> >> >> >> >> -- >> ________________________________________________________ >> | >> >> >> >> >> >> >> >> >> >> | >> | FATAL >> ERROR --- >> O X | >> |_______________________________________________________| >> | You have moved the >> mouse. | >> | Windows must be restarted for the changes to take effect. | >> | >> | >> ####################################/ >> >> >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/651ff24d/attachment.html From grae at digilord.net Wed Sep 23 08:19:26 2009 From: grae at digilord.net (digilord) Date: Wed, 23 Sep 2009 08:19:26 -0700 Subject: [Freeswitch-users] Automagic Phone Provisioning Message-ID: <1253719166.4738.7.camel@digilords-desktop.digilord.net> Hello all, I know this is done and I think I figured out how to do it but I don't want to reinvent the wheel so here goes. I am looking for a program that will sit on the PBX. This program will intercept DHCP reply packets destined for phones, inject "option 66" into the packet and release it back onto the network. Some of you might be wondering why I want a program like this. Simple. Lazy clients. They don't want to mess with their network infrastructure to assist us with automated deployment of SIP devices. They also don't want 50-100 devices connecting to an off site server downloading 20-40MB of firmware on a reboot. The PBX is not hard coded with an IP address. It's DHCP. They were willing to allow the PBX on the network and assign it a static DHCP address. Is what I am looking for not possible? Does someone have a sensible solution that doesn't involve dropping the client (yes someone suggested that)? Thanks in advance for any help you can give. DigiLord -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/88418f61/attachment-0001.bin From aep.lists at it46.se Wed Sep 23 08:32:45 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Wed, 23 Sep 2009 17:32:45 +0200 Subject: [Freeswitch-users] Call Files for a dialer engine Message-ID: <994476df9083131426fb1d69e257c93b.squirrel@correo.nodo50.org> I am exploring the possibility of building a Dialer that emulates the logic of Call Files in asterisk. A CallerID catcher is creating CUSTOM events that I can store in a database. I can trigger callbacks using ESL but I wonder what is the best way/nicer/geekier to do something like outgoing calls in * /aep -- Stopping junk mailers is good for the environment From shaheryarkh at googlemail.com Wed Sep 23 08:35:20 2009 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Wed, 23 Sep 2009 21:35:20 +0600 Subject: [Freeswitch-users] Using Sofia-SIP with ICE NAT Traversal Mechanism In-Reply-To: <32A6F39D-08E2-4C45-A98C-1E6F44BCEC7F@jerris.com> References: <97C8AEB9-F4A1-467B-BFE9-70A122E51E2E@freeswitch.org> <32A6F39D-08E2-4C45-A98C-1E6F44BCEC7F@jerris.com> Message-ID: Personally i am not a fan of GLib as well and always prefer STL over it due to so many good reasons. But on the other hand libnice is the only library that has Microsoft extensions to ICE protocol, which are required for mod_msn to work. So far on mod_msn, i am able to send and receive voice call requests to / from WLM 2009, but upon answer two way voice is not working. I am able to hear voice from WML 2009 but WML 2009 client can't hear anything. So nowadays i am reviewing libNICE to fix this problem. If this does not work (and so far it does not seems to work) then i would like to use FS ICE library, provided that you guys allow me to extend it to support Microsoft extensions..! Thank you. On Wed, Sep 23, 2009 at 8:40 PM, Michael Jerris wrote: > We already have ice support in freeswitch, granted it is the slightly > twisted ice from the old jingle, but this should not be difficult to fix. > Knowing what I know about libnice architechture I can say almost without > doubt that it will never fit well into freeeswitch. Is the basis of this > question and you loooking for an ice library on the sofia list just to > support ice in sip? If so, for both sip and msn the path of least > resistance and probably the only way that would work would be to address > this within our existing ice implementation. > > Mike > > On Sep 23, 2009, at 10:14 AM, Brian West wrote: > > I'm not comfortable adding libnice into FreeSWITCH as it depends on glib > and that would add bloat in my opinion... is there no other license > compatible option? > /b > > On Sep 23, 2009, at 8:42 AM, Muhammad Shahzad wrote: > > Yup, that's a good idea but not in my project list right now. > > Thank you. > > > On Wed, Sep 23, 2009 at 7:17 PM, afshin afzali < > a.afzali2003 at gmail.com> wrote: > >> Don't you think is better to integrate LibNICE to FreeSWITCH's RTP stack >> than a module such as mod_msn / mod_dingaling ? >> >> -- afshin >> >> On Wed, Sep 23, 2009 at 4:21 PM, Muhammad Shahzad < >> shaheryarkh at googlemail.com> wrote: >> >>> No mod_dingaling does not use LibNICE. However, i have plans to integrate >>> NICE with Sofia in mod_msn project, which is at the moment moving with very >>> slow pace due to some trouble in reverse engineering MSNP-18 protocol (used >>> in Windows Live Messenger 2009). >>> >>> Thank you. >>> >>> >>> On Wed, Sep 23, 2009 at 6:37 PM, afshin afzali < >>> a.afzali2003 at gmail.com> wrote: >>> >>>> Hi, >>>> >>>> I know that FreeSWITCH uses libdingaling to talk to Jingle call parties. >>>> Also I know that Jingle Protocol uses ICE protocol to traverse NAT devices. >>>> So : >>>> does libdingaling use an open library such as libnice for ICE? >>>> Is it possible to use the ICE implementation in Sofia-SIP endpoint? >>>> If not, how could I integrate an open ICE library in Sofia-SIP? >>>> >>>> Regards, >>>> -- afshin >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> ________________________________________________________ >>> | >>> | >>> | FATAL ERROR >>> --- O X | >>> |_______________________________________________________| >>> | You have moved the mouse. >>> | >>> | Windows must be restarted for the changes to take effect. | >>> | >>> | >>> ####################################/ >>> >>> >>> Muhammad Shahzad >>> ----------------------------------- >>> CISCO Rich Media Communication Specialist (CRMCS) >>> CISCO Certified Network Associate (CCNA) >>> Cell: +92 334 422 40 88 >>> MSN: shari_786pk at hotmail.com >>> Email: shaheryarkh at googlemail.com >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> >>> FreeSWITCH-users at 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 at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > ________________________________________________________ > | > | > | FATAL ERROR --- > O X | > |_______________________________________________________| > | You have moved the mouse. > | > | Windows must be restarted for the changes to take effect. | > | > | > ####################################/ > > > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > _______________________________________________ > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- ________________________________________________________ | | | FATAL ERROR --- O X | |_______________________________________________________| | You have moved the mouse. | | Windows must be restarted for the changes to take effect. | | | ####################################/ Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/5ad345e2/attachment.html From rupa at rupa.com Wed Sep 23 08:45:21 2009 From: rupa at rupa.com (Rupa Schomaker) Date: Wed, 23 Sep 2009 10:45:21 -0500 Subject: [Freeswitch-users] Automagic Phone Provisioning In-Reply-To: <1253719166.4738.7.camel@digilords-desktop.digilord.net> References: <1253719166.4738.7.camel@digilords-desktop.digilord.net> Message-ID: On Wed, Sep 23, 2009 at 10:19 AM, digilord wrote: > Hello all, > ? ? ? ?I know this is done and I think I figured out how to do it but I don't > want to reinvent the wheel so here goes. ?I am looking for a program > that will sit on the PBX. ?This program will intercept DHCP reply > packets destined for phones, inject "option 66" into the packet and > release it back onto the network. > Not going to happen. Once the DHCP reply is on the wire it is going to be picked up by the phone. You could as the lazy client to setup your phones on their own VLAN and have the pbx provide dhcp services to that VLAN but that might be more work than having them set option 66 to begin with. -- -Rupa From brian at freeswitch.org Wed Sep 23 08:46:18 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Sep 2009 10:46:18 -0500 Subject: [Freeswitch-users] Automagic Phone Provisioning In-Reply-To: <1253719166.4738.7.camel@digilords-desktop.digilord.net> References: <1253719166.4738.7.camel@digilords-desktop.digilord.net> Message-ID: What you want is NOT possible the way you describe it. Snom does a multicast PNP which lets you reply with a notify. Polycom does a DHCPINFORM which lets you respond with a DHCPACK with additional options. Aastra does MDNS which dictates where to go get the configs. /b On Sep 23, 2009, at 10:19 AM, digilord wrote: > Hello all, > I know this is done and I think I figured out how to do it but I > don't > want to reinvent the wheel so here goes. I am looking for a program > that will sit on the PBX. This program will intercept DHCP reply > packets destined for phones, inject "option 66" into the packet and > release it back onto the network. > > Some of you might be wondering why I want a program like this. > Simple. > Lazy clients. They don't want to mess with their network > infrastructure > to assist us with automated deployment of SIP devices. They also > don't > want 50-100 devices connecting to an off site server downloading > 20-40MB > of firmware on a reboot. > > The PBX is not hard coded with an IP address. It's DHCP. They were > willing to allow the PBX on the network and assign it a static DHCP > address. > > Is what I am looking for not possible? Does someone have a sensible > solution that doesn't involve dropping the client (yes someone > suggested > that)? > > Thanks in advance for any help you can give. > > DigiLord From anthony.minessale at gmail.com Wed Sep 23 08:49:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Sep 2009 10:49:20 -0500 Subject: [Freeswitch-users] Call Files for a dialer engine In-Reply-To: <994476df9083131426fb1d69e257c93b.squirrel@correo.nodo50.org> References: <994476df9083131426fb1d69e257c93b.squirrel@correo.nodo50.org> Message-ID: <191c3a030909230849o351bdbaah3dd48c85db07c56b@mail.gmail.com> make an esl script that monitors a dir for new files, and push the contents into your same db? On Wed, Sep 23, 2009 at 10:32 AM, Alberto Escudero wrote: > I am exploring the possibility of building a Dialer that emulates the > logic of Call Files in asterisk. > A CallerID catcher is creating CUSTOM events that I can store in a > database. I can trigger callbacks using ESL but I wonder what is the best > way/nicer/geekier to do something like outgoing calls in * > > /aep > > -- > Stopping junk mailers is good for the environment > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/2e2488d2/attachment.html From dave at 3c.co.uk Wed Sep 23 08:51:46 2009 From: dave at 3c.co.uk (David Knell) Date: Wed, 23 Sep 2009 18:51:46 +0300 Subject: [Freeswitch-users] Automagic Phone Provisioning In-Reply-To: <1253719166.4738.7.camel@digilords-desktop.digilord.net> References: <1253719166.4738.7.camel@digilords-desktop.digilord.net> Message-ID: <1253721106.18177.47.camel@dk-d820> I don't think it's trivially possible, unless you can stick the PBX between the DHCP server and the rest of the network. The reason is that DHCP reply packets are not broadcast, but sent back to the MAC address of the originator, so your Ethernet switch won't even let your PBX see the replies. Even if it could see them, add something to them and retransmit, the client will almost certainly already have seen the original reply and would be likely just to ignore the later one. Any reason why you can't just ask them to add that option to their existing DHCP server? And how do you know it's done, and, if you've figured out how to do it, could you share? --Dave > Hello all, > I know this is done and I think I figured out how to do it but I don't > want to reinvent the wheel so here goes. I am looking for a program > that will sit on the PBX. This program will intercept DHCP reply > packets destined for phones, inject "option 66" into the packet and > release it back onto the network. > > Some of you might be wondering why I want a program like this. Simple. > Lazy clients. They don't want to mess with their network infrastructure > to assist us with automated deployment of SIP devices. They also don't > want 50-100 devices connecting to an off site server downloading 20-40MB > of firmware on a reboot. > > The PBX is not hard coded with an IP address. It's DHCP. They were > willing to allow the PBX on the network and assign it a static DHCP > address. > > Is what I am looking for not possible? Does someone have a sensible > solution that doesn't involve dropping the client (yes someone suggested > that)? > > Thanks in advance for any help you can give. > > DigiLord > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- David Knell, Director, 3C Limited T: +44 20 3298 2000 E: dave at 3c.co.uk W: http://www.3c.co.uk From anthony.minessale at gmail.com Wed Sep 23 08:53:52 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 23 Sep 2009 10:53:52 -0500 Subject: [Freeswitch-users] Using Sofia-SIP with ICE NAT Traversal Mechanism In-Reply-To: References: <97C8AEB9-F4A1-467B-BFE9-70A122E51E2E@freeswitch.org> <32A6F39D-08E2-4C45-A98C-1E6F44BCEC7F@jerris.com> Message-ID: <191c3a030909230853v5e92cfebl40dc93e7bd2e3df3@mail.gmail.com> libdingaling does not do the gtalk ice stuff, mod_dingaling does by using utils in the FS core. We have stun client code, random string generators in switch_utils.c and settings in the rtp stack to send and recv authed stun packets using the aforementioned functions mixed in with the audio. Whatever you have to change will have to be added to FS core and it cannot be with any lib that depends on glib or it would make our core depend on glib which is not going to happen because it would make support on windows a nightmare. On Wed, Sep 23, 2009 at 10:35 AM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > Personally i am not a fan of GLib as well and always prefer STL over it due > to so many good reasons. But on the other hand libnice is the only library > that has Microsoft extensions to ICE protocol, which are required for > mod_msn to work. > > So far on mod_msn, i am able to send and receive voice call requests to / > from WLM 2009, but upon answer two way voice is not working. I am able to > hear voice from WML 2009 but WML 2009 client can't hear anything. So > nowadays i am reviewing libNICE to fix this problem. If this does not work > (and so far it does not seems to work) then i would like to use FS ICE > library, provided that you guys allow me to extend it to support Microsoft > extensions..! > > Thank you. > > > > On Wed, Sep 23, 2009 at 8:40 PM, Michael Jerris wrote: > >> We already have ice support in freeswitch, granted it is the slightly >> twisted ice from the old jingle, but this should not be difficult to fix. >> Knowing what I know about libnice architechture I can say almost without >> doubt that it will never fit well into freeeswitch. Is the basis of this >> question and you loooking for an ice library on the sofia list just to >> support ice in sip? If so, for both sip and msn the path of least >> resistance and probably the only way that would work would be to address >> this within our existing ice implementation. >> >> Mike >> >> On Sep 23, 2009, at 10:14 AM, Brian West wrote: >> >> I'm not comfortable adding libnice into FreeSWITCH as it depends on glib >> and that would add bloat in my opinion... is there no other license >> compatible option? >> /b >> >> On Sep 23, 2009, at 8:42 AM, Muhammad Shahzad wrote: >> >> Yup, that's a good idea but not in my project list right now. >> >> Thank you. >> >> >> On Wed, Sep 23, 2009 at 7:17 PM, afshin afzali < >> a.afzali2003 at gmail.com> wrote: >> >>> Don't you think is better to integrate LibNICE to FreeSWITCH's RTP stack >>> than a module such as mod_msn / mod_dingaling ? >>> >>> -- afshin >>> >>> On Wed, Sep 23, 2009 at 4:21 PM, Muhammad Shahzad < >>> shaheryarkh at googlemail.com> wrote: >>> >>>> No mod_dingaling does not use LibNICE. However, i have plans to >>>> integrate NICE with Sofia in mod_msn project, which is at the moment moving >>>> with very slow pace due to some trouble in reverse engineering MSNP-18 >>>> protocol (used in Windows Live Messenger 2009). >>>> >>>> Thank you. >>>> >>>> >>>> On Wed, Sep 23, 2009 at 6:37 PM, afshin afzali < >>>> a.afzali2003 at gmail.com> wrote: >>>> >>>>> Hi, >>>>> >>>>> I know that FreeSWITCH uses libdingaling to talk to Jingle call >>>>> parties. Also I know that Jingle Protocol uses ICE protocol to traverse NAT >>>>> devices. So : >>>>> does libdingaling use an open library such as libnice for ICE? >>>>> Is it possible to use the ICE implementation in Sofia-SIP endpoint? >>>>> If not, how could I integrate an open ICE library in Sofia-SIP? >>>>> >>>>> Regards, >>>>> -- afshin >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> -- >>>> ________________________________________________________ >>>> | >>>> | >>>> | FATAL ERROR >>>> --- O X | >>>> |_______________________________________________________| >>>> | You have moved the mouse. >>>> | >>>> | Windows must be restarted for the changes to take effect. | >>>> | >>>> | >>>> ####################################/ >>>> >>>> >>>> Muhammad Shahzad >>>> ----------------------------------- >>>> CISCO Rich Media Communication Specialist (CRMCS) >>>> CISCO Certified Network Associate (CCNA) >>>> Cell: +92 334 422 40 88 >>>> MSN: shari_786pk at hotmail.com >>>> Email: shaheryarkh at googlemail.com >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> >>>> FreeSWITCH-users at 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 at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE: >>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> ________________________________________________________ >> | >> | >> | FATAL ERROR >> --- O X | >> |_______________________________________________________| >> | You have moved the mouse. >> | >> | Windows must be restarted for the changes to take effect. | >> | >> | >> ####################################/ >> >> >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> _______________________________________________ >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > ________________________________________________________ > | > | > | FATAL ERROR --- > O X | > |_______________________________________________________| > | You have moved the mouse. > | > | Windows must be restarted for the changes to take effect. | > | > | > ####################################/ > > > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/d2b55c86/attachment-0001.html From mattdfong at gmail.com Wed Sep 23 09:03:53 2009 From: mattdfong at gmail.com (Matthew Fong) Date: Wed, 23 Sep 2009 23:03:53 +0700 Subject: [Freeswitch-users] Call Files for a dialer engine In-Reply-To: <994476df9083131426fb1d69e257c93b.squirrel@correo.nodo50.org> References: <994476df9083131426fb1d69e257c93b.squirrel@correo.nodo50.org> Message-ID: <4256bf830909230903q710887cfx47cf33fd166b8849@mail.gmail.com> ESL is probably the way to go tho...if you want to build a dialer. The Dial Plans can get pretty advanced in FreeSWITCH...and if that is not enough you might consider using mod_perl or something of that sort. --matt Voice Broadcasting & Predictive Dialer based on FreeSWITCH On Wed, Sep 23, 2009 at 10:32 PM, Alberto Escudero wrote: > I am exploring the possibility of building a Dialer that emulates the > logic of Call Files in asterisk. > A CallerID catcher is creating CUSTOM events that I can store in a > database. I can trigger callbacks using ESL but I wonder what is the best > way/nicer/geekier to do something like outgoing calls in * > > /aep > > -- > Stopping junk mailers is good for the environment > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/fda8c512/attachment.html From quentusrex at gmail.com Wed Sep 23 09:08:27 2009 From: quentusrex at gmail.com (William King) Date: Wed, 23 Sep 2009 09:08:27 -0700 Subject: [Freeswitch-users] Status of ubuntu/debian packages. Message-ID: <4ABA47FB.2050100@gmail.com> Just to give everyone an update. There are working Ubuntu packages in a launchpad ppa. Debian users can add the ppa to their apt sources and build the package on your box. I'm currently using the packages on my home box and it is working great. Alright. I'm looking for people who want to use the packages. There are built packages for Ubuntu 8.04, 8.10, 9.04, and I'm working on 9.10 as well. I'm building two apt repos. One will have nightly builds and the other will be for tagged releases, plus any major bug fixes. Thanks to Frank, we now have everything split out into separate files for everything. This is to try to reduce the amount of stuff you 'have' to download by default. We have the en-us-callie sounds packaged at 8k, 16k, 32k, and 48k, we also have packaged the russian-elena and music on hold at the same qualities. If there are other languages, or voices I'd be more than happy to package them. I just need the 48k. Also we have separated the packages out so you can specify which mods you want, such as mod_perl, mod_python, mod_lua are all separate so you can install them if you want. nightlies: https://launchpad.net/~pbxbuntu-drivers/+archive/ppa tagged releases: https://launchpad.net/~freeswitch-drivers/+archive/ppa The tagged released packages will start with the 1.0.5 tagged release which some say should be coming out soon. Any one who would like to help out feel free to sign up with a launchpad account, and request to join the driver team of either or both ppa's. Once we have some people using the packages, and testing them, I have already talked to some of the ubuntu official package maintainers about what would need to be done to add freeswitch into the ubuntu multiverse repo. Any questions? -William King (quentusrex) From grae at digilord.net Wed Sep 23 09:16:28 2009 From: grae at digilord.net (digilord) Date: Wed, 23 Sep 2009 09:16:28 -0700 Subject: [Freeswitch-users] Automagic Phone Provisioning In-Reply-To: References: <1253719166.4738.7.camel@digilords-desktop.digilord.net> Message-ID: <1253722588.3596.2.camel@digilords-desktop.digilord.net> Brian, Is there code someplace that I can get that will help with the Polycom DHCPINFORM way? Thanks On Wed, 2009-09-23 at 10:46 -0500, Brian West wrote: > What you want is NOT possible the way you describe it. Snom does a > multicast PNP which lets you reply with a notify. Polycom does a > DHCPINFORM which lets you respond with a DHCPACK with additional > options. Aastra does MDNS which dictates where to go get the configs. > > /b > > On Sep 23, 2009, at 10:19 AM, digilord wrote: > > > Hello all, > > I know this is done and I think I figured out how to do it but I > > don't > > want to reinvent the wheel so here goes. I am looking for a program > > that will sit on the PBX. This program will intercept DHCP reply > > packets destined for phones, inject "option 66" into the packet and > > release it back onto the network. > > > > Some of you might be wondering why I want a program like this. > > Simple. > > Lazy clients. They don't want to mess with their network > > infrastructure > > to assist us with automated deployment of SIP devices. They also > > don't > > want 50-100 devices connecting to an off site server downloading > > 20-40MB > > of firmware on a reboot. > > > > The PBX is not hard coded with an IP address. It's DHCP. They were > > willing to allow the PBX on the network and assign it a static DHCP > > address. > > > > Is what I am looking for not possible? Does someone have a sensible > > solution that doesn't involve dropping the client (yes someone > > suggested > > that)? > > > > Thanks in advance for any help you can give. > > > > DigiLord > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/18f9c855/attachment.bin From grae at digilord.net Wed Sep 23 09:17:50 2009 From: grae at digilord.net (digilord) Date: Wed, 23 Sep 2009 09:17:50 -0700 Subject: [Freeswitch-users] Automagic Phone Provisioning In-Reply-To: <1253721106.18177.47.camel@dk-d820> References: <1253719166.4738.7.camel@digilords-desktop.digilord.net> <1253721106.18177.47.camel@dk-d820> Message-ID: <1253722670.3596.4.camel@digilords-desktop.digilord.net> I thought I had it figured out and was just missing one piece. Brian provided me with the missing info. My assumptions were wrong. Now I need to find software that will do what he described. On Wed, 2009-09-23 at 18:51 +0300, David Knell wrote: > I don't think it's trivially possible, unless you can stick the PBX > between the DHCP server and the rest of the network. The reason is that > DHCP reply packets are not broadcast, but sent back to the MAC address > of the originator, so your Ethernet switch won't even let your PBX see > the replies. > > Even if it could see them, add something to them and retransmit, the > client will almost certainly already have seen the original reply and > would be likely just to ignore the later one. > > Any reason why you can't just ask them to add that option to their > existing DHCP server? > > And how do you know it's done, and, if you've figured out how to do it, > could you share? > > --Dave > > > Hello all, > > I know this is done and I think I figured out how to do it but I don't > > want to reinvent the wheel so here goes. I am looking for a program > > that will sit on the PBX. This program will intercept DHCP reply > > packets destined for phones, inject "option 66" into the packet and > > release it back onto the network. > > > > Some of you might be wondering why I want a program like this. Simple. > > Lazy clients. They don't want to mess with their network infrastructure > > to assist us with automated deployment of SIP devices. They also don't > > want 50-100 devices connecting to an off site server downloading 20-40MB > > of firmware on a reboot. > > > > The PBX is not hard coded with an IP address. It's DHCP. They were > > willing to allow the PBX on the network and assign it a static DHCP > > address. > > > > Is what I am looking for not possible? Does someone have a sensible > > solution that doesn't involve dropping the client (yes someone suggested > > that)? > > > > Thanks in advance for any help you can give. > > > > DigiLord > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/c58d332c/attachment.bin From aep.lists at it46.se Wed Sep 23 09:22:40 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Wed, 23 Sep 2009 18:22:40 +0200 Subject: [Freeswitch-users] [Freeswitch-dev] Status of ubuntu/debian packages. In-Reply-To: <4ABA47FB.2050100@gmail.com> References: <4ABA47FB.2050100@gmail.com> Message-ID: Hi William, I will be very happy to test them, can you share the source and procedure to create the .debs? It will be also very good to find ways to have a cepstral package included *pending the licence* of course :) /aep -- Stopping junk mailers is good for the environment > Just to give everyone an update. There are working Ubuntu packages in a > launchpad ppa. Debian users can add the ppa to their apt sources and > build the package on your box. I'm currently using the packages on my > home box and it is working great. > > Alright. I'm looking for people who want to use the packages. There are > built packages for Ubuntu 8.04, 8.10, 9.04, and I'm working on 9.10 as > well. I'm building two apt repos. One will have nightly builds and the > other will be for tagged releases, plus any major bug fixes. > > Thanks to Frank, we now have everything split out into separate files > for everything. This is to try to reduce the amount of stuff you 'have' > to download by default. We have the en-us-callie sounds packaged at 8k, > 16k, 32k, and 48k, we also have packaged the russian-elena and music on > hold at the same qualities. If there are other languages, or voices I'd > be more than happy to package them. I just need the 48k. Also we have > separated the packages out so you can specify which mods you want, such > as mod_perl, mod_python, mod_lua are all separate so you can install > them if you want. > > nightlies: > https://launchpad.net/~pbxbuntu-drivers/+archive/ppa > > > tagged releases: > https://launchpad.net/~freeswitch-drivers/+archive/ppa > > > The tagged released packages will start with the 1.0.5 tagged release > which some say should be coming out soon. > > Any one who would like to help out feel free to sign up with a launchpad > account, and request to join the driver team of either or both ppa's. > > Once we have some people using the packages, and testing them, I have > already talked to some of the ubuntu official package maintainers about > what would need to be done to add freeswitch into the ubuntu multiverse > repo. > > Any questions? > > -William King (quentusrex) > > _______________________________________________ > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev > http://www.freeswitch.org > > From oseslija at gmail.com Wed Sep 23 09:26:19 2009 From: oseslija at gmail.com (Ognjen Seslija) Date: Wed, 23 Sep 2009 18:26:19 +0200 Subject: [Freeswitch-users] Automagic Phone Provisioning In-Reply-To: <1253722588.3596.2.camel@digilords-desktop.digilord.net> References: <1253719166.4738.7.camel@digilords-desktop.digilord.net> <1253722588.3596.2.camel@digilords-desktop.digilord.net> Message-ID: <4468a6770909230926k6409d33es4fadcdf721eab1a0@mail.gmail.com> Polycom responds to SIP server dhcp option (I think it's 120). I haven't still seen any other phone doing that. Ognjen On Wed, Sep 23, 2009 at 6:16 PM, digilord wrote: > Brian, > Is there code someplace that I can get that will help with the > Polycom > DHCPINFORM way? > > Thanks > > On Wed, 2009-09-23 at 10:46 -0500, Brian West wrote: > > What you want is NOT possible the way you describe it. Snom does a > > multicast PNP which lets you reply with a notify. Polycom does a > > DHCPINFORM which lets you respond with a DHCPACK with additional > > options. Aastra does MDNS which dictates where to go get the configs. > > > > /b > > > > On Sep 23, 2009, at 10:19 AM, digilord wrote: > > > > > Hello all, > > > I know this is done and I think I figured out how to do it but I > > > don't > > > want to reinvent the wheel so here goes. I am looking for a program > > > that will sit on the PBX. This program will intercept DHCP reply > > > packets destined for phones, inject "option 66" into the packet and > > > release it back onto the network. > > > > > > Some of you might be wondering why I want a program like this. > > > Simple. > > > Lazy clients. They don't want to mess with their network > > > infrastructure > > > to assist us with automated deployment of SIP devices. They also > > > don't > > > want 50-100 devices connecting to an off site server downloading > > > 20-40MB > > > of firmware on a reboot. > > > > > > The PBX is not hard coded with an IP address. It's DHCP. They were > > > willing to allow the PBX on the network and assign it a static DHCP > > > address. > > > > > > Is what I am looking for not possible? Does someone have a sensible > > > solution that doesn't involve dropping the client (yes someone > > > suggested > > > that)? > > > > > > Thanks in advance for any help you can give. > > > > > > DigiLord > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/0d411572/attachment.html From quentusrex at gmail.com Wed Sep 23 09:30:01 2009 From: quentusrex at gmail.com (William King) Date: Wed, 23 Sep 2009 09:30:01 -0700 Subject: [Freeswitch-users] [Freeswitch-dev] Status of ubuntu/debian packages. In-Reply-To: References: <4ABA47FB.2050100@gmail.com> Message-ID: <4ABA4D09.10206@gmail.com> I would be more than happy to share the code I use. Here is the git repo: http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/ If anyone wants git commit access just send me your ssh public key and I'll add it to the repo. As you can see Frank and I have been busy for the last week. -William King Alberto Escudero wrote: > Hi William, > > I will be very happy to test them, can you share the source and procedure > to create the .debs? > It will be also very good to find ways to have a cepstral package included > *pending the licence* of course :) > > /aep > > From msc at freeswitch.org Wed Sep 23 09:34:32 2009 From: msc at freeswitch.org (Michael Collins) Date: Wed, 23 Sep 2009 09:34:32 -0700 Subject: [Freeswitch-users] Mod_perl $session in not hangup In-Reply-To: <20090923060525.GA8185@jdc.jasonjgw.net> References: <25530646.post@talk.nabble.com> <191c3a030909220739p66502dc3yc207f791bb158f8c@mail.gmail.com> <7d79b3930909222233p11504f17jeee7b4f923fafea9@mail.gmail.com> <20090923060525.GA8185@jdc.jasonjgw.net> Message-ID: <87f2f3b90909230934ifd78cdau53fa83f4a634c2d4@mail.gmail.com> On Tue, Sep 22, 2009 at 11:05 PM, Jason White wrote: > lakshmanan ganapathy wrote: > > Thanks for your replay. I don't know what is latest trunk. Is it latest > > version? I'm using freeswitch 1.0.4. > > It's the latest version from the svn repository. Use svn checkout, then > compile it as documented on the wiki. > > If you're on Linux/Unix you can go into the freeswitch src directory and do "make current" which will do all the work for you. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/c27c7563/attachment.html From aep.lists at it46.se Wed Sep 23 10:26:50 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Wed, 23 Sep 2009 19:26:50 +0200 Subject: [Freeswitch-users] Call Files for a dialer engine In-Reply-To: <191c3a030909230849o351bdbaah3dd48c85db07c56b@mail.gmail.com> References: <994476df9083131426fb1d69e257c93b.squirrel@correo.nodo50.org> <191c3a030909230849o351bdbaah3dd48c85db07c56b@mail.gmail.com> Message-ID: Yes, sounds the best way to go. I assume that Unique-ID is the unique key to track the call via ESL Unique-ID: a984afd4-a865-11de-a5b4-fb5a867b002c and Answer-State: the variable to determine if the call is successful? Or should wait for the reason of CS_DESTROY message. I want to avoid to keep track of the whole state machine to know if a call has been completed successfully or not. /aep Unique-ID: 53f51090-a865-11de-a5b4-fb5a867b002c Call-Direction: inbound Presence-Call-Direction: inbound Answer-State: answered -- Stopping junk mailers is good for the environment > make an esl script that monitors a dir for new files, and push the > contents > into your same db? > > > On Wed, Sep 23, 2009 at 10:32 AM, Alberto Escudero > wrote: > >> I am exploring the possibility of building a Dialer that emulates the >> logic of Call Files in asterisk. >> A CallerID catcher is creating CUSTOM events that I can store in a >> database. I can trigger callbacks using ESL but I wonder what is the >> best >> way/nicer/geekier to do something like outgoing calls in * >> >> /aep >> >> -- >> Stopping junk mailers is good for the environment >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dwhite at olp.net Wed Sep 23 10:09:17 2009 From: dwhite at olp.net (Dan White) Date: Wed, 23 Sep 2009 12:09:17 -0500 Subject: [Freeswitch-users] Automagic Phone Provisioning In-Reply-To: <1253719166.4738.7.camel@digilords-desktop.digilord.net> References: <1253719166.4738.7.camel@digilords-desktop.digilord.net> Message-ID: <20090923170917.GA8729@dan.olp.net> There are several ways to accomplish this, with enough effort. The most straight forward approach is to run two DHCP servers on that network, with each of the DHCP servers ignoring the OUIs of devices they do not wish to manage. You'll need to configure two separate pools on the two servers (and turn off the authoritative option in ISC DHCP). But those pools could be in the same subnet if needed. Another approach might be to run a custom DHCP relay on the PBX which injects the appropriate option into the response; again, with the primary DHCP ignoring requests from your phones based on a class statement (e.g.). I've done something similar in an ISP environment before. I can send you my rather ugly PERL code if you'd like. On 23/09/09?08:19?-0700, digilord wrote: >Hello all, > I know this is done and I think I figured out how to do it but I don't >want to reinvent the wheel so here goes. I am looking for a program >that will sit on the PBX. This program will intercept DHCP reply >packets destined for phones, inject "option 66" into the packet and >release it back onto the network. > >Some of you might be wondering why I want a program like this. Simple. >Lazy clients. They don't want to mess with their network infrastructure >to assist us with automated deployment of SIP devices. They also don't >want 50-100 devices connecting to an off site server downloading 20-40MB >of firmware on a reboot. > >The PBX is not hard coded with an IP address. It's DHCP. They were >willing to allow the PBX on the network and assign it a static DHCP >address. > >Is what I am looking for not possible? Does someone have a sensible >solution that doesn't involve dropping the client (yes someone suggested >that)? > >Thanks in advance for any help you can give. > >DigiLord -- Dan White From grae at digilord.net Wed Sep 23 10:44:55 2009 From: grae at digilord.net (Daniel Morrigan) Date: Wed, 23 Sep 2009 10:44:55 -0700 Subject: [Freeswitch-users] Polycom MWI Forgetfulness Message-ID: <68755a9c0909231044ta70d850vab3ce7971992433d@mail.gmail.com> Hello, I have a number of Polycom phones that appear to forget that they have a message waiting then a little while later they seem to remember again. This was not the case with the same phones under Asterisk (Sorry to say that here but it's the only comparison I have). I have looked at a number of settings in sofia and can't seem to find one that makes any difference. Does anyone else have this issue? How did you solve it? DigiLord -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/2f022c7b/attachment.html From brian at freeswitch.org Wed Sep 23 10:46:11 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Sep 2009 12:46:11 -0500 Subject: [Freeswitch-users] Automagic Phone Provisioning In-Reply-To: <20090923170917.GA8729@dan.olp.net> References: <1253719166.4738.7.camel@digilords-desktop.digilord.net> <20090923170917.GA8729@dan.olp.net> Message-ID: <0E0D9F1B-D533-4152-9A55-3300626A5310@freeswitch.org> I have perl code for both snom-pnp and polycom dhcp inform but my wish list looks attractive ;) /b On Sep 23, 2009, at 12:09 PM, Dan White wrote: > There are several ways to accomplish this, with enough effort. > > The most straight forward approach is to run two DHCP servers on that > network, with each of the DHCP servers ignoring the OUIs of devices > they do > not wish to manage. > > You'll need to configure two separate pools on the two > servers (and turn off the authoritative option in ISC DHCP). But those > pools could be in the same subnet if needed. > > Another approach might be to run a custom DHCP relay on the PBX which > injects the appropriate option into the response; again, with the > primary > DHCP ignoring requests from your phones based on a class statement > (e.g.). > I've done something similar in an ISP environment before. I can send > you > my rather ugly PERL code if you'd like. From frank at carmickle.com Wed Sep 23 11:09:07 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 23 Sep 2009 14:09:07 -0400 Subject: [Freeswitch-users] [Freeswitch-dev] Status of ubuntu/debian packages. In-Reply-To: References: <4ABA47FB.2050100@gmail.com> Message-ID: <20090923180907.GO30343@base.carmickle.com> Hello Alberto On Wed, Sep 23, Alberto Escudero wrote: > Hi William, > > I will be very happy to test them, can you share the source and procedure > to create the .debs? > It will be also very good to find ways to have a cepstral package included > *pending the licence* of course :) Mario Lang has a pretty neat way of building these in the gnome-speech package. On installation of gnome-speech-swift it builds the support against the installed cepstral. This is how I plan on dealing with this. Unfortunately there are a number of other items that need to be fixed before we add this. If you are willing to help it would be greatly appreciated. --FC From diego.viola at gmail.com Wed Sep 23 11:10:54 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 23 Sep 2009 18:10:54 +0000 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <0f9b01ca3c57$c789ab30$569d0190$@com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <249BB60C-79F6-4A3A-8E3B-0E821D4D9046@freeswitch.org> <0f3c01ca3c48$dc802d20$95808760$@com> <0f9b01ca3c57$c789ab30$569d0190$@com> Message-ID: <86a32abc0909231110x2356c3b5y8f8746a64f0849eb@mail.gmail.com> Should I delete the directory "default" and default.xml when I copy default to foo.org and bar.org etc? Diego On Wed, Sep 23, 2009 at 2:11 PM, Peder wrote: > That's a good idea. I thought about using a DB, but I was going to have to > use a lua script to look stuff up. I didn't think about easyroute or lcr. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rupa > Schomaker > Sent: Wednesday, September 23, 2009 8:47 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Multitenancy > > Use something like mod_easyroute to consult a database of DIDs. If > you host the DID it'll give you a route to dial out. If not, route > out your gateway. > > Or load your lcr tables with your own DIDs. Consult mod_lcr and use > it's dialstring. It'll prefer longer prefix matches, so you will > always win with your own customers. > > On Wed, Sep 23, 2009 at 7:24 AM, Peder wrote: > > So if you do this, how do you call between contexts? Say you have 100 > > tenants on one box each with their own domain and they are all 4 digit > for > > local dialing. If they call a 10 digit number like they are calling > > outbound and it is another tenant on the same box, they don?t want to go > out > > and back in, they just want to be bridged over to the other context. I > > imagine you need to create a file for each context and add every other > > context and did to it to check if they are on the same box, otherwise > dial > > outbound to the PSTN. That doesn?t scale very well though since if there > > are 100 tenants, and you add another, you need to modify the other 100 > > contexts to add the new DIDs. Or is there some built in way to do this > > easily? > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Brian > > West > > Sent: Wednesday, September 23, 2009 12:18 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Multitenancy > > > > > > > > Then setup two domains in your directory and setup proper DNS its really > > just that simple. > > > > > > > > /b > > > > > > > > On Sep 23, 2009, at 12:11 AM, Diego Viola wrote: > > > > I want user 1000-1010 to belong to foo.org and 2000-2020 to belong > > to bar.org, and I want both of those domains to have their own > > dialplan/context. > > > > > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > -- > -Rupa > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/5e2b5bf6/attachment-0001.html From brian at freeswitch.org Wed Sep 23 11:15:23 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Sep 2009 13:15:23 -0500 Subject: [Freeswitch-users] Polycom MWI Forgetfulness In-Reply-To: <68755a9c0909231044ta70d850vab3ce7971992433d@mail.gmail.com> References: <68755a9c0909231044ta70d850vab3ce7971992433d@mail.gmail.com> Message-ID: <437FA030-C96F-4E46-8727-E3F4FA04D6EF@freeswitch.org> I have never seen this behavior. What do you have your MWI callback method setup as? /b On Sep 23, 2009, at 12:44 PM, Daniel Morrigan wrote: > Hello, > I have a number of Polycom phones that appear to forget that > they have a message waiting then a little while later they seem to > remember again. This was not the case with the same phones under > Asterisk (Sorry to say that here but it's the only comparison I have). > > I have looked at a number of settings in sofia and can't seem to > find one that makes any difference. > > Does anyone else have this issue? How did you solve it? > > DigiLord From brian at freeswitch.org Wed Sep 23 11:16:14 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Sep 2009 13:16:14 -0500 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909231110x2356c3b5y8f8746a64f0849eb@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <249BB60C-79F6-4A3A-8E3B-0E821D4D9046@freeswitch.org> <0f3c01ca3c48$dc802d20$95808760$@com> <0f9b01ca3c57$c789ab30$569d0190$@com> <86a32abc0909231110x2356c3b5y8f8746a64f0849eb@mail.gmail.com> Message-ID: <7AE4706B-2D2A-4ED7-B58F-F933726AF1E4@freeswitch.org> Thats up to you :P /b On Sep 23, 2009, at 1:10 PM, Diego Viola wrote: > Should I delete the directory "default" and default.xml when I copy > default to foo.org and bar.org etc? > > Diego -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/3547145f/attachment.html From codecomplete at free.fr Wed Sep 23 11:25:44 2009 From: codecomplete at free.fr (Fred-145) Date: Wed, 23 Sep 2009 11:25:44 -0700 (PDT) Subject: [Freeswitch-users] Porting Freeswitch to ARM? Message-ID: <25531239.post@talk.nabble.com> Hello I don't have the technical expertise to tell, so here goes: Unless mistaken, Freeswitch is written in C and/or C++, so I guess it's not linked to the x86 instruction set. Is so, would it be a lot of work of porting it so it runs on ARM processors? It'd be cool because it could run on tiny platforms like the http://en.wikipedia.org/wiki/SheevaPlug Sheeva Plug . Or maybe parts of it are written in assembly, so it's a lot of work? Thank you. -- View this message in context: http://www.nabble.com/Porting-Freeswitch-to-ARM--tp25531239p25531239.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From mike at jerris.com Wed Sep 23 11:27:31 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 14:27:31 -0400 Subject: [Freeswitch-users] Call Files for a dialer engine In-Reply-To: References: <994476df9083131426fb1d69e257c93b.squirrel@correo.nodo50.org> <191c3a030909230849o351bdbaah3dd48c85db07c56b@mail.gmail.com> Message-ID: You can use Answer-State, CS_DESTROY won't happen until the call is over. On Sep 23, 2009, at 1:26 PM, Alberto Escudero wrote: > Yes, sounds the best way to go. > > I assume that Unique-ID is the unique key to track the call via ESL > Unique-ID: a984afd4-a865-11de-a5b4-fb5a867b002c > > and Answer-State: the variable to determine if the call is successful? > > Or should wait for the reason of CS_DESTROY message. I want to avoid > to > keep track of the whole state machine to know if a call has been > completed > successfully or not. > > /aep > > > > > Unique-ID: 53f51090-a865-11de-a5b4-fb5a867b002c > Call-Direction: inbound > Presence-Call-Direction: inbound > Answer-State: answered > > -- > Stopping junk mailers is good for the environment > >> make an esl script that monitors a dir for new files, and push the >> contents >> into your same db? >> >> >> On Wed, Sep 23, 2009 at 10:32 AM, Alberto Escudero >> wrote: >> >>> I am exploring the possibility of building a Dialer that emulates >>> the >>> logic of Call Files in asterisk. >>> A CallerID catcher is creating CUSTOM events that I can store in a >>> database. I can trigger callbacks using ESL but I wonder what is the >>> best >>> way/nicer/geekier to do something like outgoing calls in * From sranil at gmail.com Wed Sep 23 11:37:41 2009 From: sranil at gmail.com (Anil Kumar S. R.) Date: Thu, 24 Sep 2009 00:07:41 +0530 Subject: [Freeswitch-users] User Creation with DB in Freeswitch In-Reply-To: <347D3032-F778-46F4-AECE-94DECD27E849@freeswitch.org> References: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> <566E110B-72CC-428E-AEE8-94AB600587B0@enterux.com> <347D3032-F778-46F4-AECE-94DECD27E849@freeswitch.org> Message-ID: <1b2118200909231137w79ea33afq6050133fb50b32a0@mail.gmail.com> I didn't get much help for my problem with XML CURL. What I meant to say is, suppose I want to have some 10000 users on freeswitch. Do we have to create some many xml files in the directory or is there some way in which the users can be put in the db ? Also. my another question is, what is the command by which we can get the number of users that are currently registered on the freeswitch? Thanks, Anil 2009/9/21 Brian West > You can't put the users directly into a db with FreeSWITCH you'll have to > serve up the XML document via XML CURL or write your own module to do so via > the module interfaces provided. > /b > > On Sep 20, 2009, at 10:44 PM, Mitul Limbani wrote: > > Yes use odbc in fs > > Thanks & Regards,Mitul Limbani, > Founder & CEO, > Enterux Solutions Pvt. Ltd., > The Enterprise Linux Company (r), > http://www.enterux.com > http://www.entVoice.com > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anil Kumar S. R. http://sranil.googlepages.com/ "The best way to succeed in this world is to act on the advice you give to others." -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/b02bcc37/attachment.html From mike at jerris.com Wed Sep 23 11:40:59 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 14:40:59 -0400 Subject: [Freeswitch-users] Porting Freeswitch to ARM? In-Reply-To: <25531239.post@talk.nabble.com> References: <25531239.post@talk.nabble.com> Message-ID: <737802AA-46AB-4327-8F26-E0F7D3E6D921@jerris.com> http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Cross_Compiling_for_ARM_on_Linux Mike On Sep 23, 2009, at 2:25 PM, Fred-145 wrote: > > Hello > > I don't have the technical expertise to tell, so here goes: Unless > mistaken, > Freeswitch is written in C and/or C++, so I guess it's not linked to > the x86 > instruction set. > > Is so, would it be a lot of work of porting it so it runs on ARM > processors? > It'd be cool because it could run on tiny platforms like the > http://en.wikipedia.org/wiki/SheevaPlug Sheeva Plug . > > Or maybe parts of it are written in assembly, so it's a lot of work? From mitul at enterux.com Wed Sep 23 11:41:47 2009 From: mitul at enterux.com (Mitul Limbani) Date: Thu, 24 Sep 2009 00:11:47 +0530 Subject: [Freeswitch-users] Automagic Phone Provisioning In-Reply-To: <0E0D9F1B-D533-4152-9A55-3300626A5310@freeswitch.org> References: <1253719166.4738.7.camel@digilords-desktop.digilord.net> <20090923170917.GA8729@dan.olp.net> <0E0D9F1B-D533-4152-9A55-3300626A5310@freeswitch.org> Message-ID: <8F8EAECD-A698-4E9B-9D1E-2CA5B9DE172B@enterux.com> Hey Brian, On 23-Sep-2009, at 11:16 PM, Brian West wrote: > I have perl code for both snom-pnp and polycom dhcp inform but my wish > list looks attractive ;) > Aahan so is it (wish list) documented anywhere? Lol Mitul Limbani Enterux Solutions Pvt. Ltd. www. Enterux.com From brian at freeswitch.org Wed Sep 23 11:45:03 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Sep 2009 13:45:03 -0500 Subject: [Freeswitch-users] Automagic Phone Provisioning In-Reply-To: <8F8EAECD-A698-4E9B-9D1E-2CA5B9DE172B@enterux.com> References: <1253719166.4738.7.camel@digilords-desktop.digilord.net> <20090923170917.GA8729@dan.olp.net> <0E0D9F1B-D533-4152-9A55-3300626A5310@freeswitch.org> <8F8EAECD-A698-4E9B-9D1E-2CA5B9DE172B@enterux.com> Message-ID: <0291851F-1FB3-422D-BF3B-A5D2D9522D07@freeswitch.org> its in the FAQ first question ;) /b On Sep 23, 2009, at 1:41 PM, Mitul Limbani wrote: > Aahan so is it (wish list) documented anywhere? Lol From diego.viola at gmail.com Wed Sep 23 11:52:23 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 23 Sep 2009 18:52:23 +0000 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <7AE4706B-2D2A-4ED7-B58F-F933726AF1E4@freeswitch.org> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <249BB60C-79F6-4A3A-8E3B-0E821D4D9046@freeswitch.org> <0f3c01ca3c48$dc802d20$95808760$@com> <0f9b01ca3c57$c789ab30$569d0190$@com> <86a32abc0909231110x2356c3b5y8f8746a64f0849eb@mail.gmail.com> <7AE4706B-2D2A-4ED7-B58F-F933726AF1E4@freeswitch.org> Message-ID: <86a32abc0909231152t2fdeea11s25f7c13e29459678@mail.gmail.com> I was having some issues with DNS, I tried to register with the new directory and domain but I got "can't find user" until I commented force-register-domain and force-register-db-domain from the profile. Thanks for the tip Brian :). Diego On Wed, Sep 23, 2009 at 6:16 PM, Brian West wrote: > Thats up to you :P > /b > > On Sep 23, 2009, at 1:10 PM, Diego Viola wrote: > > Should I delete the directory "default" and default.xml when I copy default > to foo.org and bar.org etc? > > Diego > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/c20426d9/attachment-0001.html From mike at jerris.com Wed Sep 23 12:08:58 2009 From: mike at jerris.com (Michael Jerris) Date: Wed, 23 Sep 2009 15:08:58 -0400 Subject: [Freeswitch-users] User Creation with DB in Freeswitch In-Reply-To: <1b2118200909231137w79ea33afq6050133fb50b32a0@mail.gmail.com> References: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> <566E110B-72CC-428E-AEE8-94AB600587B0@enterux.com> <347D3032-F778-46F4-AECE-94DECD27E849@freeswitch.org> <1b2118200909231137w79ea33afq6050133fb50b32a0@mail.gmail.com> Message-ID: <14AF3361-958D-40ED-A70A-FD9124A112C9@jerris.com> There are tons of details on this at http://wiki.freeswitch.org/wiki/Mod_xml_curl Are you having an issue? Mike On Sep 23, 2009, at 2:37 PM, Anil Kumar S. R. wrote: > I didn't get much help for my problem with XML CURL. What I meant to > say is, suppose I want to have some 10000 users on freeswitch. Do we > have to create some many xml files in the directory or is there some > way in which the users can be put in the db ? > > Also. my another question is, what is the command by which we can > get the number of users that are currently registered on the > freeswitch? > > Thanks, > Anil > > 2009/9/21 Brian West > You can't put the users directly into a db with FreeSWITCH you'll > have to serve up the XML document via XML CURL or write your own > module to do so via the module interfaces provided. > > /b > > On Sep 20, 2009, at 10:44 PM, Mitul Limbani wrote: > >> Yes use odbc in fs -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/87ce1f41/attachment.html From diego.viola at gmail.com Wed Sep 23 12:40:12 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 23 Sep 2009 19:40:12 +0000 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909231152t2fdeea11s25f7c13e29459678@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <249BB60C-79F6-4A3A-8E3B-0E821D4D9046@freeswitch.org> <0f3c01ca3c48$dc802d20$95808760$@com> <0f9b01ca3c57$c789ab30$569d0190$@com> <86a32abc0909231110x2356c3b5y8f8746a64f0849eb@mail.gmail.com> <7AE4706B-2D2A-4ED7-B58F-F933726AF1E4@freeswitch.org> <86a32abc0909231152t2fdeea11s25f7c13e29459678@mail.gmail.com> Message-ID: <86a32abc0909231240q5fce2bd9ya6d0e12c049dce8e@mail.gmail.com> Ok I have configured the two domains with their own directory and I can register fine with them now. But I need to configure two different dialplans with their own profiles. How do I tell a specific domain to use a specific profile/dialplan? Thanks, Diego On Wed, Sep 23, 2009 at 6:52 PM, Diego Viola wrote: > I was having some issues with DNS, I tried to register with the new > directory and domain but I got "can't find user" until I commented > force-register-domain and force-register-db-domain from the profile. > > Thanks for the tip Brian :). > > Diego > > On Wed, Sep 23, 2009 at 6:16 PM, Brian West wrote: > >> Thats up to you :P >> /b >> >> On Sep 23, 2009, at 1:10 PM, Diego Viola wrote: >> >> Should I delete the directory "default" and default.xml when I copy >> default to foo.org and bar.org etc? >> >> Diego >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/549c6f18/attachment.html From diego.viola at gmail.com Wed Sep 23 12:42:48 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 23 Sep 2009 19:42:48 +0000 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909231240q5fce2bd9ya6d0e12c049dce8e@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <249BB60C-79F6-4A3A-8E3B-0E821D4D9046@freeswitch.org> <0f3c01ca3c48$dc802d20$95808760$@com> <0f9b01ca3c57$c789ab30$569d0190$@com> <86a32abc0909231110x2356c3b5y8f8746a64f0849eb@mail.gmail.com> <7AE4706B-2D2A-4ED7-B58F-F933726AF1E4@freeswitch.org> <86a32abc0909231152t2fdeea11s25f7c13e29459678@mail.gmail.com> <86a32abc0909231240q5fce2bd9ya6d0e12c049dce8e@mail.gmail.com> Message-ID: <86a32abc0909231242v580b381eld776adeb351ce8d1@mail.gmail.com> s/directory/directories/ Should I use "context" for that? On Wed, Sep 23, 2009 at 7:40 PM, Diego Viola wrote: > Ok I have configured the two domains with their own directory and I can > register fine with them now. > > But I need to configure two different dialplans with their own profiles. > > How do I tell a specific domain to use a specific profile/dialplan? > > Thanks, > > Diego > > > On Wed, Sep 23, 2009 at 6:52 PM, Diego Viola wrote: > >> I was having some issues with DNS, I tried to register with the new >> directory and domain but I got "can't find user" until I commented >> force-register-domain and force-register-db-domain from the profile. >> >> Thanks for the tip Brian :). >> >> Diego >> >> On Wed, Sep 23, 2009 at 6:16 PM, Brian West wrote: >> >>> Thats up to you :P >>> /b >>> >>> On Sep 23, 2009, at 1:10 PM, Diego Viola wrote: >>> >>> Should I delete the directory "default" and default.xml when I copy >>> default to foo.org and bar.org etc? >>> >>> Diego >>> >>> >>> >>> _______________________________________________ >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/8d08fbce/attachment.html From diego.viola at gmail.com Wed Sep 23 12:48:11 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 23 Sep 2009 19:48:11 +0000 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909231242v580b381eld776adeb351ce8d1@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <249BB60C-79F6-4A3A-8E3B-0E821D4D9046@freeswitch.org> <0f3c01ca3c48$dc802d20$95808760$@com> <0f9b01ca3c57$c789ab30$569d0190$@com> <86a32abc0909231110x2356c3b5y8f8746a64f0849eb@mail.gmail.com> <7AE4706B-2D2A-4ED7-B58F-F933726AF1E4@freeswitch.org> <86a32abc0909231152t2fdeea11s25f7c13e29459678@mail.gmail.com> <86a32abc0909231240q5fce2bd9ya6d0e12c049dce8e@mail.gmail.com> <86a32abc0909231242v580b381eld776adeb351ce8d1@mail.gmail.com> Message-ID: <86a32abc0909231248w1822610fw3f9aaf6efda8d552@mail.gmail.com> Do I specific the context as a per-user thing, can I specific the context as a per-domain way? Diego On Wed, Sep 23, 2009 at 7:42 PM, Diego Viola wrote: > s/directory/directories/ > > Should I use "context" for that? > > > On Wed, Sep 23, 2009 at 7:40 PM, Diego Viola wrote: > >> Ok I have configured the two domains with their own directory and I can >> register fine with them now. >> >> But I need to configure two different dialplans with their own profiles. >> >> How do I tell a specific domain to use a specific profile/dialplan? >> >> Thanks, >> >> Diego >> >> >> On Wed, Sep 23, 2009 at 6:52 PM, Diego Viola wrote: >> >>> I was having some issues with DNS, I tried to register with the new >>> directory and domain but I got "can't find user" until I commented >>> force-register-domain and force-register-db-domain from the profile. >>> >>> Thanks for the tip Brian :). >>> >>> Diego >>> >>> On Wed, Sep 23, 2009 at 6:16 PM, Brian West wrote: >>> >>>> Thats up to you :P >>>> /b >>>> >>>> On Sep 23, 2009, at 1:10 PM, Diego Viola wrote: >>>> >>>> Should I delete the directory "default" and default.xml when I copy >>>> default to foo.org and bar.org etc? >>>> >>>> Diego >>>> >>>> >>>> >>>> _______________________________________________ >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/0e4c4502/attachment.html From diego.viola at gmail.com Wed Sep 23 12:48:52 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 23 Sep 2009 19:48:52 +0000 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909231248w1822610fw3f9aaf6efda8d552@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <0f3c01ca3c48$dc802d20$95808760$@com> <0f9b01ca3c57$c789ab30$569d0190$@com> <86a32abc0909231110x2356c3b5y8f8746a64f0849eb@mail.gmail.com> <7AE4706B-2D2A-4ED7-B58F-F933726AF1E4@freeswitch.org> <86a32abc0909231152t2fdeea11s25f7c13e29459678@mail.gmail.com> <86a32abc0909231240q5fce2bd9ya6d0e12c049dce8e@mail.gmail.com> <86a32abc0909231242v580b381eld776adeb351ce8d1@mail.gmail.com> <86a32abc0909231248w1822610fw3f9aaf6efda8d552@mail.gmail.com> Message-ID: <86a32abc0909231248m30d8f72cnbcdb06a328b2cfbe@mail.gmail.com> I prefer to specify the context as a per-domain so it affects all the users on the domain directly... On Wed, Sep 23, 2009 at 7:48 PM, Diego Viola wrote: > Do I specific the context as a per-user thing, can I specific the context > as a per-domain way? > > Diego > > > On Wed, Sep 23, 2009 at 7:42 PM, Diego Viola wrote: > >> s/directory/directories/ >> >> Should I use "context" for that? >> >> >> On Wed, Sep 23, 2009 at 7:40 PM, Diego Viola wrote: >> >>> Ok I have configured the two domains with their own directory and I can >>> register fine with them now. >>> >>> But I need to configure two different dialplans with their own profiles. >>> >>> How do I tell a specific domain to use a specific profile/dialplan? >>> >>> Thanks, >>> >>> Diego >>> >>> >>> On Wed, Sep 23, 2009 at 6:52 PM, Diego Viola wrote: >>> >>>> I was having some issues with DNS, I tried to register with the new >>>> directory and domain but I got "can't find user" until I commented >>>> force-register-domain and force-register-db-domain from the profile. >>>> >>>> Thanks for the tip Brian :). >>>> >>>> Diego >>>> >>>> On Wed, Sep 23, 2009 at 6:16 PM, Brian West wrote: >>>> >>>>> Thats up to you :P >>>>> /b >>>>> >>>>> On Sep 23, 2009, at 1:10 PM, Diego Viola wrote: >>>>> >>>>> Should I delete the directory "default" and default.xml when I copy >>>>> default to foo.org and bar.org etc? >>>>> >>>>> Diego >>>>> >>>>> >>>>> >>>>> _______________________________________________ >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/6e97c9a4/attachment-0001.html From dmitry.bely at gmail.com Wed Sep 23 12:51:07 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Wed, 23 Sep 2009 23:51:07 +0400 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <4ABA47FB.2050100@gmail.com> References: <4ABA47FB.2050100@gmail.com> Message-ID: <90823c940909231251k5f85e3dambb7506bdc62e62ac@mail.gmail.com> On Wed, Sep 23, 2009 at 8:08 PM, William King wrote: > Just to give everyone an update. There are working Ubuntu packages in a > launchpad ppa. Debian users can add the ppa to their apt sources and > build the package on your box. I'm currently using the packages on my > home box and it is working great. > > Alright. I'm looking for people who want to use the packages. There are > built packages for Ubuntu 8.04, 8.10, 9.04, and I'm working on 9.10 as > well. ?I'm building two apt repos. One will have nightly builds and the > other will be for tagged releases, plus any major bug fixes. > > Thanks to Frank, we now have everything split out into separate files > for everything. This is to try to reduce the amount of stuff you 'have' > to download by default. We have the en-us-callie sounds packaged at 8k, > 16k, 32k, and 48k, we also have packaged the russian-elena and music on > hold at the same qualities. If there are other languages, or voices I'd > be more than happy to package them. I just need the 48k. Also we have > separated the packages out so you can specify which mods you want, such > as mod_perl, mod_python, mod_lua are all separate so you can install > them if you want. > > nightlies: > https://launchpad.net/~pbxbuntu-drivers/+archive/ppa > > > tagged releases: > https://launchpad.net/~freeswitch-drivers/+archive/ppa > > > The tagged released packages will start with the 1.0.5 tagged release > which some say should be coming out soon. > > Any one who would like to help out feel free to sign up with a launchpad > account, and request to join the driver team of either or both ppa's. > > Once we have some people using the packages, and testing them, I have > already talked to some of the ubuntu official package maintainers about > what would need to be done to add freeswitch into the ubuntu multiverse > repo. > > Any questions? Can you enable mod_skypiax in your debian package? - Dmitry Bely From brian at freeswitch.org Wed Sep 23 13:08:50 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Sep 2009 15:08:50 -0500 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909231240q5fce2bd9ya6d0e12c049dce8e@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <86a32abc0909222211t2993e05bycf479d835b4bae55@mail.gmail.com> <249BB60C-79F6-4A3A-8E3B-0E821D4D9046@freeswitch.org> <0f3c01ca3c48$dc802d20$95808760$@com> <0f9b01ca3c57$c789ab30$569d0190$@com> <86a32abc0909231110x2356c3b5y8f8746a64f0849eb@mail.gmail.com> <7AE4706B-2D2A-4ED7-B58F-F933726AF1E4@freeswitch.org> <86a32abc0909231152t2fdeea11s25f7c13e29459678@mail.gmail.com> <86a32abc0909231240q5fce2bd9ya6d0e12c049dce8e@mail.gmail.com> Message-ID: <419395A5-3A71-478B-83A9-EEA7A6EAE0BE@freeswitch.org> You really don't have too but you can set a user_context on each user in the domain or on the domain level to specify the domain's or user's context. /b On Sep 23, 2009, at 2:40 PM, Diego Viola wrote: > Ok I have configured the two domains with their own directory and I > can register fine with them now. > > But I need to configure two different dialplans with their own > profiles. > > How do I tell a specific domain to use a specific profile/dialplan? > > Thanks, > > Diego From brian at freeswitch.org Wed Sep 23 13:09:46 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Sep 2009 15:09:46 -0500 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <86a32abc0909231248m30d8f72cnbcdb06a328b2cfbe@mail.gmail.com> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <0f3c01ca3c48$dc802d20$95808760$@com> <0f9b01ca3c57$c789ab30$569d0190$@com> <86a32abc0909231110x2356c3b5y8f8746a64f0849eb@mail.gmail.com> <7AE4706B-2D2A-4ED7-B58F-F933726AF1E4@freeswitch.org> <86a32abc0909231152t2fdeea11s25f7c13e29459678@mail.gmail.com> <86a32abc0909231240q5fce2bd9ya6d0e12c049dce8e@mail.gmail.com> <86a32abc0909231242v580b381eld776adeb351ce8d1@mail.gmail.com> <86a32abc0909231248w1822610fw3f9aaf6efda8d552@mail.gmail.com> <86a32abc0909231248m30d8f72cnbcdb06a328b2cfbe@mail.gmail.com> Message-ID: <82D6FACD-37D4-4B1D-8820-66E8740769CE@freeswitch.org> Can you next time pause a few moments... Think about what you're sending and send ONE email with your questions? This 10 emails from you replying to yourself things looks like you're a crazy man! :P /b PS: ask on IRC or mailing list NOT BOTH please. On Sep 23, 2009, at 2:48 PM, Diego Viola wrote: > I prefer to specify the context as a per-domain so it affects all > the users on the domain directly... > > On Wed, Sep 23, 2009 at 7:48 PM, Diego Viola > wrote: > Do I specific the context as a per-user thing, can I specific the > context as a per-domain way? > > Diego > > > On Wed, Sep 23, 2009 at 7:42 PM, Diego Viola > wrote: > s/directory/directories/ > > Should I use "context" for that? > > > On Wed, Sep 23, 2009 at 7:40 PM, Diego Viola > wrote: > Ok I have configured the two domains with their own directory and I > can register fine with them now. > > But I need to configure two different dialplans with their own > profiles. > > How do I tell a specific domain to use a specific profile/dialplan? > > Thanks, > > Diego > > > On Wed, Sep 23, 2009 at 6:52 PM, Diego Viola > wrote: > I was having some issues with DNS, I tried to register with the new > directory and domain but I got "can't find user" until I commented > force-register-domain and force-register-db-domain from the profile. > > Thanks for the tip Brian :). > > Diego > > On Wed, Sep 23, 2009 at 6:16 PM, Brian West > wrote: > Thats up to you :P > > /b > > On Sep 23, 2009, at 1:10 PM, Diego Viola wrote: > >> Should I delete the directory "default" and default.xml when I copy >> default to foo.org and bar.org etc? >> >> Diego > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/fed91574/attachment.html From dmitry.bely at gmail.com Wed Sep 23 13:14:47 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Thu, 24 Sep 2009 00:14:47 +0400 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <4ABA47FB.2050100@gmail.com> References: <4ABA47FB.2050100@gmail.com> Message-ID: <90823c940909231314j3c03b8e9u45d3104d403a92aa@mail.gmail.com> On Wed, Sep 23, 2009 at 8:08 PM, William King wrote: (..) > Any questions? Another problem (on Ubuntu jaunty): Setting up freeswitch (1.0.4+repack6-0ubuntu14925.1) ... adduser: The --group, --ingroup, and --gid options are mutually exclusive. addgroup: The user `freeswitch' does not exist. chown: invalid user: `freeswitch:freeswitch' - Dmitry Bely From diego.viola at gmail.com Wed Sep 23 13:22:33 2009 From: diego.viola at gmail.com (Diego Viola) Date: Wed, 23 Sep 2009 20:22:33 +0000 Subject: [Freeswitch-users] Multitenancy In-Reply-To: <82D6FACD-37D4-4B1D-8820-66E8740769CE@freeswitch.org> References: <86a32abc0909222157y4c841093j8ecd3aec0a4d30dd@mail.gmail.com> <0f9b01ca3c57$c789ab30$569d0190$@com> <86a32abc0909231110x2356c3b5y8f8746a64f0849eb@mail.gmail.com> <7AE4706B-2D2A-4ED7-B58F-F933726AF1E4@freeswitch.org> <86a32abc0909231152t2fdeea11s25f7c13e29459678@mail.gmail.com> <86a32abc0909231240q5fce2bd9ya6d0e12c049dce8e@mail.gmail.com> <86a32abc0909231242v580b381eld776adeb351ce8d1@mail.gmail.com> <86a32abc0909231248w1822610fw3f9aaf6efda8d552@mail.gmail.com> <86a32abc0909231248m30d8f72cnbcdb06a328b2cfbe@mail.gmail.com> <82D6FACD-37D4-4B1D-8820-66E8740769CE@freeswitch.org> Message-ID: <86a32abc0909231322q44a851di9c8e00d1534b9728@mail.gmail.com> Ok, sorry for that and thanks for the help :). Diego On Wed, Sep 23, 2009 at 8:09 PM, Brian West wrote: > Can you next time pause a few moments... Think about what you're sending > and send ONE email with your questions? This 10 emails from you replying to > yourself things looks like you're a crazy man! :P > /b > PS: ask on IRC or mailing list NOT BOTH please. > > On Sep 23, 2009, at 2:48 PM, Diego Viola wrote: > > I prefer to specify the context as a per-domain so it affects all the users > on the domain directly... > > On Wed, Sep 23, 2009 at 7:48 PM, Diego Viola wrote: > >> Do I specific the context as a per-user thing, can I specific the context >> as a per-domain way? >> >> Diego >> >> >> On Wed, Sep 23, 2009 at 7:42 PM, Diego Viola wrote: >> >>> s/directory/directories/ >>> >>> Should I use "context" for that? >>> >>> >>> On Wed, Sep 23, 2009 at 7:40 PM, Diego Viola wrote: >>> >>>> Ok I have configured the two domains with their own directory and I can >>>> register fine with them now. >>>> >>>> But I need to configure two different dialplans with their own profiles. >>>> >>>> How do I tell a specific domain to use a specific profile/dialplan? >>>> >>>> Thanks, >>>> >>>> Diego >>>> >>>> >>>> On Wed, Sep 23, 2009 at 6:52 PM, Diego Viola wrote: >>>> >>>>> I was having some issues with DNS, I tried to register with the new >>>>> directory and domain but I got "can't find user" until I commented >>>>> force-register-domain and force-register-db-domain from the profile. >>>>> >>>>> Thanks for the tip Brian :). >>>>> >>>>> Diego >>>>> >>>>> On Wed, Sep 23, 2009 at 6:16 PM, Brian West wrote: >>>>> >>>>>> Thats up to you :P >>>>>> /b >>>>>> >>>>>> On Sep 23, 2009, at 1:10 PM, Diego Viola wrote: >>>>>> >>>>>> Should I delete the directory "default" and default.xml when I copy >>>>>> default to foo.org and bar.org etc? >>>>>> >>>>>> Diego >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/7d3c2db0/attachment-0001.html From frank at carmickle.com Wed Sep 23 13:25:25 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 23 Sep 2009 16:25:25 -0400 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <90823c940909231251k5f85e3dambb7506bdc62e62ac@mail.gmail.com> References: <4ABA47FB.2050100@gmail.com> <90823c940909231251k5f85e3dambb7506bdc62e62ac@mail.gmail.com> Message-ID: <20090923202524.GR30343@base.carmickle.com> On Wed, Sep 23, Dmitry Bely wrote: > Can you enable mod_skypiax in your debian package? We will be enabling as much as we can cleanly build on debian/ubuntu. There will be a lot more to come. We will be breaking the mods and end points in to different packages so that you can install what you like. If you have something you would like to see in the package let us know. Also patches are welcome. --FC From frank at carmickle.com Wed Sep 23 13:42:41 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 23 Sep 2009 16:42:41 -0400 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <90823c940909231314j3c03b8e9u45d3104d403a92aa@mail.gmail.com> References: <4ABA47FB.2050100@gmail.com> <90823c940909231314j3c03b8e9u45d3104d403a92aa@mail.gmail.com> Message-ID: <20090923204240.GT30343@base.carmickle.com> On Thu, Sep 24, Dmitry Bely wrote: > On Wed, Sep 23, 2009 at 8:08 PM, William King wrote: > (..) > > > Any questions? > > Another problem (on Ubuntu jaunty): > > Setting up freeswitch (1.0.4+repack6-0ubuntu14925.1) ... > adduser: The --group, --ingroup, and --gid options are mutually exclusive. > addgroup: The user `freeswitch' does not exist. > chown: invalid user: `freeswitch:freeswitch' Thanks for the report. Will fix. --FC From carlos.talbot at gmail.com Wed Sep 23 13:55:12 2009 From: carlos.talbot at gmail.com (Carlos Talbot) Date: Wed, 23 Sep 2009 15:55:12 -0500 Subject: [Freeswitch-users] update on Windows installer Message-ID: <5800526b0909231355t7a039d9bm6c3f8e27fe463b43@mail.gmail.com> For those that are not aware, I've made some changes to the Windows installer over the past month. You can find a summary on this link: http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Precompiled_Binaries The latest change is the bundling of 32 and 64 bits builds in one file. The installer automatically selects the proper version based on the running OS. I encourage those who are using Windows to try out this release. (http://files.freeswitch.org/windows_installer/freeswitch.exe) I'm in the process of performance testing the 64 bit build on an HP DL380 w/ 16GB ram (8-way 3GHZ). As with the Linux build it's proven to be more scalabe than the 32 bit build on 64 bit hardware. I will follow up on wiki with my results. I'm trying to mimic this configuration: http://wiki.voiceworks.pl/display/~pawel/Maxed+out+CPS+test regards, Carlos note: it might take a while before you see this recent upload as the files server is synced via a content delivery system. When done you should see a file in that directory with today's date: LATEST_SVN_14953 From hads at nice.net.nz Wed Sep 23 13:57:02 2009 From: hads at nice.net.nz (Hadley Rich) Date: Thu, 24 Sep 2009 08:57:02 +1200 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <4ABA47FB.2050100@gmail.com> References: <4ABA47FB.2050100@gmail.com> Message-ID: <200909240857.02752.hads@nice.net.nz> On Thu, 24 Sep 2009 04:08:27 William King wrote: > Any questions? What file structure did you create the package with i.e. /opt/, /usr/local/, /usr/ - just wondering about inclusion into the archive. Also you mentioned multiverse - what parts have licensing that requires going into multiverse rather than universe? Cheers, hads -- https://nicegear.co.nz VoIP and Open Source Hardware From frank at carmickle.com Wed Sep 23 14:18:23 2009 From: frank at carmickle.com (Frank Carmickle) Date: Wed, 23 Sep 2009 17:18:23 -0400 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <200909240857.02752.hads@nice.net.nz> References: <4ABA47FB.2050100@gmail.com> <200909240857.02752.hads@nice.net.nz> Message-ID: <20090923211822.GW30343@base.carmickle.com> Hi Rich On Thu, Sep 24, Hadley Rich wrote: > What file structure did you create the package with i.e. /opt/, /usr/local/, > /usr/ - just wondering about inclusion into the archive. Currently it's /opt/freeswitch. I would like to see it move to FHS correct locations for inclusion in to debian/ubuntu. This is the next bit that I will be working on. Hopefully we can get it in good shape for the freeze of squeeze. Of course we also hope that the debian voip team will pick it up once we've cleaned it up. > Also you mentioned multiverse - what parts have licensing that requires going > into multiverse rather than universe? I am not an ubuntu guy so I can't speak to that. I would say that most of the licenses of the included packages would allow for inclusion in debian main. Things like the cepstral support would have to go in to contrib. --FC From hads at nice.net.nz Wed Sep 23 14:33:33 2009 From: hads at nice.net.nz (Hadley Rich) Date: Thu, 24 Sep 2009 09:33:33 +1200 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <20090923211822.GW30343@base.carmickle.com> References: <4ABA47FB.2050100@gmail.com> <200909240857.02752.hads@nice.net.nz> <20090923211822.GW30343@base.carmickle.com> Message-ID: <200909240933.33422.hads@nice.net.nz> On Thu, 24 Sep 2009 09:18:23 Frank Carmickle wrote: > Currently it's /opt/freeswitch. I would like to see it move to FHS correct > locations for inclusion in to debian/ubuntu. This is the next bit that I > will be working on. Yeah, the FHS stuff was the bit that I got a little stuck on a while back. > Of course we also hope that the debian voip team will pick it > up once we've cleaned it up. Sounds good. > I am not an ubuntu guy so I can't speak to that. I would say that most of > the licenses of the included packages would allow for inclusion in debian > main. Things like the cepstral support would have to go in to contrib. Gotcha, multiverse is for "not free" software, so anything that can go into main in Debian could go into universe in Ubuntu. hads -- https://nicegear.co.nz VoIP and Open Source Hardware From quentusrex at gmail.com Wed Sep 23 14:48:21 2009 From: quentusrex at gmail.com (William King) Date: Wed, 23 Sep 2009 14:48:21 -0700 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <90823c940909231251k5f85e3dambb7506bdc62e62ac@mail.gmail.com> References: <4ABA47FB.2050100@gmail.com> <90823c940909231251k5f85e3dambb7506bdc62e62ac@mail.gmail.com> Message-ID: <4ABA97A5.4070907@gmail.com> I would be willing to package it. It would go faster with some help, or a patch. One of my goals is to have all of the possible mods for freeswitch as built packages. -William King Dmitry Bely wrote: > On Wed, Sep 23, 2009 at 8:08 PM, William King wrote: > >> Just to give everyone an update. There are working Ubuntu packages in a >> launchpad ppa. Debian users can add the ppa to their apt sources and >> build the package on your box. I'm currently using the packages on my >> home box and it is working great. >> >> Alright. I'm looking for people who want to use the packages. There are >> built packages for Ubuntu 8.04, 8.10, 9.04, and I'm working on 9.10 as >> well. I'm building two apt repos. One will have nightly builds and the >> other will be for tagged releases, plus any major bug fixes. >> >> Thanks to Frank, we now have everything split out into separate files >> for everything. This is to try to reduce the amount of stuff you 'have' >> to download by default. We have the en-us-callie sounds packaged at 8k, >> 16k, 32k, and 48k, we also have packaged the russian-elena and music on >> hold at the same qualities. If there are other languages, or voices I'd >> be more than happy to package them. I just need the 48k. Also we have >> separated the packages out so you can specify which mods you want, such >> as mod_perl, mod_python, mod_lua are all separate so you can install >> them if you want. >> >> nightlies: >> https://launchpad.net/~pbxbuntu-drivers/+archive/ppa >> >> >> tagged releases: >> https://launchpad.net/~freeswitch-drivers/+archive/ppa >> >> >> The tagged released packages will start with the 1.0.5 tagged release >> which some say should be coming out soon. >> >> Any one who would like to help out feel free to sign up with a launchpad >> account, and request to join the driver team of either or both ppa's. >> >> Once we have some people using the packages, and testing them, I have >> already talked to some of the ubuntu official package maintainers about >> what would need to be done to add freeswitch into the ubuntu multiverse >> repo. >> >> Any questions? >> > > Can you enable mod_skypiax in your debian package? > > - Dmitry Bely > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From quentusrex at gmail.com Wed Sep 23 14:51:01 2009 From: quentusrex at gmail.com (William King) Date: Wed, 23 Sep 2009 14:51:01 -0700 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <200909240857.02752.hads@nice.net.nz> References: <4ABA47FB.2050100@gmail.com> <200909240857.02752.hads@nice.net.nz> Message-ID: <4ABA9845.1000200@gmail.com> I would like to get the package into which ever would work best. I haven't taken enough time to find out which repo would be the most appropriate. -William King Hadley Rich wrote: > On Thu, 24 Sep 2009 04:08:27 William King wrote: > >> Any questions? >> > > What file structure did you create the package with i.e. /opt/, /usr/local/, > /usr/ - just wondering about inclusion into the archive. > > Also you mentioned multiverse - what parts have licensing that requires going > into multiverse rather than universe? > > Cheers, > > hads > From quentusrex at gmail.com Wed Sep 23 15:02:27 2009 From: quentusrex at gmail.com (William King) Date: Wed, 23 Sep 2009 15:02:27 -0700 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <200909240933.33422.hads@nice.net.nz> References: <4ABA47FB.2050100@gmail.com> <200909240857.02752.hads@nice.net.nz> <20090923211822.GW30343@base.carmickle.com> <200909240933.33422.hads@nice.net.nz> Message-ID: <4ABA9AF3.30806@gmail.com> It seems I had a port forwarded incorrectly for the external access to the git web interface. here it is again: http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/ I've tested it to work now. -William King Hadley Rich wrote: > On Thu, 24 Sep 2009 09:18:23 Frank Carmickle wrote: > >> Currently it's /opt/freeswitch. I would like to see it move to FHS correct >> locations for inclusion in to debian/ubuntu. This is the next bit that I >> will be working on. >> > > Yeah, the FHS stuff was the bit that I got a little stuck on a while back. > > >> Of course we also hope that the debian voip team will pick it >> up once we've cleaned it up. >> > > Sounds good. > > >> I am not an ubuntu guy so I can't speak to that. I would say that most of >> the licenses of the included packages would allow for inclusion in debian >> main. Things like the cepstral support would have to go in to contrib. >> > > Gotcha, multiverse is for "not free" software, so anything that can go into > main in Debian could go into universe in Ubuntu. > > hads > From lists at venturevoip.com Wed Sep 23 15:06:34 2009 From: lists at venturevoip.com (Matt Riddell) Date: Thu, 24 Sep 2009 10:06:34 +1200 Subject: [Freeswitch-users] Call Files for a dialer engine In-Reply-To: <191c3a030909230849o351bdbaah3dd48c85db07c56b@mail.gmail.com> References: <994476df9083131426fb1d69e257c93b.squirrel@correo.nodo50.org> <191c3a030909230849o351bdbaah3dd48c85db07c56b@mail.gmail.com> Message-ID: <4ABA9BEA.6080104@venturevoip.com> On 24/09/09 3:49 AM, Anthony Minessale wrote: > make an esl script that monitors a dir for new files, and push the > contents into your same db? An easy way to do this is to use incron. -- Cheers, Matt Riddell Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) From dmitry.bely at gmail.com Wed Sep 23 15:07:39 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Thu, 24 Sep 2009 02:07:39 +0400 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <20090923202524.GR30343@base.carmickle.com> References: <4ABA47FB.2050100@gmail.com> <90823c940909231251k5f85e3dambb7506bdc62e62ac@mail.gmail.com> <20090923202524.GR30343@base.carmickle.com> Message-ID: <90823c940909231507ie10dc30x17442386f401c375@mail.gmail.com> On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle wrote: > On Wed, Sep 23, Dmitry Bely wrote: >> Can you enable mod_skypiax in your debian package? > > We will be enabling as much as we can cleanly build on debian/ubuntu. ?There will be a lot more to come. ?We will be breaking the mods and end points in to different packages so that you can install what you like. ?If you have something you would like to see in the package let us know. ?Also patches are welcome. Well, mod_skypiax just requires trivial one-line addition to debian/rules and debian/freeswitch.install. It builds OK. If the patch is required I can post it here. - Dmitry Bely From craig at vastpark.com Wed Sep 23 15:11:49 2009 From: craig at vastpark.com (Craig Presti) Date: Thu, 24 Sep 2009 08:11:49 +1000 Subject: [Freeswitch-users] Conference with pin dialplan In-Reply-To: <20090923064645.GA12159@jdc.jasonjgw.net> References: <014001ca3c12$858bccd0$90a36670$@com> <20090923064645.GA12159@jdc.jasonjgw.net> Message-ID: <02bf01ca3c9a$dab22aa0$90167fe0$@com> Thanks Jason, I'll look into your suggestions. Does this imply that creating a conference which requires a pin via my current approach will not work? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jason White Sent: Wednesday, 23 September 2009 4:47 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Conference with pin dialplan Craig Presti wrote: > I've written a C# module for FS that creates conference dialplans on > the fly. From my limited understanding the easiest way to do this is > by writing XML to the directory: conf/dialplan/default/ - so this is > the approach I've taken. mod_curl might be easier if you have a Web server handy, or you could just write code that attaches to the socket interface, or gets invoked from the dial plan, and creates the conferences. See the wiki for further details. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org __________ Information from ESET NOD32 Antivirus, version of virus signature database 4448 (20090922) __________ The message was checked by ESET NOD32 Antivirus. http://www.eset.com From quentusrex at gmail.com Wed Sep 23 15:14:33 2009 From: quentusrex at gmail.com (William King) Date: Wed, 23 Sep 2009 15:14:33 -0700 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <90823c940909231507ie10dc30x17442386f401c375@mail.gmail.com> References: <4ABA47FB.2050100@gmail.com> <90823c940909231251k5f85e3dambb7506bdc62e62ac@mail.gmail.com> <20090923202524.GR30343@base.carmickle.com> <90823c940909231507ie10dc30x17442386f401c375@mail.gmail.com> Message-ID: <4ABA9DC9.6090808@gmail.com> Sure, post it here and I'll add it in the next build in a few hours. -William King Dmitry Bely wrote: > On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle wrote: > >> On Wed, Sep 23, Dmitry Bely wrote: >> >>> Can you enable mod_skypiax in your debian package? >>> >> We will be enabling as much as we can cleanly build on debian/ubuntu. There will be a lot more to come. We will be breaking the mods and end points in to different packages so that you can install what you like. If you have something you would like to see in the package let us know. Also patches are welcome. >> > > Well, mod_skypiax just requires trivial one-line addition to > debian/rules and debian/freeswitch.install. It builds OK. If the patch > is required I can post it here. > > - Dmitry Bely > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From josh at radianttiger.com Wed Sep 23 15:59:44 2009 From: josh at radianttiger.com (Josh Rivers) Date: Wed, 23 Sep 2009 15:59:44 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5982@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B598E@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B59B3@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5B8E@mse17be1.mse17.exchange.ms> Message-ID: I've been having the same idea, except completely different. I'd probably start with StructureMap or MEF. I'm attracted to the idea of creating an alternative to mod_event_socket except using WCF as the transport, enabling both WS-* and Rest access into the FreeSWITCH core. I think an effort to create a better archtecture in mod_managed is likely to get very religious very quickly. There's a _lot_ of difference of opinion in the .NET ecosystem. Just witness Michael and I disagreeing on something as simple as how to handle unhandled exceptions. Instead of architecting a more complete solution, I think a good solution might be to de-architect the current solution to easily enable extensibility and component replacement. That way, each of us can have the high-level framework we want without disagreement on the low-level framework. This is the same model used in ASP.NET MVC, where it works well out of the box, but you can easily hook in and replace any chunk of it you want if your needs vary. I started playing with a refactoring of the current managed.dll in order to make it more pluggable. It kinda got away from me........I've published my code up on github at http://github.com/joshrivers/FreeSWITCH.Managed The first thing I did was make it _really_ easy to make your own managed.dll. There's two basic points where managed code attaches to unmanaged code. If you rewrite the loader class, you can make your own managed.dll that does anything you want. The next layer up exposes a simple IOC registry that could be modified to replace or add to the logging chain, the module loading chain, or the command execution chain. Beyond that, appdomain loading, script compilation, and inside-appdomain plugin loading are all plugabble. Current plugins should all remain compatible (mine at least haven't needed a recompile). What needs to be done: - There are a few changes in SVN that need to be integrated with my fork. - A lot more unit testing would be useful. - There should be a mechanism for indicating that a script or dll should be pre-loaded into all other appdomains (as a mechanism for defining new plugin types). - There should also be a mechanism for loading a script or dll into the primary appdomain (to allow your code to modify the core operation of mod_managed without recompiling the dll) Most of this should be simple, but I thought it was high time that I got some feedback. I think this model makes it possible to much more simply add new functionality to mod_managed allowing coders to establish their own plugin models and use frameworks such as StructureMap, Spring.NET, MEF, WCF, or Workflow to create their voip applications. Let me know what you think, Josh On Fri, Sep 11, 2009 at 12:55 AM, Raffaele P. Guidi < raffaele.p.guidi at gmail.com> wrote: > *I want to move ILoadNotificationPlugin from being this ?catch all? thing > that controls the entire assembly to something that can be used to fire up > code; effectively ?OnLoad? and ?OnUnload?. To dynamically control loading, > we?ll probably reflect on the individual plugins looking for attributes or > perhaps some sort of static load function.* > I meant to do something like that probably using spring to inject method > names to be invoked. Also event listening (wich is I believe a generic need) > could be managed this way and benefit from some abstraction. con.pop(1) is > probably the most frequently written line by every plugin developer, > probably some abstraction (an event started with his thread and the fs event > passed as an argument?) could make code more elegant > > > On Fri, Sep 11, 2009 at 00:19, Michael Giagnocavo wrote: > >> Well, we have absolutely no idea what the background thread is doing. It >> might be critical, and the fix is trivial: put a try/catch on it. This is >> the model all .NET applications have. Background threads doing bad things >> should always take down the process. >> >> >> >> However, that?s a good point about Load() failing. The approach taken is >> more or less how FreeSWITCH handles things in general now. If a module has >> an error, the switch just logs and goes on. I?m not really in favour of >> this, and suggested at least a ?required? attribute in the modules.conf that >> would prevent the switch from loading if the module fails. >> >> >> >> The fix is probably to create an attribute you can apply to the plugin >> classes that indicate what kind of failure handling you want. For the >> assembly, we?d add an attribute with an enumeration like: >> >> - Default (scan, call ILoadNotificationPlugin, log errors if >> they occur) >> >> - NoLoad (don?t load the assembly) >> >> - Critical (stop the switch if there?s an exception during >> loading) >> >> >> >> That?d provide the control you want for loading. We could do something >> similar for App/Api plugins. >> >> >> >> I want to move ILoadNotificationPlugin from being this ?catch all? thing >> that controls the entire assembly to something that can be used to fire up >> code; effectively ?OnLoad? and ?OnUnload?. To dynamically control loading, >> we?ll probably reflect on the individual plugins looking for attributes or >> perhaps some sort of static load function. >> >> >> >> How?s that sound? >> >> >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers >> *Sent:* Thursday, September 10, 2009 12:48 PM >> >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / >> .NET >> >> >> >> I'm only concerned with the difference in treatment. >> >> public class CrashFreeSWITCH : ILoadNotificationPlugin >> { >> public bool Load() >> { >> ThreadPool.QueueUserWorkItem((o) => { throw new >> NotImplementedException(); }); >> return true; >> } >> } >> >> Crashes the entire switch, terminating all calls and disconnecting from >> the PSTN. >> >> public class CrashFreeSWITCH : ILoadNotificationPlugin >> { >> public bool Load() >> { >> throw new NotImplementedException(); >> return true; >> } >> } >> >> Logs a message to the console and doesn't load the module, while leaving >> the switch operating. >> >> >> >> In my experience, exceptions in multi-threaded code: a) happen, b) are >> hard to diagnose. Is the best behavior for the environment to crash, >> providing no diagnostic information? That's hard in development, and even >> harder in production. I suppose 'terminate switch on fault' should be an >> option, to allow the operating system to restart the whole process on fault >> conditions, but if that is the intended result, shouldn't any fault do the >> same thing? What if the billing was happening in my second code block? >> >> >> >> Normally, I'd trap the ThreadException and UnhandledExceptions in my >> application, so that my code could choose the correct actions to perform >> should the application get into an unknown state. This can include >> terminating the whole application, but also logging diagnostic information, >> trying to save uncommitted data, and sending notifications of the failure. >> >> >> >> Is 'crash if it's a thread, but not if it's not' good because it's the way >> the module works now, or is it a better design for a reason I'm not >> understanding? >> >> >> >> On Wed, Sep 9, 2009 at 11:09 PM, Michael Giagnocavo >> wrote: >> >> Well, a segfault in voicemail would do the same thing. >> >> >> >> Suppose your plugin runs a thread that does something important, like >> billing or so on. That thread fails ? do you really want it to go on? >> >> >> >> Anyways, the solution is simple enough, handle your exceptions J. Every >> plugin can decide what it wants to do here. >> >> >> >> -Michael >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers >> *Sent:* Wednesday, September 09, 2009 10:41 PM >> >> >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / >> .NET >> >> >> >> The question is whether the CLR should take down the whole phone server >> due to an unhandled exception...definitely the CLR should terminate...but >> shouldn't it just log the exception to the console, not crash the core? >> >> On Wed, Sep 9, 2009 at 6:30 PM, Michael Giagnocavo >> wrote: >> >> That?s by design. If a thread fails, and there?s no handler, then the >> application could be in a corrupted state, so the CLR takes down the >> process. >> >> >> >> I think there is a .NET 1.0 compat switch you can enable in the config if >> you like exceptions to be silently ignored J. >> >> >> >> -Michael >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers >> *Sent:* Wednesday, September 09, 2009 6:39 PM >> >> >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / >> .NET >> >> >> >> I have a new thought on the crashes...I'm able to crash FreeSWITCH any >> time I like, just by having an exception in a thread. >> >> >> >> public class CrashFreeSWITCH : ILoadNotificationPlugin >> >> { >> >> public bool Load() >> >> { >> >> ThreadPool.QueueUserWorkItem((o) => { throw new >> NotImplementedException(); }); >> >> return true; >> >> } >> >> } >> >> >> >> Perhaps Application.ThreadException or AppDomain.UnhandledException need >> to be trapped? >> >> >> >> On Wed, Sep 9, 2009 at 4:51 PM, Michael Giagnocavo >> wrote: >> >> >Looks like the event object goes straight to pinvokes, so a null result >> just crashes? >> >> >> >> If it?s null, you should get a NullReferenceException. The C# compiler >> should callvirt the property getter and that?ll do a null check. If that >> isn?t happening, that?d be an interesting optimization somewhere along the >> line. >> >> >> >> -Michael >> >> >> >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Josh Rivers >> *Sent:* Wednesday, September 09, 2009 3:01 PM >> >> >> *To:* freeswitch-users at lists.freeswitch.org >> >> *Subject:* Re: [Freeswitch-users] Subscribing to events in managed C# / >> .NET >> >> >> >> A new discovery: >> >> public bool Load() >> >> { >> >> ThreadPool.QueueUserWorkItem((o) => >> >> { >> >> Log.WriteLine(LogLevel.Notice, "Thread Starting. "); >> >> EventConsumer con = new EventConsumer("all", ""); >> >> while (true) >> >> { >> >> Event ev = con.pop(0); >> >> if (ev == null) continue; >> >> Log.WriteLine(LogLevel.Notice, "Event: " + >> ev.serialized_string); >> >> } >> >> }); >> >> return true; >> >> } >> >> Does not crash. (Adding the null check prevents crash.) The backgrounded >> loop runs fine. Looks like the event object goes straight to pinvokes, so a >> null result just crashes? >> >> >> >> I like the idea of a 'startup-script' for mod_managed. It would also be >> excellent if there was an event or message informing the background code to >> terminate nicely when the module reloads. >> >> >> >> --Josh >> >> >> >> On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk wrote: >> >> >> I think the problem here is that the loader only keeps this method in >> scope >> until completion then it drops the remoted connection. Therefore you >> should >> not use threads in this method. Michael please correct me if I am wrong >> here. >> >> As an example of the failure simply just put a Sleep(10000) call in the >> thread and you will see the failure. >> >> As Michael said this method was only designed to allow the option to opt >> out >> of being loaded. >> >> In order to support this perhaps a configuration flag simular to the lua >> "startup-script" should be added. >> >> >> >> >> Here is the error I get with the loop I mentioned. -Josh >> [image: Capture.PNG] >> >> On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo >> wrote: >> >> > Hi, >> > >> > >> > >> > Can you please elaborate on the crash you receive when >> you >> > queue a thread during load? >> > >> > >> > >> > Thanks, >> > >> > Michael >> > >> > >> >> -- >> View this message in context: >> http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/f7b53337/attachment-0001.html From grae at digilord.net Wed Sep 23 16:38:18 2009 From: grae at digilord.net (Daniel Morrigan) Date: Wed, 23 Sep 2009 16:38:18 -0700 Subject: [Freeswitch-users] Polycom MWI Forgetfulness In-Reply-To: <437FA030-C96F-4E46-8727-E3F4FA04D6EF@freeswitch.org> References: <68755a9c0909231044ta70d850vab3ce7971992433d@mail.gmail.com> <437FA030-C96F-4E46-8727-E3F4FA04D6EF@freeswitch.org> Message-ID: <68755a9c0909231638t5e39cfbkc0d1b48e0891d8d8@mail.gmail.com> Brian, It was set for contact. Would that cause this behavior? Daniel On Wed, Sep 23, 2009 at 11:15 AM, Brian West wrote: > I have never seen this behavior. What do you have your MWI callback > method setup as? > > /b > > On Sep 23, 2009, at 12:44 PM, Daniel Morrigan wrote: > > > Hello, > > I have a number of Polycom phones that appear to forget that > > they have a message waiting then a little while later they seem to > > remember again. This was not the case with the same phones under > > Asterisk (Sorry to say that here but it's the only comparison I have). > > > > I have looked at a number of settings in sofia and can't seem to > > find one that makes any difference. > > > > Does anyone else have this issue? How did you solve it? > > > > DigiLord > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/897a6c32/attachment.html From grae at digilord.net Wed Sep 23 16:39:00 2009 From: grae at digilord.net (Daniel Morrigan) Date: Wed, 23 Sep 2009 16:39:00 -0700 Subject: [Freeswitch-users] Polycom MWI Forgetfulness In-Reply-To: <68755a9c0909231638t5e39cfbkc0d1b48e0891d8d8@mail.gmail.com> References: <68755a9c0909231044ta70d850vab3ce7971992433d@mail.gmail.com> <437FA030-C96F-4E46-8727-E3F4FA04D6EF@freeswitch.org> <68755a9c0909231638t5e39cfbkc0d1b48e0891d8d8@mail.gmail.com> Message-ID: <68755a9c0909231639r5a7eef13w39afc6136d68bf13@mail.gmail.com> Brian, Just for kicks I changed it to registration. Same thing is happening. Daniel On Wed, Sep 23, 2009 at 4:38 PM, Daniel Morrigan wrote: > Brian, > It was set for contact. Would that cause this behavior? > > Daniel > > > On Wed, Sep 23, 2009 at 11:15 AM, Brian West wrote: > >> I have never seen this behavior. What do you have your MWI callback >> method setup as? >> >> /b >> >> On Sep 23, 2009, at 12:44 PM, Daniel Morrigan wrote: >> >> > Hello, >> > I have a number of Polycom phones that appear to forget that >> > they have a message waiting then a little while later they seem to >> > remember again. This was not the case with the same phones under >> > Asterisk (Sorry to say that here but it's the only comparison I have). >> > >> > I have looked at a number of settings in sofia and can't seem to >> > find one that makes any difference. >> > >> > Does anyone else have this issue? How did you solve it? >> > >> > DigiLord >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/e0a07830/attachment.html From brian at freeswitch.org Wed Sep 23 17:08:39 2009 From: brian at freeswitch.org (Brian West) Date: Wed, 23 Sep 2009 19:08:39 -0500 Subject: [Freeswitch-users] Polycom MWI Forgetfulness In-Reply-To: <68755a9c0909231638t5e39cfbkc0d1b48e0891d8d8@mail.gmail.com> References: <68755a9c0909231044ta70d850vab3ce7971992433d@mail.gmail.com> <437FA030-C96F-4E46-8727-E3F4FA04D6EF@freeswitch.org> <68755a9c0909231638t5e39cfbkc0d1b48e0891d8d8@mail.gmail.com> Message-ID: NO I have never seen it happen what firmware version are you running? /b On Sep 23, 2009, at 6:38 PM, Daniel Morrigan wrote: > Brian, > It was set for contact. Would that cause this behavior? > > Daniel From mgg at giagnocavo.net Wed Sep 23 19:31:26 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Wed, 23 Sep 2009 22:31:26 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5982@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B598E@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B59B3@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5B8E@mse17be1.mse17.exchange.ms> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DCEC6B31@mse17be1.mse17.exchange.ms> Hey nice to see someone interested in this - that's a lot of files you have there, and looks like you put a lot of effort into it! I haven't had time to look at it much, but here are a few initial impressions. Right off the bat: there can be tons of cleanup and refactoring, no doubt about that. Much of the current code is to satisfy my needs in production, which it does very well. With that out of the way... I'm a bit hesitant to go too far from the FreeSWITCH core as far as architecture goes. For instance, I'm not quite sure why'd we have our own managed logging subsystem that allows them to plug in other things that aren't part of FS. Either they should use the FS logging system, or use their own such as log4net. Or perhaps I don't see why we'd want this behavior. Going away from the core as far as adding .NET specific features (like look at the static ManagedSession.Originate that takes hangup delegates, or the "nice" wrapper for Log (Write and WeiteLine, with an enum instead of a string) are keeping close to the core, just adding a tiny bit of API cleanup. FreeSWITCH exposes a lot of strings, and while maybe that's important for some languages, .NET users are going to expect stronger typing. But I don't think these types of things get people away from FreeSWITCH much. Things like making a published SOAP interface for FS seem not really related to mod_managed. They can easily be done as 3rd party plugins, or convince the core FS team that exposing via SOAP via mod_managed is the way to go. Also keep in mind that the majority of users are on Linux, so that rules out WCF and some other fun stuff that only works on the CLR - I'd say it all has to work on Mono. As for all the rest of it, can we talk interactively, perhaps with other users interested in mod_managed? Reading over your email, I think I'm not understanding many of the use cases that are being fixed. Thanks a lot! -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Wednesday, September 23, 2009 5:00 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET I've been having the same idea, except completely different. I'd probably start with StructureMap or MEF. I'm attracted to the idea of creating an alternative to mod_event_socket except using WCF as the transport, enabling both WS-* and Rest access into the FreeSWITCH core. I think an effort to create a better archtecture in mod_managed is likely to get very religious very quickly. There's a _lot_ of difference of opinion in the .NET ecosystem. Just witness Michael and I disagreeing on something as simple as how to handle unhandled exceptions. Instead of architecting a more complete solution, I think a good solution might be to de-architect the current solution to easily enable extensibility and component replacement. That way, each of us can have the high-level framework we want without disagreement on the low-level framework. This is the same model used in ASP.NET MVC, where it works well out of the box, but you can easily hook in and replace any chunk of it you want if your needs vary. I started playing with a refactoring of the current managed.dll in order to make it more pluggable. It kinda got away from me........I've published my code up on github at http://github.com/joshrivers/FreeSWITCH.Managed The first thing I did was make it _really_ easy to make your own managed.dll. There's two basic points where managed code attaches to unmanaged code. If you rewrite the loader class, you can make your own managed.dll that does anything you want. The next layer up exposes a simple IOC registry that could be modified to replace or add to the logging chain, the module loading chain, or the command execution chain. Beyond that, appdomain loading, script compilation, and inside-appdomain plugin loading are all plugabble. Current plugins should all remain compatible (mine at least haven't needed a recompile). What needs to be done: - There are a few changes in SVN that need to be integrated with my fork. - A lot more unit testing would be useful. - There should be a mechanism for indicating that a script or dll should be pre-loaded into all other appdomains (as a mechanism for defining new plugin types). - There should also be a mechanism for loading a script or dll into the primary appdomain (to allow your code to modify the core operation of mod_managed without recompiling the dll) Most of this should be simple, but I thought it was high time that I got some feedback. I think this model makes it possible to much more simply add new functionality to mod_managed allowing coders to establish their own plugin models and use frameworks such as StructureMap, Spring.NET, MEF, WCF, or Workflow to create their voip applications. Let me know what you think, Josh On Fri, Sep 11, 2009 at 12:55 AM, Raffaele P. Guidi > wrote: I want to move ILoadNotificationPlugin from being this "catch all" thing that controls the entire assembly to something that can be used to fire up code; effectively "OnLoad" and "OnUnload". To dynamically control loading, we'll probably reflect on the individual plugins looking for attributes or perhaps some sort of static load function. I meant to do something like that probably using spring to inject method names to be invoked. Also event listening (wich is I believe a generic need) could be managed this way and benefit from some abstraction. con.pop(1) is probably the most frequently written line by every plugin developer, probably some abstraction (an event started with his thread and the fs event passed as an argument?) could make code more elegant On Fri, Sep 11, 2009 at 00:19, Michael Giagnocavo > wrote: Well, we have absolutely no idea what the background thread is doing. It might be critical, and the fix is trivial: put a try/catch on it. This is the model all .NET applications have. Background threads doing bad things should always take down the process. However, that's a good point about Load() failing. The approach taken is more or less how FreeSWITCH handles things in general now. If a module has an error, the switch just logs and goes on. I'm not really in favour of this, and suggested at least a "required" attribute in the modules.conf that would prevent the switch from loading if the module fails. The fix is probably to create an attribute you can apply to the plugin classes that indicate what kind of failure handling you want. For the assembly, we'd add an attribute with an enumeration like: - Default (scan, call ILoadNotificationPlugin, log errors if they occur) - NoLoad (don't load the assembly) - Critical (stop the switch if there's an exception during loading) That'd provide the control you want for loading. We could do something similar for App/Api plugins. I want to move ILoadNotificationPlugin from being this "catch all" thing that controls the entire assembly to something that can be used to fire up code; effectively "OnLoad" and "OnUnload". To dynamically control loading, we'll probably reflect on the individual plugins looking for attributes or perhaps some sort of static load function. How's that sound? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Thursday, September 10, 2009 12:48 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET I'm only concerned with the difference in treatment. public class CrashFreeSWITCH : ILoadNotificationPlugin { public bool Load() { ThreadPool.QueueUserWorkItem((o) => { throw new NotImplementedException(); }); return true; } } Crashes the entire switch, terminating all calls and disconnecting from the PSTN. public class CrashFreeSWITCH : ILoadNotificationPlugin { public bool Load() { throw new NotImplementedException(); return true; } } Logs a message to the console and doesn't load the module, while leaving the switch operating. In my experience, exceptions in multi-threaded code: a) happen, b) are hard to diagnose. Is the best behavior for the environment to crash, providing no diagnostic information? That's hard in development, and even harder in production. I suppose 'terminate switch on fault' should be an option, to allow the operating system to restart the whole process on fault conditions, but if that is the intended result, shouldn't any fault do the same thing? What if the billing was happening in my second code block? Normally, I'd trap the ThreadException and UnhandledExceptions in my application, so that my code could choose the correct actions to perform should the application get into an unknown state. This can include terminating the whole application, but also logging diagnostic information, trying to save uncommitted data, and sending notifications of the failure. Is 'crash if it's a thread, but not if it's not' good because it's the way the module works now, or is it a better design for a reason I'm not understanding? On Wed, Sep 9, 2009 at 11:09 PM, Michael Giagnocavo > wrote: Well, a segfault in voicemail would do the same thing. Suppose your plugin runs a thread that does something important, like billing or so on. That thread fails - do you really want it to go on? Anyways, the solution is simple enough, handle your exceptions :). Every plugin can decide what it wants to do here. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Wednesday, September 09, 2009 10:41 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET The question is whether the CLR should take down the whole phone server due to an unhandled exception...definitely the CLR should terminate...but shouldn't it just log the exception to the console, not crash the core? On Wed, Sep 9, 2009 at 6:30 PM, Michael Giagnocavo > wrote: That's by design. If a thread fails, and there's no handler, then the application could be in a corrupted state, so the CLR takes down the process. I think there is a .NET 1.0 compat switch you can enable in the config if you like exceptions to be silently ignored :). -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Wednesday, September 09, 2009 6:39 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET I have a new thought on the crashes...I'm able to crash FreeSWITCH any time I like, just by having an exception in a thread. public class CrashFreeSWITCH : ILoadNotificationPlugin { public bool Load() { ThreadPool.QueueUserWorkItem((o) => { throw new NotImplementedException(); }); return true; } } Perhaps Application.ThreadException or AppDomain.UnhandledException need to be trapped? On Wed, Sep 9, 2009 at 4:51 PM, Michael Giagnocavo > wrote: >Looks like the event object goes straight to pinvokes, so a null result just crashes? If it's null, you should get a NullReferenceException. The C# compiler should callvirt the property getter and that'll do a null check. If that isn't happening, that'd be an interesting optimization somewhere along the line. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Wednesday, September 09, 2009 3:01 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET A new discovery: public bool Load() { ThreadPool.QueueUserWorkItem((o) => { Log.WriteLine(LogLevel.Notice, "Thread Starting. "); EventConsumer con = new EventConsumer("all", ""); while (true) { Event ev = con.pop(0); if (ev == null) continue; Log.WriteLine(LogLevel.Notice, "Event: " + ev.serialized_string); } }); return true; } Does not crash. (Adding the null check prevents crash.) The backgrounded loop runs fine. Looks like the event object goes straight to pinvokes, so a null result just crashes? I like the idea of a 'startup-script' for mod_managed. It would also be excellent if there was an event or message informing the background code to terminate nicely when the module reloads. --Josh On Wed, Sep 9, 2009 at 12:57 PM, Jeff Lenk > wrote: I think the problem here is that the loader only keeps this method in scope until completion then it drops the remoted connection. Therefore you should not use threads in this method. Michael please correct me if I am wrong here. As an example of the failure simply just put a Sleep(10000) call in the thread and you will see the failure. As Michael said this method was only designed to allow the option to opt out of being loaded. In order to support this perhaps a configuration flag simular to the lua "startup-script" should be added. Here is the error I get with the loop I mentioned. -Josh [image: Capture.PNG] On Tue, Sep 8, 2009 at 5:05 AM, Michael Giagnocavo >wrote: > Hi, > > > > Can you please elaborate on the crash you receive when you > queue a thread during load? > > > > Thanks, > > Michael > > -- View this message in context: http://n2.nabble.com/Subscribing-to-events-in-managed-C-NET-tp3573619p3613195.html Sent from the freeswitch-users mailing list archive at Nabble.com. _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/366e59d0/attachment-0001.html From yehavi.bourvine at gmail.com Wed Sep 23 21:07:06 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 24 Sep 2009 07:07:06 +0300 Subject: [Freeswitch-users] Bind to more than one ethernet interface Message-ID: Hello, I am trying to run FreeSwitch on a machine which has more than one interface, all of them should be used for SIP. The FreeSwitch binds only to the first one. I tried setting bind_server_ip to either "auto" or 0.0.0.0 but it doesn't help. Any idea what to do? Thanks! _Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/d2e307a0/attachment.html From dujinfang at gmail.com Wed Sep 23 21:46:20 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 24 Sep 2009 12:46:20 +0800 Subject: [Freeswitch-users] Bind to more than one ethernet interface In-Reply-To: <23f91030909232144v2389170fx5556bb7b83b5e361@mail.gmail.com> References: <23f91030909232144v2389170fx5556bb7b83b5e361@mail.gmail.com> Message-ID: <23f91030909232146q7e142066m7ec7c127e8bc8c9c@mail.gmail.com> sorry when I said "on profile" I want to say "one profile" 2009/9/24 Seven Du > It not possible to use 0.0.0.0 for on profile. however, you can create more > sip profiles for each of your interfaces. Search freeswitch-users archievs > then you will find similar topics. > > 2009/9/24 Yehavi Bourvine > >> Hello, >> >> I am trying to run FreeSwitch on a machine which has more than one >> interface, all of them should be used for SIP. The FreeSwitch binds only to >> the first one. I tried setting bind_server_ip to either "auto" or 0.0.0.0 >> but it doesn't help. >> >> Any idea what to do? >> >> Thanks! _Yehavi: >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/b5544603/attachment.html From dujinfang at gmail.com Wed Sep 23 21:44:21 2009 From: dujinfang at gmail.com (Seven Du) Date: Thu, 24 Sep 2009 12:44:21 +0800 Subject: [Freeswitch-users] Bind to more than one ethernet interface In-Reply-To: References: Message-ID: <23f91030909232144v2389170fx5556bb7b83b5e361@mail.gmail.com> It not possible to use 0.0.0.0 for on profile. however, you can create more sip profiles for each of your interfaces. Search freeswitch-users archievs then you will find similar topics. 2009/9/24 Yehavi Bourvine > Hello, > > I am trying to run FreeSwitch on a machine which has more than one > interface, all of them should be used for SIP. The FreeSwitch binds only to > the first one. I tried setting bind_server_ip to either "auto" or 0.0.0.0 > but it doesn't help. > > Any idea what to do? > > Thanks! _Yehavi: > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/529945d3/attachment.html From yehavi.bourvine at gmail.com Wed Sep 23 22:05:32 2009 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 24 Sep 2009 08:05:32 +0300 Subject: [Freeswitch-users] Bind to more than one ethernet interface In-Reply-To: <23f91030909232144v2389170fx5556bb7b83b5e361@mail.gmail.com> References: <23f91030909232144v2389170fx5556bb7b83b5e361@mail.gmail.com> Message-ID: Thanks! __Yehavi: 2009/9/24 Seven Du > It not possible to use 0.0.0.0 for on profile. however, you can create more > sip profiles for each of your interfaces. Search freeswitch-users archievs > then you will find similar topics. > > 2009/9/24 Yehavi Bourvine > >> Hello, >> >> I am trying to run FreeSwitch on a machine which has more than one >> interface, all of them should be used for SIP. The FreeSwitch binds only to >> the first one. I tried setting bind_server_ip to either "auto" or 0.0.0.0 >> but it doesn't help. >> >> Any idea what to do? >> >> Thanks! _Yehavi: >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/e53cc894/attachment.html From gabe at gundy.org Wed Sep 23 22:15:15 2009 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 23 Sep 2009 23:15:15 -0600 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <4ABA47FB.2050100@gmail.com> References: <4ABA47FB.2050100@gmail.com> Message-ID: <903da5680909232215r1002b906j50b11d6da7cc6550@mail.gmail.com> On Wed, Sep 23, 2009 at 10:08 AM, William King wrote: >Once we have some people using the packages, and testing them, I have > already talked to some of the ubuntu official package maintainers about > what would need to be done to add freeswitch into the ubuntu multiverse > repo. > > Any questions? No questions. I'll start testing. I'll be testing 8.04 and check in on 9.10 from time to time. Thanks for the great work. Keep it up! Best, Gabe From gabe at gundy.org Wed Sep 23 22:22:38 2009 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 23 Sep 2009 23:22:38 -0600 Subject: [Freeswitch-users] Bind to more than one ethernet interface In-Reply-To: <23f91030909232144v2389170fx5556bb7b83b5e361@mail.gmail.com> References: <23f91030909232144v2389170fx5556bb7b83b5e361@mail.gmail.com> Message-ID: <903da5680909232222n244b924bp4d875a6988e71733@mail.gmail.com> On Wed, Sep 23, 2009 at 10:44 PM, Seven Du wrote: > It not possible to use 0.0.0.0 for on profile. however, you can create more > sip profiles for each of your interfaces. Search freeswitch-users archievs > then you will find similar topics. It sure would be nice to be able to provide a list of IPs in one profile. Seems like I run into that need often. Instead, I end up making two profiles that are mostly the same. Is there a reason why this is the way it is? Regards, Gabe From gabe at gundy.org Wed Sep 23 22:36:41 2009 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 23 Sep 2009 23:36:41 -0600 Subject: [Freeswitch-users] User Creation with DB in Freeswitch In-Reply-To: <1b2118200909231137w79ea33afq6050133fb50b32a0@mail.gmail.com> References: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> <566E110B-72CC-428E-AEE8-94AB600587B0@enterux.com> <347D3032-F778-46F4-AECE-94DECD27E849@freeswitch.org> <1b2118200909231137w79ea33afq6050133fb50b32a0@mail.gmail.com> Message-ID: <903da5680909232236v21cd9f47xa2aaa83749a16fab@mail.gmail.com> On Wed, Sep 23, 2009 at 12:37 PM, Anil Kumar S. R. wrote: > I didn't get much help for my problem with XML CURL. What I meant to say is, > suppose I want to have some 10000 users on freeswitch. Do we have to create > some many xml files in the directory or is there some way in which the users > can be put in the db ? That's the whole point. You serve up the XML from *your* web server using whatever technologies that *you* would like on the back-end. You'll want to make that XML reply dynamic. Use php, perl, python, c#, java or whatever other language *you* want. Pull the data from MySQL, flat-files, PostgreSQL, MSSQL, LDAP or whatever *you* want. Just serve up the right bit of configuration to FreeSWITCH and you're done. Good luck. Spend more time in the docs. Others have posted the links. Best, Gabe From josh at radianttiger.com Wed Sep 23 23:31:34 2009 From: josh at radianttiger.com (Josh Rivers) Date: Wed, 23 Sep 2009 23:31:34 -0700 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DCEC6B31@mse17be1.mse17.exchange.ms> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B598E@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B59B3@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5B8E@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DCEC6B31@mse17be1.mse17.exchange.ms> Message-ID: On Wed, Sep 23, 2009 at 7:31 PM, Michael Giagnocavo wrote: > Right off the bat: there can be tons of cleanup and refactoring, no doubt > about that. Much of the current code is to satisfy my needs in production, > which it does very well. > The current base doesn't have anything wrong with it for sure, in fact, I learned a good bit about PInvoke. AppDomains, and In-Process Remoting in the last week. My refactoring had the following goals (in no particular order) - Testability - I'd really like to see a decent unit test suite on the more module so that we can change it with confidence. Also, it's been drilled into me that a testable design is a good design. - Clarity - Where possible, I extracted blocks of code that served a particular purpose so that purpose could be self-documenting in the method calls rather than mixed in. - Modularity - I wanted to make it easy to remove or add alternative behavior to the managed.dll. > I?m a bit hesitant to go too far from the FreeSWITCH core as far as > architecture goes. For instance, I?m not quite sure why?d we have our own > managed logging subsystem that allows them to plug in other things that > aren?t part of FS. Either they should use the FS logging system, or use > their own such as log4net. Or perhaps I don?t see why we?d want this > behavior. > I completely agree, with the following caveats: 1) I'd like to see things testable. It's very hard to do isolation testing with classes making direct calls out to a static Log class that in turn pinvokes out to unmanaged code. 2) I'd like to allow folk to make changes to the default behavior (optimally) without recompiling managed.dll. One thing at issue here is that there are two principal purposes for managed.dll. The first is to provide an interface into unmanaged code. The second is a module/plugin extensibility framework. The first purpose should absolutely provide the thinnest layer possible. The second purpose is very likely to need a lot of change and adaptation as people come up with development models that they would like to follow in using freeswitch. The extensibility framework should be mostly managed code, coded to interfaces for mock-ability and testabiliy. It should also be able to just push it out of the way and hook your own extensibilty framework in instead. > Going away from the core as far as adding .NET specific features (like > look at the static ManagedSession.Originate that takes hangup delegates, or > the ?nice? wrapper for Log (Write and WeiteLine, with an enum instead of a > string) are keeping close to the core, just adding a tiny bit of API > cleanup. FreeSWITCH exposes a lot of strings, and while maybe that?s > important for some languages, .NET users are going to expect stronger > typing. But I don?t think these types of things get people away from > FreeSWITCH much. > No disagreement here. I would like to see these things made available by interface rather than concrete implementation. It's currently not possible to test a plugin without loading it into FS. That precludes automated testing, and leaves a pretty big round-trip to test a tweak. I'm a sloppy coder too, so I'm always introducing interesting regressions, and that's why I like doing my testing without having to bring up a full process :) > Things like making a published SOAP interface for FS seem not really > related to mod_managed. They can easily be done as 3rd party plugins, or > convince the core FS team that exposing via SOAP via mod_managed is the way > to go. Also keep in mind that the majority of users are on Linux, so that > rules out WCF and some other fun stuff that only works on the CLR ? I?d say > it all has to work on Mono. > This kind of stuff is definitely beyond the scope of mod_managed. Although there is a slippery slope since we're building in an extensibility model. I don't think a WCF host, or a winforms host, or any of that should be baked in. Rather, I think we should provide the hooks for adding such a thing. If somebody wants to build ESL via WCF, why should they need to leave managed code? If the module system is general enough, then such a thing should just be a module. (BTW, I think WCF-Mono is getting there http://www.mono-project.com/WCF_Development) Absolutely, everything in mod_managed and managed.dll should run on mono and the CLR. However, there shouldn't be any reason that a Win-only developer can't build a complete FS application framework that plugs in and only runs on Windows. > As for all the rest of it, can we talk interactively, perhaps with other > users interested in mod_managed? Reading over your email, I think I?m not > understanding many of the use cases that are being fixed. > I'd be very glad to get a discussion going. I definitely haven't covered all of the issues here. -Josh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090923/72445c47/attachment-0001.html From velu.technical at gmail.com Wed Sep 23 23:38:14 2009 From: velu.technical at gmail.com (velusamy velu) Date: Thu, 24 Sep 2009 12:08:14 +0530 Subject: [Freeswitch-users] How to play more than one voice file in play_and_get_digits function Message-ID: <1452e2980909232338k3434c32ch7e0fe8b9bf3d405b@mail.gmail.com> Dear All, I am in the process of doing IVR development on FreeSWITCH. I am having doubt in the play_and_get_digits application. I am using Perl language for handling IVR. How can I play more than one sound file in play_get_digits application? For an example, $conn->execute("play_and_get_digit", 1 1 1000 # /usr/local/freeswitch/en/us/callie/sounds/ivr/please.wav , /usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+); Is the above statement is right? Can I use more than one file in it? OR Can I use the play_get_digits as following? $conn->execute("play_and_get_digit", 1 1 1000 # /usr/local/freeswitch/en/us/callie/sounds/ivr/please.av /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+); $conn->execute("play_and_get_digit", 1 1 1000 # /usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+); In the above statements how can I assure that the first application or second application is executed? When the digit is get while playing the first application the second application should not be played. How Can I do that? Is this my understanding wrong? Correct me If I am wrong? Please help me? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/f7fe1d2d/attachment.html From msc at freeswitch.org Thu Sep 24 00:02:52 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 24 Sep 2009 00:02:52 -0700 Subject: [Freeswitch-users] How to play more than one voice file in play_and_get_digits function In-Reply-To: <1452e2980909232338k3434c32ch7e0fe8b9bf3d405b@mail.gmail.com> References: <1452e2980909232338k3434c32ch7e0fe8b9bf3d405b@mail.gmail.com> Message-ID: <87f2f3b90909240002t274128bdha650aa1b64390bf2@mail.gmail.com> Make sure that mod_file_string is built and loaded and then try the syntax that is described here: http://wiki.freeswitch.org/wiki/Mod_file_string#Examples Instead of a comma separated list you can use ! and be sure NOT to put a space after the ! because the function delimits the arguments with spaces. Try something like this: $conn->execute("play_and_get_digits", 1 1 1000 # /usr/local/freeswitch/en/us/callie/sounds/ivr/please.wav!/usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+); Let us know how it goes. -MC On Wed, Sep 23, 2009 at 11:38 PM, velusamy velu wrote: > Dear All, > > I am in the process of doing IVR development on FreeSWITCH. I am > having doubt in the play_and_get_digits application. I am using Perl > language for handling IVR. > > How can I play more than one sound file in play_get_digits > application? > For an example, > > $conn->execute("play_and_get_digit", 1 1 1000 # > /usr/local/freeswitch/en/us/callie/sounds/ivr/please.wav , > /usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav > /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+); > > Is the above statement is right? Can I use more than one file in it? > > OR > > Can I use the play_get_digits as following? > > $conn->execute("play_and_get_digit", 1 1 1000 # > /usr/local/freeswitch/en/us/callie/sounds/ivr/please.av > /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+); > > $conn->execute("play_and_get_digit", 1 1 1000 # > /usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav > /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+); > > In the above statements how can I assure that the first application or > second application is executed? > > When the digit is get while playing the first application the second > application should not be played. How Can I do that? > > Is this my understanding wrong? > Correct me If I am wrong? > > > Please help me? > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/b68cf2f1/attachment.html From math.parent at gmail.com Thu Sep 24 00:36:34 2009 From: math.parent at gmail.com (Mathieu Parent) Date: Thu, 24 Sep 2009 09:36:34 +0200 Subject: [Freeswitch-users] mod_fax not working In-Reply-To: <25123485.314001253742294303.JavaMail.nabble@isper.nabble.com> References: <25123485.314001253742294303.JavaMail.nabble@isper.nabble.com> Message-ID: <960738410909240036q19f71189q663ea5954c3695a4@mail.gmail.com> On Wed, Sep 23, 2009 at 11:44 PM, wrote: > Hi Mathieu Hi > Does this mean you are able to use email-to-fax? > If yes, would yes would you care to briefly describe how you configured that. Not yet, but I plan to do so. Il will post my setup in FS wiki. I first have to solve the receiving problem (fax2mail is more important than mail2fax for us). > Thanks > MC > Mathieu Parent From mgg at giagnocavo.net Thu Sep 24 01:01:39 2009 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 24 Sep 2009 04:01:39 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B598E@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B59B3@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5B8E@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DCEC6B31@mse17be1.mse17.exchange.ms> Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702DCEC6B3C@mse17be1.mse17.exchange.ms> Great - hopefully we'll meet on IRC or the conference sometime on Friday. Email me when you're on. A few questions I have: Clarity - I agree with you there, and thanks! Testability - is this even remotely practical? Looking at our FS code plugins, there's simply no way any amount of test environment code would get us to anything testable. We make tons of direct P/Invoke calls, and the whole model for what variables are set when, the state machine progression, etc. does not seem like something that we can hope to possibly model right. And it's subject to many external influences (all the modules you have loaded in FS). Logging is a pretty simple case, sure, we can make it not call FS for testing. But in a real app, it just seems that there are way too many dependencies, no? Maybe others who have apps written can chime in? Modularity - I agree there are two parts. But, I think they are pretty tightly coupled. The FS interface into unmanaged code is done via unmanaged code and is really clear: App, Api, ApiBackground. The other ways I can think of are FS-specific, such as XML binding interface and so on. But those are things we should just add to the mod_managed core and be done with. I'm thinking maybe we are talking about different things? Can you provide some user stories that we want to cover with a pluggable loader/executor/etc.? Thanks for putting up with me! -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Rivers Sent: Thursday, September 24, 2009 12:32 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Subscribing to events in managed C# / .NET On Wed, Sep 23, 2009 at 7:31 PM, Michael Giagnocavo > wrote: Right off the bat: there can be tons of cleanup and refactoring, no doubt about that. Much of the current code is to satisfy my needs in production, which it does very well. The current base doesn't have anything wrong with it for sure, in fact, I learned a good bit about PInvoke. AppDomains, and In-Process Remoting in the last week. My refactoring had the following goals (in no particular order) - Testability - I'd really like to see a decent unit test suite on the more module so that we can change it with confidence. Also, it's been drilled into me that a testable design is a good design. - Clarity - Where possible, I extracted blocks of code that served a particular purpose so that purpose could be self-documenting in the method calls rather than mixed in. - Modularity - I wanted to make it easy to remove or add alternative behavior to the managed.dll. I'm a bit hesitant to go too far from the FreeSWITCH core as far as architecture goes. For instance, I'm not quite sure why'd we have our own managed logging subsystem that allows them to plug in other things that aren't part of FS. Either they should use the FS logging system, or use their own such as log4net. Or perhaps I don't see why we'd want this behavior. I completely agree, with the following caveats: 1) I'd like to see things testable. It's very hard to do isolation testing with classes making direct calls out to a static Log class that in turn pinvokes out to unmanaged code. 2) I'd like to allow folk to make changes to the default behavior (optimally) without recompiling managed.dll. One thing at issue here is that there are two principal purposes for managed.dll. The first is to provide an interface into unmanaged code. The second is a module/plugin extensibility framework. The first purpose should absolutely provide the thinnest layer possible. The second purpose is very likely to need a lot of change and adaptation as people come up with development models that they would like to follow in using freeswitch. The extensibility framework should be mostly managed code, coded to interfaces for mock-ability and testabiliy. It should also be able to just push it out of the way and hook your own extensibilty framework in instead. Going away from the core as far as adding .NET specific features (like look at the static ManagedSession.Originate that takes hangup delegates, or the "nice" wrapper for Log (Write and WeiteLine, with an enum instead of a string) are keeping close to the core, just adding a tiny bit of API cleanup. FreeSWITCH exposes a lot of strings, and while maybe that's important for some languages, .NET users are going to expect stronger typing. But I don't think these types of things get people away from FreeSWITCH much. No disagreement here. I would like to see these things made available by interface rather than concrete implementation. It's currently not possible to test a plugin without loading it into FS. That precludes automated testing, and leaves a pretty big round-trip to test a tweak. I'm a sloppy coder too, so I'm always introducing interesting regressions, and that's why I like doing my testing without having to bring up a full process :) Things like making a published SOAP interface for FS seem not really related to mod_managed. They can easily be done as 3rd party plugins, or convince the core FS team that exposing via SOAP via mod_managed is the way to go. Also keep in mind that the majority of users are on Linux, so that rules out WCF and some other fun stuff that only works on the CLR - I'd say it all has to work on Mono. This kind of stuff is definitely beyond the scope of mod_managed. Although there is a slippery slope since we're building in an extensibility model. I don't think a WCF host, or a winforms host, or any of that should be baked in. Rather, I think we should provide the hooks for adding such a thing. If somebody wants to build ESL via WCF, why should they need to leave managed code? If the module system is general enough, then such a thing should just be a module. (BTW, I think WCF-Mono is getting there http://www.mono-project.com/WCF_Development) Absolutely, everything in mod_managed and managed.dll should run on mono and the CLR. However, there shouldn't be any reason that a Win-only developer can't build a complete FS application framework that plugs in and only runs on Windows. As for all the rest of it, can we talk interactively, perhaps with other users interested in mod_managed? Reading over your email, I think I'm not understanding many of the use cases that are being fixed. I'd be very glad to get a discussion going. I definitely haven't covered all of the issues here. -Josh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/7b17a792/attachment-0001.html From R.Kloosterman at mtel.nl Thu Sep 24 01:09:39 2009 From: R.Kloosterman at mtel.nl (Remko Kloosterman) Date: Thu, 24 Sep 2009 10:09:39 +0200 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <87f2f3b90909171357y40fe0db4kef1d59fb86e9790c@mail.gmail.com> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com><11372C8B9E603F4FACDE6AB18256DEC695A9EC@srvmtel.office.mtel.nl> <87f2f3b90909171357y40fe0db4kef1d59fb86e9790c@mail.gmail.com> Message-ID: <11372C8B9E603F4FACDE6AB18256DEC601D8BC94@srvmtel.office.mtel.nl> Hello Michael, Do you still want to follow up on this? I'm having difficulty gathering the old stuff in an understandable form. Also, it looks like the open source ACD Spice Telephony by Andrew Thompson can do just what you might need. Remko Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Michael Collins Verzonden: donderdag 17 september 2009 22:58 Aan: freeswitch-users at lists.freeswitch.org Onderwerp: Re: [Freeswitch-users] skill-based ACD On Thu, Sep 17, 2009 at 12:13 PM, Remko Kloosterman wrote: I have been working on several voice projects in the past with ACD features, mostly based on TDM technology. It's all commercial stuff, but I have the experience and I am willing to share that. If anyone wishes to start such a development I'm sure I can dig up a functional model and help with the design. I would like to see the functional model. That sounds interesting. We could take it from there. Perhaps the FS community will have a few members willing to help out. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/e03731eb/attachment.html From jim.page at redmatter.com Thu Sep 24 01:25:20 2009 From: jim.page at redmatter.com (Jim Page) Date: Thu, 24 Sep 2009 09:25:20 +0100 Subject: [Freeswitch-users] ASR using mod_unimrcp, vlingo Message-ID: Hi All Has anyone had any experience doing ASR with mod_unimrcp in javascript? In particular, how do you deal with grammars? A simple piece of demo code would be massively appreciated - the documentation on mod_unimrcp ASR javascript bindings is TBD, which I assume means 'to be documented' ... unless it means 'to be developed' ... Also is anyone aware of a vlingo integration for freeswitch? All the best Jim From aep.lists at it46.se Thu Sep 24 02:23:01 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Thu, 24 Sep 2009 11:23:01 +0200 Subject: [Freeswitch-users] checking subscribed/subscribers to events Message-ID: Hi, Is there any simple way to know: who is subscribed to certain events via ESL? check which events i have subscribed during a ESL session? control which events can one user subscribe? disable the subscription of certain events and not all at the same time? /aep -- Stopping junk mailers is good for the environment From aep.lists at it46.se Thu Sep 24 02:24:56 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Thu, 24 Sep 2009 11:24:56 +0200 Subject: [Freeswitch-users] Bind to more than one ethernet interface In-Reply-To: References: Message-ID: <651ae6e4db89db00f6426ad36ce63587.squirrel@correo.nodo50.org> If I am correct you need to create a sip profile per interface and hardcode/set the IP address of each interface correctly in the SIP RTP fields of the profile. Then you need to set carefully the correct NAT and auth options for each profile /aep -- Stopping junk mailers is good for the environment > Hello, > > I am trying to run FreeSwitch on a machine which has more than one > interface, all of them should be used for SIP. The FreeSwitch binds only > to > the first one. I tried setting bind_server_ip to either "auto" or 0.0.0.0 > but it doesn't help. > > Any idea what to do? > > Thanks! _Yehavi: > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From Claudio.Cavalera at italtel.it Thu Sep 24 02:34:33 2009 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Thu, 24 Sep 2009 11:34:33 +0200 Subject: [Freeswitch-users] Freeswitch on embedded device: interesting or not? Message-ID: Hello guys, lately I've been trying to compile Freeswitch for a MIPS architecture. With the help of the community I've understood that my target architecture was wrong because of limitations in the SDK toolchain's. I'm not writing now to get help but to start (I hope) a discussion. I would like to understand your points of view about the general idea of porting FS on embedded devices. I'm not hardware expert at all; someone says that porting FS to any appliance which is not x86 based is a loss of time, because ARM and MIPS processors just lack computational power, this could be true, but maybe it depends on what you expect FS to do on such an embedded architecture. We are now all used to the amazing performance of FS on multicores 64bit cpus but still the one line description of FS is: "FreeSWITCH is an open source telephony platform designed to facilitate the creation of voice and chat driven products scaling from a soft-phone up to a soft-switch". Therefore a scaled down FS could be done, do you think is interesting? I'm not speaking here from a technical point of view, I know others have already compiled FS for ARM and MIPS and their experience is on the wiki. Would you consider a scaled down FS only for x86 architectures (e.g. the pfSense package or Atom)? Regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From dome at tel.co.th Thu Sep 24 03:09:22 2009 From: dome at tel.co.th (Dome Charoenyost) Date: Thu, 24 Sep 2009 17:09:22 +0700 Subject: [Freeswitch-users] Looking For A-Z Message-ID: <8ccbff060909240309s4b43d014w1b5cf270edafefa7@mail.gmail.com> Dear Sir, I'm looking for A-Z price and quality should be same http://voicetrading.com. Now i use http://voicetrading.com it's good quality but very bad support. some time i can payment by credit card, paypal some time can't i don't know why. BG Dome C. From tculjaga at gmail.com Thu Sep 24 03:10:17 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 24 Sep 2009 12:10:17 +0200 Subject: [Freeswitch-users] mod_cdr_csv missing timestamps in A-LEG Message-ID: <65d96fc80909240310q31cf2cbby1ed82a66d6555ce5@mail.gmail.com> hello, i'm on latest trunk and for some reason i cannot get timestamps dumped in my cdrs. I use mod_cdr_csv with default settings plus i enabled to get both a and b legs dumped. cdr_csv.conf.xml: call flow is the following: CALLER => FS => CALLED FS answers the call from CALLER, plays an announcement and bridges towards CALLED. I get different behavior when the call is released by Caller and by Called. Released by Caller: the CDR is ok having all timestamps OK CDR: Outbound LEG => "016659280","016659280","0914392122","public","2009-09-24 12:02:48","2009-09-24 12:02:54","2009-09-24 12:03:01","13","7","NORMAL_CLEARING","699cc2d0-a8f1-11de-962a-e328afdb9d8d","","","PCMA","PCMA" Inbound LEG => "016659280","016659280","05000403","public","2009-09-24 12:02:27","2009-09-24 12:02:41","2009-09-24 12:03:01","34","20","NORMAL_CLEARING","5d530192-a8f1-11de-962a-e328afdb9d8d","699cc2d0-a8f1-11de-962a-e328afdb9d8d","","PCMA","PCMA" Released by Called: the CDR is NOT OK as timestamps are missing NOT OK CDR: Inbound LEG => "016659280","016659280","0914392122","public","2009-09-24 12:05:20","2009-09-24 12:05:30","2009-09-24 12:05:39","19","9","NORMAL_CLEARING","c479411a-a8f1-11de-962a-e328afdb9d8d","","","PCMA","PCMA" Outbound LEG =>"016659280","016659280","015000403","public",*"","","",* "0","0","NORMAL_CLEARING","b82f2046-a8f1-11de-962a-e328afdb9d8d","c479411a-a8f1-11de-962a-e328afdb9d8d","","PCMA","PCMA" What can be wrong? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/5a702802/attachment-0001.html From aep.lists at it46.se Thu Sep 24 04:18:47 2009 From: aep.lists at it46.se (Alberto Escudero) Date: Thu, 24 Sep 2009 13:18:47 +0200 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <4ABA9AF3.30806@gmail.com> References: <4ABA47FB.2050100@gmail.com> <200909240857.02752.hads@nice.net.nz> <20090923211822.GW30343@base.carmickle.com> <200909240933.33422.hads@nice.net.nz> <4ABA9AF3.30806@gmail.com> Message-ID: <9b7e814b1c060bd9215292de644fb6a3.squirrel@correo.nodo50.org> Hi, Now I seem to reach the webserver. How do i checkout a local copy to run the builder? /aep -- Stopping junk mailers is good for the environment > It seems I had a port forwarded incorrectly for the external access to > the git web interface. here it is again: > > http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/ > > I've tested it to work now. > > -William King > > Hadley Rich wrote: >> On Thu, 24 Sep 2009 09:18:23 Frank Carmickle wrote: >> >>> Currently it's /opt/freeswitch. I would like to see it move to FHS >>> correct >>> locations for inclusion in to debian/ubuntu. This is the next bit >>> that I >>> will be working on. >>> >> >> Yeah, the FHS stuff was the bit that I got a little stuck on a while >> back. >> >> >>> Of course we also hope that the debian voip team will pick it >>> up once we've cleaned it up. >>> >> >> Sounds good. >> >> >>> I am not an ubuntu guy so I can't speak to that. I would say that most >>> of >>> the licenses of the included packages would allow for inclusion in >>> debian >>> main. Things like the cepstral support would have to go in to >>> contrib. >>> >> >> Gotcha, multiverse is for "not free" software, so anything that can go >> into >> main in Debian could go into universe in Ubuntu. >> >> hads >> > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From helmut.kuper at ewetel.de Thu Sep 24 04:50:29 2009 From: helmut.kuper at ewetel.de (Helmut Kuper) Date: Thu, 24 Sep 2009 13:50:29 +0200 Subject: [Freeswitch-users] A little Q931 tool Message-ID: <4ABB5D05.4050908@ewetel.de> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I ported my perl based FS-Logfile-Q931-HexDumps-to-pcap script to C (linux). It reads FS's logfile (loglevel DEBUG) grabs the Q931 hex dumps and puts them in a .pcap file which is directly readable and decodeable by Wireshark/tshark. The big plus here is, that you are able to get an isdn Q931 trace even of a call in the past (as long as you have FS in DEBUG loglevel). regards Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (MingW32) iD8DBQFKu10F4tZeNddg3dwRAjV3AKCltpyAhdqB0usC4Z2AFReRNUj/5ACfTmyH y6O0PXB/IFzXlSFpPQN13JA= =OWEy -----END PGP SIGNATURE----- From brian at freeswitch.org Thu Sep 24 05:58:22 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 24 Sep 2009 07:58:22 -0500 Subject: [Freeswitch-users] Looking For A-Z In-Reply-To: <8ccbff060909240309s4b43d014w1b5cf270edafefa7@mail.gmail.com> References: <8ccbff060909240309s4b43d014w1b5cf270edafefa7@mail.gmail.com> Message-ID: This belongs on freeswitch-biz /b On Sep 24, 2009, at 5:09 AM, Dome Charoenyost wrote: > Dear Sir, > I'm looking for A-Z price and quality should be same > http://voicetrading.com. Now i use http://voicetrading.com it's good > quality but very bad support. some time i can payment by credit card, > paypal some time can't i don't know why. > > BG > > Dome C. > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From brian at freeswitch.org Thu Sep 24 06:09:23 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 24 Sep 2009 08:09:23 -0500 Subject: [Freeswitch-users] How to play more than one voice file in play_and_get_digits function In-Reply-To: <87f2f3b90909240002t274128bdha650aa1b64390bf2@mail.gmail.com> References: <1452e2980909232338k3434c32ch7e0fe8b9bf3d405b@mail.gmail.com> <87f2f3b90909240002t274128bdha650aa1b64390bf2@mail.gmail.com> Message-ID: You can also use phrase macros. (and no its not just for TTS ;) ) /b On Sep 24, 2009, at 2:02 AM, Michael Collins wrote: > Make sure that mod_file_string is built and loaded and then try the > syntax that is described here: > http://wiki.freeswitch.org/wiki/Mod_file_string#Examples > > Instead of a comma separated list you can use ! and be sure NOT to > put a space after the ! because the function delimits the arguments > with spaces. Try something like this: > > $conn->execute("play_and_get_digits", 1 1 1000 # /usr/local/ > freeswitch/en/us/callie/sounds/ivr/please.wav!/usr/local/freeswitch/ > en/us/callie/sounds/ivr/press-1.wav /usr/loca/freeswitch/en/us/ > callie/sounds/ivr/invalid.wav res \\d+); > > Let us know how it goes. > -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/76411440/attachment.html From costa.zikalala at gmail.com Thu Sep 24 04:02:08 2009 From: costa.zikalala at gmail.com (Costa Zikalala) Date: Thu, 24 Sep 2009 13:02:08 +0200 Subject: [Freeswitch-users] User Creation with DB in Freeswitch In-Reply-To: <903da5680909232236v21cd9f47xa2aaa83749a16fab@mail.gmail.com> References: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> <566E110B-72CC-428E-AEE8-94AB600587B0@enterux.com> <347D3032-F778-46F4-AECE-94DECD27E849@freeswitch.org> <1b2118200909231137w79ea33afq6050133fb50b32a0@mail.gmail.com> <903da5680909232236v21cd9f47xa2aaa83749a16fab@mail.gmail.com> Message-ID: <59daa2cd0909240402r1fe089e3r52414a2ac41c255a@mail.gmail.com> Hi Gabe Thanks for you response to this question. Do you perhaps have a link to an example (or just further detail) to what you've descibed below. I guess one would also use a similar setup to generate dialplans from web forms. Thanks again, Costa 2009/9/24 Gabriel Gunderson > On Wed, Sep 23, 2009 at 12:37 PM, Anil Kumar S. R. > wrote: > > I didn't get much help for my problem with XML CURL. What I meant to say > is, > > suppose I want to have some 10000 users on freeswitch. Do we have to > create > > some many xml files in the directory or is there some way in which the > users > > can be put in the db ? > > That's the whole point. You serve up the XML from *your* web server > using whatever technologies that *you* would like on the back-end. > You'll want to make that XML reply dynamic. > > Use php, perl, python, c#, java or whatever other language *you* want. > Pull the data from MySQL, flat-files, PostgreSQL, MSSQL, LDAP or > whatever *you* want. Just serve up the right bit of configuration to > FreeSWITCH and you're done. > > Good luck. Spend more time in the docs. Others have posted the links. > > > Best, > Gabe > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/a6a80684/attachment.html From frank at carmickle.com Thu Sep 24 06:59:12 2009 From: frank at carmickle.com (Frank Carmickle) Date: Thu, 24 Sep 2009 09:59:12 -0400 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <9b7e814b1c060bd9215292de644fb6a3.squirrel@correo.nodo50.org> References: <4ABA47FB.2050100@gmail.com> <200909240857.02752.hads@nice.net.nz> <20090923211822.GW30343@base.carmickle.com> <200909240933.33422.hads@nice.net.nz> <4ABA9AF3.30806@gmail.com> <9b7e814b1c060bd9215292de644fb6a3.squirrel@correo.nodo50.org> Message-ID: <20090924135911.GB30343@base.carmickle.com> On Thu, Sep 24, Alberto Escudero wrote: > Hi, > > Now I seem to reach the webserver. How do i checkout a local copy to run > the builder? If you just want to build then you can put a line like deb-src http://ppa.launchpad.net/pbxbuntu-drivers/ppa/ubuntu jaunty main in your sources.list replace jaunty with what ever your running. Then make sure you are in a directory where you want the source and apt-get update -f ; apt-get source freeswitch HTH --FC From david.nembrot at sogeti.com Thu Sep 24 07:39:26 2009 From: david.nembrot at sogeti.com (David Nembrot) Date: Thu, 24 Sep 2009 16:39:26 +0200 Subject: [Freeswitch-users] Intradomain File Transfer - Unsupported Format Message-ID: <20090924163926.o8h9c2srg0swwgo8@mail.sogeti.com> Hello list, I'm currently testing file transfer within the same SIP domain and the situation has just got odd! When I send a PDF or MP3 file, Freeswitch allows its transfer (meaning SIP Traffic is okay: INVITE, 180 Ringing, 200 OK, SEND Transaction, BYE, 200 OK) But when I try to send JPEG files, Freeswitch answers to the INVITE message with 415 Unsupported Media Type...so there is no SEND transaction! The fact is that I've checked the supported media types list in the config file 'mime.types' and I positively found the line: 'image/jpeg jpeg jpg jpe' There seems to be something wrong in FS, and well actually I don't know how to get rid of this Freeswitch's negative reply... Could you help me validating such transfer please? Thanks in advance David N. From anthony.minessale at gmail.com Thu Sep 24 08:07:20 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 24 Sep 2009 10:07:20 -0500 Subject: [Freeswitch-users] Intradomain File Transfer - Unsupported Format In-Reply-To: <20090924163926.o8h9c2srg0swwgo8@mail.sogeti.com> References: <20090924163926.o8h9c2srg0swwgo8@mail.sogeti.com> Message-ID: <191c3a030909240807y5d43301ci58fc07ad760ed265@mail.gmail.com> mime.types file is for http server stuff not SIP we have never even tried to support file transfer over SIP, it's a feature request at this point. On Thu, Sep 24, 2009 at 9:39 AM, David Nembrot wrote: > Hello list, > > I'm currently testing file transfer within the same SIP domain and > the situation has just got odd! When I send a PDF or MP3 file, > Freeswitch allows its transfer (meaning SIP Traffic is okay: > INVITE, 180 Ringing, 200 OK, SEND Transaction, BYE, 200 OK) > But when I try to send JPEG files, Freeswitch answers to the > INVITE message with 415 Unsupported Media Type...so there is > no SEND transaction! > > The fact is that I've checked the supported media types list in > the config file 'mime.types' and I positively found the line: > 'image/jpeg jpeg jpg jpe' > > There seems to be something wrong in FS, and well actually > I don't know how to get rid of this Freeswitch's negative reply... > Could you help me validating such transfer please? > > Thanks in advance > > David N. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/ca9c234f/attachment-0001.html From msc at freeswitch.org Thu Sep 24 08:40:16 2009 From: msc at freeswitch.org (msc) Date: Thu, 24 Sep 2009 08:40:16 -0700 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <11372C8B9E603F4FACDE6AB18256DEC601D8BC94@srvmtel.office.mtel.nl> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <11372C8B9E603F4FACDE6AB18256DEC695A9EC@srvmtel.office.mtel.nl> <87f2f3b90909171357y40fe0db4kef1d59fb86e9790c@mail.gmail.com> <11372C8B9E603F4FACDE6AB18256DEC601D8BC94@srvmtel.office.mtel.nl> Message-ID: <87f2f3b90909240840u74a322d4m33255c46244b17b5@mail.gmail.com> On Thu, Sep 24, 2009 at 1:09 AM, Remko Kloosterman wrote: > Hello Michael, > > > > Do you still want to follow up on this? I?m having difficulty gathering the > old stuff in an understandable form. Also, it looks like the open source ACD > Spice Telephony by Andrew Thompson can do just what you might need. > > I had totally forgotten about Andrew's stuff! Unless people want to build their own 100% community/free/DIY version of a skill-based ACD then I say let's all play with SpiceCSM and help improve it. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/3ffc42d4/attachment.html From msc at freeswitch.org Thu Sep 24 09:03:49 2009 From: msc at freeswitch.org (msc) Date: Thu, 24 Sep 2009 09:03:49 -0700 Subject: [Freeswitch-users] Event-Name=CHANNEL_DATA - duplicate variables In-Reply-To: <65d96fc80909230715m43ff6774k3fb495277743425f@mail.gmail.com> References: <65d96fc80909230715m43ff6774k3fb495277743425f@mail.gmail.com> Message-ID: <87f2f3b90909240903p67b33aew98f2956a53e4c461@mail.gmail.com> Can you move this question over to the -dev list? Pretty much anything dealing with source code and custom modules will be better served being discussed by the developers. Thanks! -MC On Wed, Sep 23, 2009 at 7:15 AM, Tihomir Culjaga wrote: > > Hello, > > the setup is like this: > > > CALLING_USER(context:Public) => FS => ENDPOINT > > > > > The CALLING_USER places a call towards FS. The call is answered and a > welcome prompt is being played. After that, FS bridges the call towards an > ENDPOINT. > > I've built a custom module where among other stuff i subscribe to events... > > > > switch_core_add_state_handler(&albatross_state_handler); > > switch_state_handler_table_t albatross_state_handler = > { > /* on_init */ process_init, > /* on_routing */ NULL, /* Need to add a check here for anything in > their account before routing */ > /* on_execute */ NULL, /* Turn on heartbeat for this session and do an > initial account check */ > /* on_hangup */ process_hangup, /* On hangup - most important place to > go bill */ > /* on_exch_media */ NULL, > /* on_soft_exec */ NULL, > /* on_consume_med */ NULL, > /* on_hibernate */ NULL, > /* on_reset */ NULL > }; > > > > So, according to the callflow specified above, on hangup i get two events > ... (1st B-LEG followed by A-LEG). > > > well .. it sems i have double variables on the A-LEG ?!?!?! .. whats wrong > hewre ? > > > variable_duration=0 > variable_billsec=0 > variable_progresssec=0 > variable_answersec=0 > variable_progress_mediasec=0 > variable_flow_billsec=0 > variable_mduration=0 > variable_billmsec=0 > variable_progressmsec=0 > variable_answermsec=0 > variable_progress_mediamsec=0 > variable_flow_billmsec=0 > variable_uduration=0 > variable_billusec=0 > variable_progressusec=0 > variable_answerusec=0 > variable_progress_mediausec=0 > variable_flow_billusec=0 > > > > > Dump:** > > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Event-Name=CHANNEL_DATA *<= B_LEG (Outbound call)* > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Core-UUID=ca97d306-a840-11de-8ae7-5585e27e6446 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > FreeSWITCH-Hostname=node1 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > FreeSWITCH-IPv4=192.168.102.81 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 FreeSWITCH-IPv6=::1 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Event-Date-Local=2009-09-23 14:59:22 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Date-GMT=Wed, > 23 Sep 2009 12:59:22 GMT > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Event-Date-Timestamp=1253710762323094 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Event-Calling-File=mod_albatross.c > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Event-Calling-Function=process_hangup > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Event-Calling-Line-Number=2366 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-State=CS_HANGUP > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-State-Number=10 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-Name=sofia/external/0914392122 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Unique-ID=de716bd0-a840-11de-8ae7-5585e27e6446 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Call-Direction=outbound > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Presence-Call-Direction=outbound > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Answer-State=answered > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-Read-Codec-Name=PCMA > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-Read-Codec-Rate=8000 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-Write-Codec-Name=PCMA > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-Write-Codec-Rate=8000 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Username=016659280 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Dialplan=XML > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Caller-ID-Name=016659280 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Caller-ID-Number=016659280 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Network-Addr=172.16.1.20 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Destination-Number=0914392122 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Unique-ID=de716bd0-a840-11de-8ae7-5585e27e6446 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Source=mod_sofia > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Context=public > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Channel-Name=sofia/external/0914392122 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Profile-Index=1 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Profile-Created-Time=1253710743103107 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Channel-Created-Time=1253710743103107 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Channel-Answered-Time=1253710748807099 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Channel-Progress-Time=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Channel-Progress-Media-Time=1253710745451100 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Channel-Hangup-Time=1253710762323094 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Channel-Transfer-Time=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Screen-Bit=true > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Privacy-Hide-Name=false > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Privacy-Hide-Number=false > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Username=016659280 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Dialplan=XML > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Caller-ID-Name=016659280 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Caller-ID-Number=016659280 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Network-Addr=172.16.1.20 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Destination-Number=012468601_Kviz > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Unique-ID=d2242912-a840-11de-8ae7-5585e27e6446 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Source=mod_sofia > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Context=public > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Channel-Name=sofia/external/016659280 at sip-priv.amis.hr > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Profile-Created-Time=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Channel-Created-Time=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Channel-Answered-Time=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Channel-Progress-Time=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Channel-Progress-Media-Time=1253710745451100 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Channel-Hangup-Time=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Channel-Transfer-Time=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Screen-Bit=true > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Privacy-Hide-Name=false > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Other-Leg-Privacy-Hide-Number=false > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_gateway_name=gw1 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_channel_name=sofia/external/0914392122 > 2009-09-23 14:59:22.323094 [NOTICE] mod_albatross.c:2320 *** HANGUP > RECEIVED, BYE... > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_destination_url=sip:0914392122 at 172.16.1.20 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_is_outbound=true > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_h_P-Access-Network-Info=ADSL;dsl_location="NOA=4;APRI=1;ADD=3851";network-provided > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_cid_type=pid > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_max_forwards=30 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_originator_codec=PCMA at 8000h@20i > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_originator=d2242912-a840-11de-8ae7-5585e27e6446 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_execute_on_answer=updateQuizServiceStatus_ch in 012468601, in > connected > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_switch_m_sdp=v=0 > o=cp10 125371067255 125371067255 IN IP4 172.16.2.42 > s=SIP Call > c=IN IP4 172.16.2.42 > t=0 0 > m=audio 34474 RTP/AVP 8 0 18 125 101 > b=AS:64 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:18 G729/8000/1 > a=fmtp:18 annexb=no > a=rtpmap:125 CLEARMODE/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_service_number=012468601 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_dialed_number=0914392122 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_originate_early_media=true > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sofia_profile_name=external > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_call_id=b5cfd0c9-22e3-122d-2c98-001e684a25c5 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_switch_r_sdp=v=0 > o=cp10 125371069278 125371069279 IN IP4 172.16.2.42 > s=SIP Call > c=IN IP4 172.16.2.42 > t=0 0 > m=audio 34972 RTP/AVP 8 0 > b=AS:64 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:0 PCMU/8000/1 > a=ptime:20 > > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_remote_media_ip=172.16.2.42 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_remote_media_port=34972 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_read_codec=PCMA > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_read_rate=8000 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Event-Name=CHANNEL_DATA *<= A-LEG (Inbound)* > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Core-UUID=ca97d306-a840-11de-8ae7-5585e27e6446 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > FreeSWITCH-Hostname=node1 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > FreeSWITCH-IPv4=192.168.102.81 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 FreeSWITCH-IPv6=::1 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Event-Date-Local=2009-09-23 14:59:22 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Date-GMT=Wed, > 23 Sep 2009 12:59:22 GMT > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Event-Date-Timestamp=1253710762323094 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Event-Calling-File=mod_albatross.c > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Event-Calling-Function=process_hangup > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_write_codec=PCMA > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_write_rate=8000 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_local_media_ip=172.16.3.2 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_local_media_port=42684 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_bridge_channel=sofia/external/016659280 at sip-priv.amis.hr > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_bridge_uuid=d2242912-a840-11de-8ae7-5585e27e6446 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_signal_bond=d2242912-a840-11de-8ae7-5585e27e6446 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_reply_host=172.16.1.20 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_reply_port=5060 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_endpoint_disposition=ANSWER > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_current_application_data=in 012468601, in connected > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_current_application=updateQuizServiceStatus_ch > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_term_status=200 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_term_cause=16 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_user_agent=Cirpack/v4.42d (gw_sip) > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_hangup_disposition=recv_bye > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_hangup_cause=NORMAL_CLEARING > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_hangup_cause_q850=16 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_digits_dialed=none > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_start_stamp=2009-09-23 14:59:03 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_profile_start_stamp=2009-09-23 14:59:03 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_answer_stamp=2009-09-23 14:59:08 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progress_media_stamp=2009-09-23 14:59:05 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_end_stamp=2009-09-23 14:59:22 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_start_epoch=1253710743 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_start_uepoch=1253710743103107 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_profile_start_epoch=1253710743 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_profile_start_uepoch=1253710743103107 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_answer_epoch=1253710748 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_answer_uepoch=1253710748807099 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progress_epoch=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progress_uepoch=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progress_media_epoch=1253710745 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progress_media_uepoch=1253710745451100 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_end_epoch=1253710762 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_end_uepoch=1253710762323094 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_last_app=updateQuizServiceStatus_ch > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_last_arg=in > 012468601, in connected > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Event-Calling-Line-Number=2366 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_caller_id="016659280" <016659280> > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_duration=19 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-State=CS_HANGUP > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_billsec=14 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-State-Number=10 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progresssec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-Name=sofia/external/016659280 at sip-priv.amis.hr > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_answersec=5 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progress_mediasec=2 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Unique-ID=d2242912-a840-11de-8ae7-5585e27e6446 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_flow_billsec=19 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Call-Direction=inbound > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_mduration=19220 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Presence-Call-Direction=inbound > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_billmsec=13516 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Answer-State=answered > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progressmsec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-Read-Codec-Name=PCMA > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_answermsec=5704 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-Read-Codec-Rate=8000 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progress_mediamsec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-Write-Codec-Name=PCMA > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_flow_billmsec=19220 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Channel-Write-Codec-Rate=8000 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_uduration=19219987 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_billusec=13515995 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Username=016659280 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progressusec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Dialplan=XML > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_answerusec=5703992 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progress_mediausec=2347993 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Caller-ID-Name=016659280 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_flow_billusec=19219987 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Caller-ID-Number=016659280 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_raw_bytes=144996 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_media_bytes=144996 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Network-Addr=172.16.1.20 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_packet_count=843 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Destination-Number=012468601_Kviz > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_media_packet_count=843 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Unique-ID=d2242912-a840-11de-8ae7-5585e27e6446 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_skip_packet_count=1 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Source=mod_sofia > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_jb_packet_count=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Context=public > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_dtmf_packet_count=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Channel-Name=sofia/external/016659280 at sip-priv.amis.hr > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_cng_packet_count=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Screen-Bit=true > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_flush_packet_count=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Privacy-Hide-Name=false > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_raw_bytes=143964 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > Caller-Privacy-Hide-Number=false > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_media_bytes=143964 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_received_ip=172.16.1.20 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_packet_count=837 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_received_port=5060 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_media_packet_count=837 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_via_protocol=udp > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_skip_packet_count=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_from_params=user=phone > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_dtmf_packet_count=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_from_user=016659280 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_cng_packet_count=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_from_uri=016659280 at sip-priv.amis.hr > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_from_host=sip-priv.amis.hr > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_from_user_stripped=016659280 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_from_tag=11746-FX-007c2917-23e2091e3 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sofia_profile_name=external > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_P-Asserted-Identity=016659280 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_cid_type=pid > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2407 siId: (null) > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_req_params=user=phone > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2408 ani: 016659280 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_req_user=012468601 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2409 dnis: 0914392122 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_req_port=5060 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2410 dialed_number: > 0914392122 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2412 start_stamp: > 2009-09-23 14:59:03 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_req_uri=012468601 at 172.16.3.2:5060 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2413 answer_stamp: > 2009-09-23 14:59:08 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2414 progress_stamp: > (null) > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_req_host=172.16.3.2 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_to_params=user=phone > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2415 > progress_media_stamp: 2009-09-23 14:59:05 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_to_user=012468601 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2416 end_stamp: > 2009-09-23 14:59:22 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_sip_to_uri= > 012468601 at 172.16.3.2 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2420 privacy_hide_number: > false > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_to_host=172.16.3.2 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_contact_user=nobody > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_contact_port=5060 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_contact_uri=nobody at 172.16.1.20:5060 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_contact_host=172.16.1.20 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_channel_name=sofia/external/016659280 at sip-priv.amis.hr > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_call_id=11746-QJ-007c2916-59db25232 at sip-priv.amis.hr > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_user_agent=Cirpack/v4.42d (gw_sip) > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_via_host=172.16.1.20 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_via_port=5060 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_max_forwards=31 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_h_P-Access-Network-Info=ADSL;dsl_location="NOA=4;APRI=1;ADD=3851";network-provided > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_switch_r_sdp=v=0 > o=cp10 125371067255 125371067255 IN IP4 172.16.2.42 > s=SIP Call > c=IN IP4 172.16.2.42 > t=0 0 > m=audio 34474 RTP/AVP 8 0 18 125 101 > b=AS:64 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:0 PCMU/8000/1 > a=rtpmap:18 G729/8000/1 > a=fmtp:18 annexb=no > a=rtpmap:125 CLEARMODE/8000/1 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-15 > a=ptime:20 > > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_remote_media_ip=172.16.2.42 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_remote_media_port=34474 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_read_codec=PCMA > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_read_rate=8000 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_write_codec=PCMA > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_write_rate=8000 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_service_type_id=1 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_local_media_ip=172.16.3.2 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_local_media_port=27686 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_price_prompt=3.66kn_novo_upozorenje.wav > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_dialed_number=012468601 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_bNum=012468601 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_service_status1=win > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_number_2_connect=0914392122 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_next_number_2_connect=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_next_number_2_display=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_service_instance=130 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_service_id=2 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_quiz_status=win > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_not_working_prompt=2/emisija trenutno nije u tijeku-za telefone.wav > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_service_win_prompt=2/bit cete spojeni u emisiju1.wav > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_service_loose_prompt=2/zovi ponovo.wav > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_outside_call=true > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_endpoint_disposition=ANSWER > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_playback_ms=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_playback_samples=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_bypass_media=false > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_hangup_after_bridge=true > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_nolocal:execute_on_answer=updateQuizServiceStatus_ch in 012468601, > in connected > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_export_vars=nolocal:execute_on_answer > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_current_application_data=[service_number=012468601,dialed_number=0914392122]sofia/gateway/gw1/0914392122 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_current_application=bridge > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_switch_m_sdp=v=0 > o=cp10 125371069278 125371069279 IN IP4 172.16.2.42 > s=SIP Call > c=IN IP4 172.16.2.42 > t=0 0 > m=audio 34972 RTP/AVP 8 0 > b=AS:64 > a=rtpmap:8 PCMA/8000/1 > a=rtpmap:0 PCMU/8000/1 > a=ptime:20 > > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_originate_disposition=SUCCESS > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_bridge_channel=sofia/external/0914392122 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_bridge_uuid=de716bd0-a840-11de-8ae7-5585e27e6446 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_signal_bond=de716bd0-a840-11de-8ae7-5585e27e6446 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_hangup_phrase=OK > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_bridge_hangup_cause=NORMAL_CLEARING > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_hangup_cause=NORMAL_CLEARING > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_hangup_cause_q850=16 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_digits_dialed=none > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_last_app=bridge > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_last_arg=[service_number=012468601,dialed_number=0914392122]sofia/gateway/gw1/0914392122 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_caller_id="016659280" <016659280> > *2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_duration=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_billsec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progresssec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_answersec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progress_mediasec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_flow_billsec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_mduration=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_billmsec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progressmsec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_answermsec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progress_mediamsec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_flow_billmsec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_uduration=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_billusec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progressusec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_answerusec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_progress_mediausec=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_flow_billusec=0* > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_sip_hangup_disposition=send_bye > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_raw_bytes=341764 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_media_bytes=341764 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_packet_count=1987 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_media_packet_count=1987 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_skip_packet_count=10 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_jb_packet_count=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_dtmf_packet_count=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_cng_packet_count=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_in_flush_packet_count=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_raw_bytes=304440 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_media_bytes=304440 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_packet_count=1770 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_media_packet_count=1770 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_skip_packet_count=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_dtmf_packet_count=0 > 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 > variable_rtp_audio_out_cng_packet_count=0 > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/9217fa1e/attachment-0001.html From msc at freeswitch.org Thu Sep 24 09:08:28 2009 From: msc at freeswitch.org (msc) Date: Thu, 24 Sep 2009 09:08:28 -0700 Subject: [Freeswitch-users] No ring tone while recording incoming call. Please help. In-Reply-To: <94790b850909230734x1d791927ifb0a52a3b1d1d2f1@mail.gmail.com> References: <94790b850909230734x1d791927ifb0a52a3b1d1d2f1@mail.gmail.com> Message-ID: <87f2f3b90909240908j2c0c512en46bd9bc5a193eeab@mail.gmail.com> On Wed, Sep 23, 2009 at 7:34 AM, Svetik VOIP wrote: > Brian, > > Thank yo very much for your reply. > > I have tried to add transfer_ringback action, but it did not solve my > problem. > Destination phone is ringing, but the person who is calling does not hear > ringing tone in hte handset. > > Is there anything in the logfile that can help you to identify the problem? > What kind of system is the calling party connected to? It looks like a 180 is sent out by FS: 2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/ sip:main at 192.168.0.121:5060 entering state [proceeding][180] At that point the server at the originating side *should* generate pretend ringing for the calling phone. If that is not happening then you need to see what's going on at the originating side. Is it a SIP provider? -MC > > Closest I can see is: > 2009-09-22 17:18:05.444402 [DEBUG] switch_ivr_originate.c:1738 Raw Codec > Activation Success L16 at 8000hz 1 channel 20ms > 2009-09-22 17:18:05.444402 [DEBUG] switch_ivr_originate.c:1797 Play > Ringback Tone [%(2000,4000,440.0,480.0)] > 2009-09-22 17:18:05.447237 [DEBUG] switch_core_io.c:232 sofia/external/ > 4163641113 at 67.205.74.164 receive message [TRANSCODING_NECESSARY] > 2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/ > sip:main at 192.168.0.121:5060 entering state [proceeding][180] > 2009-09-22 17:18:05.463192 [NOTICE] sofia.c:3376 Ring-Ready sofia/internal/ > sip:main at 192.168.0.121:5060! > 2009-09-22 17:18:14.739182 [DEBUG] sofia.c:3312 Channel sofia/external/ > 4163641113 at 67.205.74.164 entering state [terminated][487] > 2009-09-22 17:18:14.739182 [NOTICE] sofia.c:3873 Hangup sofia/external/ > 4163641113 at 67.205.74.164 [CS_EXECUTE] [ORIGINATOR_CANCEL] > > Thank you, > > Igor > > >set ringback before record_session and also set transfer_ringback > >because record_session causes an pre-answer. > > > >/b > > > >On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote: > > > >> Hi, > >> > >> I have trouble recording incoming calls with FreeSwitch. > >> > >> I have followed the instruction from Misc. Dialplan Tools record > >> session > >> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session) > >> It works well for outgoing calls, but I have the problem with > >> incoming calls. > >> > >> The person who is calling does not hear ring tone, he hears just the > >> silence until > >> I pick up the phone. Everything else is working, we can talk, > >> conversation is recorded. > >> > >> Here is a copy of my dialplan for incoming calls > >> /usr/local/freeswitch/conf/dialplan/public/voipms.xml > >> > >> > >> > >> >> expression="XXXXXXXXXX"> > >> > >> > >> >> data="RECORD_SOFTWARE=FreeSwitch"/> > >> >> data="RECORD_ARTIST=FreeSwitch"/> > >> >> data="RECORD_COMMENT=FreeSwitch"/> > >> > >> > >> > >> > >> > >> > >> > >> > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/a1a7cb05/attachment.html From achaloyan at yahoo.com Thu Sep 24 09:23:32 2009 From: achaloyan at yahoo.com (Arsen Chaloyan) Date: Thu, 24 Sep 2009 09:23:32 -0700 (PDT) Subject: [Freeswitch-users] ASR using mod_unimrcp, vlingo In-Reply-To: References: Message-ID: <217503.51780.qm@web111306.mail.gq1.yahoo.com> Hi Jim, >From conceptual viewpoint, mod_unimrcp is just an alternate implementation of an abstract ASR/TTS interface FreeSWITCH provides. Therefore you can use it exactly the same way as other ASR/TTS modules. See scripts/javascript/ps_pizza.js in FS tree for a working example. The only thing you should know and change there is module name < var asr = new SpeechDetect(session, "pocketsphinx"); > var asr = new SpeechDetect(session, "unimrcp"); Typically you can specify any grammar your MRCP server supports. ________________________________ From: Jim Page To: "freeswitch-users at lists.freeswitch.org" Sent: Thursday, September 24, 2009 1:25:20 PM Subject: [Freeswitch-users] ASR using mod_unimrcp, vlingo Hi All Has anyone had any experience doing ASR with mod_unimrcp in javascript? In particular, how do you deal with grammars? A simple piece of demo code would be massively appreciated - the documentation on mod_unimrcp ASR javascript bindings is TBD, which I assume means 'to be documented' ... unless it means 'to be developed' ... Also is anyone aware of a vlingo integration for freeswitch? All the best Jim _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/d2bf6ade/attachment.html From chris at cloudtel.com Thu Sep 24 09:43:52 2009 From: chris at cloudtel.com (Chris Burns) Date: Thu, 24 Sep 2009 12:43:52 -0400 Subject: [Freeswitch-users] Polycom MWI Forgetfulness In-Reply-To: References: <68755a9c0909231044ta70d850vab3ce7971992433d@mail.gmail.com> <68755a9c0909231638t5e39cfbkc0d1b48e0891d8d8@mail.gmail.com> Message-ID: <200909241243.52381.chris@cloudtel.com> This happens with our polycoms as well ... NAT on phone and PBX. Still haven't had time to look into it so I disabled the sound for new message waiting ... for now it doesn't keep beeping every few minutes. On September 23, 2009 08:08:39 pm Brian West wrote: > NO I have never seen it happen what firmware version are you running? > > /b > > On Sep 23, 2009, at 6:38 PM, Daniel Morrigan wrote: > > Brian, > > It was set for contact. Would that cause this behavior? > > > > Daniel > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From diego.viola at gmail.com Thu Sep 24 09:55:18 2009 From: diego.viola at gmail.com (Diego Viola) Date: Thu, 24 Sep 2009 16:55:18 +0000 Subject: [Freeswitch-users] How to play more than one voice file in play_and_get_digits function In-Reply-To: References: <1452e2980909232338k3434c32ch7e0fe8b9bf3d405b@mail.gmail.com> <87f2f3b90909240002t274128bdha650aa1b64390bf2@mail.gmail.com> Message-ID: <86a32abc0909240955i7afd40b5ub0333242d17202f2@mail.gmail.com> Use phrase macros as Brian said. On Thu, Sep 24, 2009 at 1:09 PM, Brian West wrote: > You can also use phrase macros. (and no its not just for TTS ;) ) > /b > > On Sep 24, 2009, at 2:02 AM, Michael Collins wrote: > > Make sure that mod_file_string is built and loaded and then try the syntax > that is described here: > http://wiki.freeswitch.org/wiki/Mod_file_string#Examples > > Instead of a comma separated list you can use ! and be sure NOT to put a > space after the ! because the function delimits the arguments with spaces. > Try something like this: > > $conn->execute("play_and_get_digits", 1 1 1000 # > /usr/local/freeswitch/en/us/callie/sounds/ivr/please.wav!/usr/local/freeswitch/en/us/callie/sounds/ivr/press-1.wav > /usr/loca/freeswitch/en/us/callie/sounds/ivr/invalid.wav res \\d+); > > Let us know how it goes. > -MC > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/dd3f7522/attachment.html From jim.page at redmatter.com Thu Sep 24 10:07:53 2009 From: jim.page at redmatter.com (Jim Page) Date: Thu, 24 Sep 2009 18:07:53 +0100 Subject: [Freeswitch-users] ASR using mod_unimrcp, vlingo In-Reply-To: <217503.51780.qm@web111306.mail.gq1.yahoo.com> References: <217503.51780.qm@web111306.mail.gq1.yahoo.com> Message-ID: Hi Arsen Thanks for your message - it inspired us to do what we should have done in the first place, and look at the code. The problem we were having was related to grammar files not being available locally. Now we have discovered the "builtin:" keyword we are up and running :) Many thanks Jim From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Arsen Chaloyan Sent: 24 September 2009 18:24 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] ASR using mod_unimrcp, vlingo Hi Jim, >From conceptual viewpoint, mod_unimrcp is just an alternate implementation of an abstract ASR/TTS interface FreeSWITCH provides. Therefore you can use it exactly the same way as other ASR/TTS modules. See scripts/javascript/ps_pizza.js in FS tree for a working example. The only thing you should know and change there is module name < var asr = new SpeechDetect(session, "pocketsphinx"); > var asr = new SpeechDetect(session, "unimrcp"); Typically you can specify any grammar your MRCP server supports. ________________________________ From: Jim Page To: "freeswitch-users at lists.freeswitch.org" Sent: Thursday, September 24, 2009 1:25:20 PM Subject: [Freeswitch-users] ASR using mod_unimrcp, vlingo Hi All Has anyone had any experience doing ASR with mod_unimrcp in javascript? In particular, how do you deal with grammars? A simple piece of demo code would be massively appreciated - the documentation on mod_unimrcp ASR javascript bindings is TBD, which I assume means 'to be documented' ... unless it means 'to be developed' ... Also is anyone aware of a vlingo integration for freeswitch? All the best Jim _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/7ee6c194/attachment-0001.html From msc at freeswitch.org Thu Sep 24 10:22:14 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 24 Sep 2009 10:22:14 -0700 Subject: [Freeswitch-users] A little Q931 tool In-Reply-To: <4ABB5D05.4050908@ewetel.de> References: <4ABB5D05.4050908@ewetel.de> Message-ID: <87f2f3b90909241022m5f3ba72bq64c2de9bd70fca2c@mail.gmail.com> Excellent, thanks! -MC On Thu, Sep 24, 2009 at 4:50 AM, Helmut Kuper wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Hello, > > I ported my perl based FS-Logfile-Q931-HexDumps-to-pcap script to C > (linux). It reads FS's logfile (loglevel DEBUG) grabs the Q931 hex dumps > and puts them in a .pcap file which is directly readable and decodeable > by Wireshark/tshark. > > The big plus here is, that you are able to get an isdn Q931 trace even > of a call in the past (as long as you have FS in DEBUG loglevel). > > > regards > Helmut > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v1.4.7 (MingW32) > > iD8DBQFKu10F4tZeNddg3dwRAjV3AKCltpyAhdqB0usC4Z2AFReRNUj/5ACfTmyH > y6O0PXB/IFzXlSFpPQN13JA= > =OWEy > -----END PGP SIGNATURE----- > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/512e6632/attachment.html From dmitry.bely at gmail.com Thu Sep 24 10:29:34 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Thu, 24 Sep 2009 21:29:34 +0400 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <4ABA9DC9.6090808@gmail.com> References: <4ABA47FB.2050100@gmail.com> <90823c940909231251k5f85e3dambb7506bdc62e62ac@mail.gmail.com> <20090923202524.GR30343@base.carmickle.com> <90823c940909231507ie10dc30x17442386f401c375@mail.gmail.com> <4ABA9DC9.6090808@gmail.com> Message-ID: <90823c940909241029w1bc6c302obf602f94a5654362@mail.gmail.com> On Thu, Sep 24, 2009 at 2:14 AM, William King wrote: > Sure, post it here and I'll add it in the next build in a few hours. See attached file. Unfortunately mod_skypiax author did not placed config files (skypiax.conf.xml, skypiax.X.conf) into conf/autoload_configs, so they are not included into freeswitch-config and should be added manually. BTW, why swig is a dependency for the source package? I recall Brian's post where he insists that swig is never needed to build Freeswitch. > -William King > > Dmitry Bely wrote: >> On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle wrote: >> >>> On Wed, Sep 23, Dmitry Bely wrote: >>> >>>> Can you enable mod_skypiax in your debian package? >>>> >>> We will be enabling as much as we can cleanly build on debian/ubuntu. ?There will be a lot more to come. ?We will be breaking the mods and end points in to different packages so that you can install what you like. ?If you have something you would like to see in the package let us know. ?Also patches are welcome. >>> >> >> Well, mod_skypiax just requires trivial one-line addition to >> debian/rules and debian/freeswitch.install. It builds OK. If the patch >> is required I can post it here. - Dmitry Bely -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.diff Type: application/octet-stream Size: 1207 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/4cf9c7ef/attachment.obj From msc at freeswitch.org Thu Sep 24 10:42:03 2009 From: msc at freeswitch.org (Michael Collins) Date: Thu, 24 Sep 2009 10:42:03 -0700 Subject: [Freeswitch-users] Bind to more than one ethernet interface In-Reply-To: <903da5680909232222n244b924bp4d875a6988e71733@mail.gmail.com> References: <23f91030909232144v2389170fx5556bb7b83b5e361@mail.gmail.com> <903da5680909232222n244b924bp4d875a6988e71733@mail.gmail.com> Message-ID: <87f2f3b90909241042m27da4a80m4dd67b7e0a4c3471@mail.gmail.com> On Wed, Sep 23, 2009 at 10:22 PM, Gabriel Gunderson wrote: > On Wed, Sep 23, 2009 at 10:44 PM, Seven Du wrote: > > It not possible to use 0.0.0.0 for on profile. however, you can create > more > > sip profiles for each of your interfaces. Search freeswitch-users > archievs > > then you will find similar topics. > > It sure would be nice to be able to provide a list of IPs in one > profile. Seems like I run into that need often. Instead, I end up > making two profiles that are mostly the same. > > Is there a reason why this is the way it is? > A SIP profile is a user agent. By design it services exactly one IP/Port combo. If you are finding that you need to have multiple IP's or ports in a single SIP profile then that suggests to me that you might be trying to use the wrong tool for the job. What are the scenarios where you feel you need a single profile to handle multiple IP addresses? My guess is that bkw and the gang will have suggestions for an elegant solution that will fit your needs. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/34d25ede/attachment.html From quentusrex at gmail.com Thu Sep 24 10:49:02 2009 From: quentusrex at gmail.com (William King) Date: Thu, 24 Sep 2009 10:49:02 -0700 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <90823c940909241029w1bc6c302obf602f94a5654362@mail.gmail.com> References: <4ABA47FB.2050100@gmail.com> <90823c940909231251k5f85e3dambb7506bdc62e62ac@mail.gmail.com> <20090923202524.GR30343@base.carmickle.com> <90823c940909231507ie10dc30x17442386f401c375@mail.gmail.com> <4ABA9DC9.6090808@gmail.com> <90823c940909241029w1bc6c302obf602f94a5654362@mail.gmail.com> Message-ID: <4ABBB10E.1080900@gmail.com> Hmm... That is interesting... swig is needed I believe only for the mod_perl or the esl modules. I'll find out more information and put it on the correct package. I will also update the mod_skypiax config files in the *.install files. -William King Dmitry Bely wrote: > On Thu, Sep 24, 2009 at 2:14 AM, William King wrote: > >> Sure, post it here and I'll add it in the next build in a few hours. >> > > See attached file. > > Unfortunately mod_skypiax author did not placed config files > (skypiax.conf.xml, skypiax.X.conf) into conf/autoload_configs, so they > are not included into freeswitch-config and should be added manually. > > BTW, why swig is a dependency for the source package? I recall Brian's > post where he insists that swig is never needed to build Freeswitch. > > >> -William King >> >> Dmitry Bely wrote: >> >>> On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle wrote: >>> >>> >>>> On Wed, Sep 23, Dmitry Bely wrote: >>>> >>>> >>>>> Can you enable mod_skypiax in your debian package? >>>>> >>>>> >>>> We will be enabling as much as we can cleanly build on debian/ubuntu. There will be a lot more to come. We will be breaking the mods and end points in to different packages so that you can install what you like. If you have something you would like to see in the package let us know. Also patches are welcome. >>>> >>>> >>> Well, mod_skypiax just requires trivial one-line addition to >>> debian/rules and debian/freeswitch.install. It builds OK. If the patch >>> is required I can post it here. >>> > > - Dmitry Bely > > ------------------------------------------------------------------------ > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Thu Sep 24 11:29:19 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 24 Sep 2009 13:29:19 -0500 Subject: [Freeswitch-users] Bind to more than one ethernet interface In-Reply-To: <87f2f3b90909241042m27da4a80m4dd67b7e0a4c3471@mail.gmail.com> References: <23f91030909232144v2389170fx5556bb7b83b5e361@mail.gmail.com> <903da5680909232222n244b924bp4d875a6988e71733@mail.gmail.com> <87f2f3b90909241042m27da4a80m4dd67b7e0a4c3471@mail.gmail.com> Message-ID: <191c3a030909241129x329001a6x5df062950b85f8ef@mail.gmail.com> if you bind the same profile to more than one ip, all your traffic would come in one ip and out another and cause tremendous confusion. To see a working example of this problem see asterisk https://issues.asterisk.org/view.php?id=2358 (note bkw and mikej contribute to this bug) Here is me reporting a similar issue in IAX (my only negative karma ever on that site) https://issues.asterisk.org/view.php?id=7315 moral of the story is, it's unwise to bind multiple ip to a server interface that uses UDP signalling and the SIP spec requires a UA to have one specific URL On Thu, Sep 24, 2009 at 12:42 PM, Michael Collins wrote: > > > On Wed, Sep 23, 2009 at 10:22 PM, Gabriel Gunderson wrote: > >> On Wed, Sep 23, 2009 at 10:44 PM, Seven Du wrote: >> > It not possible to use 0.0.0.0 for on profile. however, you can create >> more >> > sip profiles for each of your interfaces. Search freeswitch-users >> archievs >> > then you will find similar topics. >> >> It sure would be nice to be able to provide a list of IPs in one >> profile. Seems like I run into that need often. Instead, I end up >> making two profiles that are mostly the same. >> >> Is there a reason why this is the way it is? >> > > A SIP profile is a user agent. By design it services exactly one IP/Port > combo. If you are finding that you need to have multiple IP's or ports in a > single SIP profile then that suggests to me that you might be trying to use > the wrong tool for the job. What are the scenarios where you feel you need a > single profile to handle multiple IP addresses? My guess is that bkw and the > gang will have suggestions for an elegant solution that will fit your needs. > > -MC > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/8ff1cefa/attachment.html From brian at freeswitch.org Thu Sep 24 11:32:55 2009 From: brian at freeswitch.org (Brian West) Date: Thu, 24 Sep 2009 13:32:55 -0500 Subject: [Freeswitch-users] Polycom MWI Forgetfulness In-Reply-To: <200909241243.52381.chris@cloudtel.com> References: <68755a9c0909231044ta70d850vab3ce7971992433d@mail.gmail.com> <68755a9c0909231638t5e39cfbkc0d1b48e0891d8d8@mail.gmail.com> <200909241243.52381.chris@cloudtel.com> Message-ID: <8173A603-8012-4499-9AFB-C9C39EFA3557@freeswitch.org> It beeps every few min cuz you register and we send you a notify again. /b On Sep 24, 2009, at 11:43 AM, Chris Burns wrote: > This happens with our polycoms as well ... NAT on phone and PBX. > Still haven't > had time to look into it so I disabled the sound for new message > waiting ... > for now it doesn't keep beeping every few minutes. From harry at vangberg.name Thu Sep 24 11:35:29 2009 From: harry at vangberg.name (Harry Vangberg) Date: Thu, 24 Sep 2009 20:35:29 +0200 Subject: [Freeswitch-users] Transfer hangs. Message-ID: <74d41a3d0909241135y75d4ddaeq24392f7239b46d54@mail.gmail.com> Hello My setup is this (I've simplified everything, because a lot of my logic isn't necesarry for showcasing this): A calls in, transfer is bound as meta app, B is bridged. When the meta app is processed, the call is transfered to a new extension, which rebridges A. But! After triggering the meta app, it hangs 20 seconds, until transfering to the new extension, unless the B-leg hangs up manually. It should be noted that I've set dtmf-type=sip-info, as I would like to bypass media?if there's a better solution to get DTMF events while bypassing media, please say so, as I know the SIP INFO solution is kinda havoced. This is my dialplan: ... A full trace of a session with A calling in, B answering, B triggering meta app, waiting for transfer, and finally bridge to C is attached. This is using freeswitch-trunk at 14962 -------------- next part -------------- A non-text attachment was scrubbed... Name: trace.log Type: application/octet-stream Size: 51721 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/36ce5084/attachment-0001.obj From anthony.minessale at gmail.com Thu Sep 24 11:50:39 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 24 Sep 2009 13:50:39 -0500 Subject: [Freeswitch-users] Transfer hangs. In-Reply-To: <74d41a3d0909241135y75d4ddaeq24392f7239b46d54@mail.gmail.com> References: <74d41a3d0909241135y75d4ddaeq24392f7239b46d54@mail.gmail.com> Message-ID: <191c3a030909241150w1a04759dg5d14751dfd49bcf8@mail.gmail.com> because it's waiting for the other party to answer if you want to hear ringback or music while you are waiting see: http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones specifically transfer_ringback On Thu, Sep 24, 2009 at 1:35 PM, Harry Vangberg wrote: > Hello > > My setup is this (I've simplified everything, because a lot of my > logic isn't necesarry for showcasing this): A calls in, transfer is > bound as meta app, B is bridged. When the meta app is processed, the > call is transfered to a new extension, which rebridges A. But! After > triggering the meta app, it hangs 20 seconds, until transfering to the > new extension, unless the B-leg hangs up manually. > > It should be noted that I've set dtmf-type=sip-info, as I would like > to bypass media?if there's a better solution to get DTMF events while > bypassing media, please say so, as I know the SIP INFO solution is > kinda havoced. > > This is my dialplan: > > > > > > > > > data="sofia/gateway/gw1.fonet.dk/46934488" /> > > > > > > data="sofia/gateway/gw1.fonet.dk/31354228" /> > > > ... > > > > A full trace of a session with A calling in, B answering, B triggering > meta app, waiting for transfer, and finally bridge to C is attached. > > This is using freeswitch-trunk at 14962 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/ea39bdb9/attachment.html From anthony.minessale at gmail.com Thu Sep 24 12:01:32 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 24 Sep 2009 14:01:32 -0500 Subject: [Freeswitch-users] ASR using mod_unimrcp, vlingo In-Reply-To: References: <217503.51780.qm@web111306.mail.gq1.yahoo.com> Message-ID: <191c3a030909241201t6e1c6af6l52168c261f3c246d@mail.gmail.com> if you find the time, can you add that to the wiki too? On Thu, Sep 24, 2009 at 12:07 PM, Jim Page wrote: > Hi Arsen > > > > Thanks for your message ? it inspired us to do what we should have done in > the first place, and look at the code. The problem we were having was > related to grammar files not being available locally. Now we have discovered > the ?builtin:? keyword we are up and running J > > > > Many thanks > > Jim > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Arsen > Chaloyan > *Sent:* 24 September 2009 18:24 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Re: [Freeswitch-users] ASR using mod_unimrcp, vlingo > > > > Hi Jim, > > >From conceptual viewpoint, mod_unimrcp is just an alternate implementation > of an abstract ASR/TTS interface FreeSWITCH provides. > Therefore you can use it exactly the same way as other ASR/TTS modules. > See scripts/javascript/ps_pizza.js in FS tree for a working example. > > The only thing you should know and change there is module name > < var asr = new SpeechDetect(session, "pocketsphinx"); > > var asr = new SpeechDetect(session, "unimrcp"); > > Typically you can specify any grammar your MRCP server supports. > ------------------------------ > > *From:* Jim Page > *To:* "freeswitch-users at lists.freeswitch.org" < > freeswitch-users at lists.freeswitch.org> > *Sent:* Thursday, September 24, 2009 1:25:20 PM > *Subject:* [Freeswitch-users] ASR using mod_unimrcp, vlingo > > Hi All > > Has anyone had any experience doing ASR with mod_unimrcp in javascript? In > particular, how do you deal with grammars? A simple piece of demo code would > be massively appreciated - the documentation on mod_unimrcp ASR javascript > bindings is TBD, which I assume means 'to be documented' ... unless it means > 'to be developed' ... > > Also is anyone aware of a vlingo integration for freeswitch? > > All the best > Jim > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/4e44d37a/attachment.html From harry at vangberg.name Thu Sep 24 12:04:03 2009 From: harry at vangberg.name (Harry Vangberg) Date: Thu, 24 Sep 2009 21:04:03 +0200 Subject: [Freeswitch-users] Transfer hangs. In-Reply-To: <191c3a030909241150w1a04759dg5d14751dfd49bcf8@mail.gmail.com> References: <74d41a3d0909241135y75d4ddaeq24392f7239b46d54@mail.gmail.com> <191c3a030909241150w1a04759dg5d14751dfd49bcf8@mail.gmail.com> Message-ID: <74d41a3d0909241204m5a2caaf7q370c8322b253bb24@mail.gmail.com> Not exactly, as I said, if the original B-leg doesn't hang up, it will wait 20 second before transfering to the new extension (check the timestamps!) - but if the original B leg hangs up, it gets transfered to the extension immediately. Look at this: 2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_async.c:2042 sofia/external/hemmeligt at 129.142.224.250 Processing meta digit '1' [transfer::ff-transfer XML public] 2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:813 Send signal sofia/external/hemmeligt at 129.142.224.250 [BREAK] 2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_bridge.c:228 sofia/external/hemmeligt at 129.142.224.250 receive message [UNBRIDGE] 2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:630 Send signal sofia/external/hemmeligt at 129.142.224.250 [BREAK] 2009-09-24 18:29:48.278342 [DEBUG] switch_core_session.c:813 Send signal sofia/external/46934488 [BREAK] 2009-09-24 18:29:48.298341 [DEBUG] switch_ivr_bridge.c:228 sofia/external/46934488 receive message [UNBRIDGE] 2009-09-24 18:29:48.298341 [DEBUG] switch_core_session.c:630 Send signal sofia/external/46934488 [BREAK] 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr.c:540 sofia/external/46934488 Command Execute playback(local_stream://moh) EXECUTE sofia/external/46934488 playback(local_stream://moh) 2009-09-24 18:29:48.438320 [WARNING] mod_local_stream.c:318 Unknown source moh, trying 'default' 2009-09-24 18:29:48.438320 [ERR] mod_local_stream.c:327 Unknown source default 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:231 sofia/external/46934488 receive message [BRIDGE] 2009-09-24 18:29:48.438320 [DEBUG] switch_core_session.c:630 Send signal sofia/external/46934488 [BREAK] 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:233 Send signal sofia/external/hemmeligt at 129.142.224.250 [BREAK] 2009-09-24 18:30:09.111448 [DEBUG] switch_ivr.c:540 sofia/external/hemmeligt at 129.142.224.250 Command Execute transfer(ff-transfer XML public) EXECUTE sofia/external/hemmeligt at 129.142.224.250 transfer(ff-transfer XML public) >From 18:29:48 to 19:30:09 nothing happens - it's first then it's transferred to the new extension, and first after that that the new B-leg will even get called. 2009/9/24 Anthony Minessale : > because it's waiting for the other party to answer > > if you want to hear ringback or music while you are waiting > see: > http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones > > specifically transfer_ringback > > > On Thu, Sep 24, 2009 at 1:35 PM, Harry Vangberg wrote: >> >> Hello >> >> My setup is this (I've simplified everything, because a lot of my >> logic isn't necesarry for showcasing this): A calls in, transfer is >> bound as meta app, B is bridged. When the meta app is processed, the >> call is transfered to a new extension, which rebridges A. But! After >> triggering the meta app, it hangs 20 seconds, until transfering to the >> new extension, unless the B-leg hangs up manually. >> >> It should be noted that I've set dtmf-type=sip-info, as I would like >> to bypass media?if there's a better solution to get DTMF events while >> bypassing media, please say so, as I know the SIP INFO solution is >> kinda havoced. >> >> This is my dialplan: >> >> >> ? >> >> ? ? >> ? ? ? >> ? ? ? ? >> ? ? ? ? >> ? ? ? ?> data="sofia/gateway/gw1.fonet.dk/46934488" /> >> ? ? ? >> ? ? >> >> ? ? >> ? ? ? >> ? ? ? ?> data="sofia/gateway/gw1.fonet.dk/31354228" /> >> ? ? ? >> ? ? >> ? ?... >> ? >> >> >> A full trace of a session with A calling in, B answering, B triggering >> meta app, waiting for transfer, and finally bridge to C is attached. >> >> This is using freeswitch-trunk at 14962 >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > iax:guest at conference.freeswitch.org/888 > googletalk:conf+888 at conference.freeswitch.org > pstn:213-799-1400 > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From tculjaga at gmail.com Thu Sep 24 12:06:50 2009 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 24 Sep 2009 21:06:50 +0200 Subject: [Freeswitch-users] Event-Name=CHANNEL_DATA - duplicate variables In-Reply-To: <87f2f3b90909240903p67b33aew98f2956a53e4c461@mail.gmail.com> References: <65d96fc80909230715m43ff6774k3fb495277743425f@mail.gmail.com> <87f2f3b90909240903p67b33aew98f2956a53e4c461@mail.gmail.com> Message-ID: <65d96fc80909241206t4fa80e17ga1b85c88cc5bdcae@mail.gmail.com> of course... On Thu, Sep 24, 2009 at 6:03 PM, msc wrote: > Can you move this question over to the -dev list? Pretty much anything > dealing with source code and custom modules will be better served being > discussed by the developers. Thanks! > -MC > > On Wed, Sep 23, 2009 at 7:15 AM, Tihomir Culjaga wrote: > >> >> Hello, >> >> the setup is like this: >> >> >> CALLING_USER(context:Public) => FS => ENDPOINT >> >> >> >> >> The CALLING_USER places a call towards FS. The call is answered and a >> welcome prompt is being played. After that, FS bridges the call towards an >> ENDPOINT. >> >> I've built a custom module where among other stuff i subscribe to >> events... >> >> >> switch_core_add_state_handler(&albatross_state_handler); >> >> switch_state_handler_table_t albatross_state_handler = >> { >> /* on_init */ process_init, >> /* on_routing */ NULL, /* Need to add a check here for anything in >> their account before routing */ >> /* on_execute */ NULL, /* Turn on heartbeat for this session and do >> an initial account check */ >> /* on_hangup */ process_hangup, /* On hangup - most important place to >> go bill */ >> /* on_exch_media */ NULL, >> /* on_soft_exec */ NULL, >> /* on_consume_med */ NULL, >> /* on_hibernate */ NULL, >> /* on_reset */ NULL >> }; >> >> >> >> So, according to the callflow specified above, on hangup i get two events >> ... (1st B-LEG followed by A-LEG). >> >> >> well .. it sems i have double variables on the A-LEG ?!?!?! .. whats wrong >> hewre ? >> >> >> variable_duration=0 >> variable_billsec=0 >> variable_progresssec=0 >> variable_answersec=0 >> variable_progress_mediasec=0 >> variable_flow_billsec=0 >> variable_mduration=0 >> variable_billmsec=0 >> variable_progressmsec=0 >> variable_answermsec=0 >> variable_progress_mediamsec=0 >> variable_flow_billmsec=0 >> variable_uduration=0 >> variable_billusec=0 >> variable_progressusec=0 >> variable_answerusec=0 >> variable_progress_mediausec=0 >> variable_flow_billusec=0 >> >> >> >> >> Dump:** >> >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Event-Name=CHANNEL_DATA *<= B_LEG (Outbound call)* >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Core-UUID=ca97d306-a840-11de-8ae7-5585e27e6446 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> FreeSWITCH-Hostname=node1 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> FreeSWITCH-IPv4=192.168.102.81 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 FreeSWITCH-IPv6=::1 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Event-Date-Local=2009-09-23 14:59:22 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Date-GMT=Wed, >> 23 Sep 2009 12:59:22 GMT >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Event-Date-Timestamp=1253710762323094 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Event-Calling-File=mod_albatross.c >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Event-Calling-Function=process_hangup >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Event-Calling-Line-Number=2366 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-State=CS_HANGUP >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-State-Number=10 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-Name=sofia/external/0914392122 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Unique-ID=de716bd0-a840-11de-8ae7-5585e27e6446 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Call-Direction=outbound >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Presence-Call-Direction=outbound >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Answer-State=answered >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-Read-Codec-Name=PCMA >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-Read-Codec-Rate=8000 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-Write-Codec-Name=PCMA >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-Write-Codec-Rate=8000 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Username=016659280 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Dialplan=XML >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Caller-ID-Name=016659280 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Caller-ID-Number=016659280 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Network-Addr=172.16.1.20 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Destination-Number=0914392122 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Unique-ID=de716bd0-a840-11de-8ae7-5585e27e6446 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Source=mod_sofia >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Context=public >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Channel-Name=sofia/external/0914392122 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Profile-Index=1 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Profile-Created-Time=1253710743103107 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Channel-Created-Time=1253710743103107 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Channel-Answered-Time=1253710748807099 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Channel-Progress-Time=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Channel-Progress-Media-Time=1253710745451100 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Channel-Hangup-Time=1253710762323094 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Channel-Transfer-Time=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Screen-Bit=true >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Privacy-Hide-Name=false >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Privacy-Hide-Number=false >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Username=016659280 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Dialplan=XML >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Caller-ID-Name=016659280 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Caller-ID-Number=016659280 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Network-Addr=172.16.1.20 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Destination-Number=012468601_Kviz >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Unique-ID=d2242912-a840-11de-8ae7-5585e27e6446 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Source=mod_sofia >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Context=public >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Channel-Name=sofia/external/016659280 at sip-priv.amis.hr >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Profile-Created-Time=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Channel-Created-Time=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Channel-Answered-Time=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Channel-Progress-Time=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Channel-Progress-Media-Time=1253710745451100 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Channel-Hangup-Time=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Channel-Transfer-Time=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Screen-Bit=true >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Privacy-Hide-Name=false >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Other-Leg-Privacy-Hide-Number=false >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_gateway_name=gw1 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_channel_name=sofia/external/0914392122 >> 2009-09-23 14:59:22.323094 [NOTICE] mod_albatross.c:2320 *** HANGUP >> RECEIVED, BYE... >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_destination_url=sip:0914392122 at 172.16.1.20 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_is_outbound=true >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_h_P-Access-Network-Info=ADSL;dsl_location="NOA=4;APRI=1;ADD=3851";network-provided >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_cid_type=pid >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_max_forwards=30 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_originator_codec=PCMA at 8000h@20i >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_originator=d2242912-a840-11de-8ae7-5585e27e6446 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_execute_on_answer=updateQuizServiceStatus_ch in 012468601, in >> connected >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_switch_m_sdp=v=0 >> o=cp10 125371067255 125371067255 IN IP4 172.16.2.42 >> s=SIP Call >> c=IN IP4 172.16.2.42 >> t=0 0 >> m=audio 34474 RTP/AVP 8 0 18 125 101 >> b=AS:64 >> a=rtpmap:8 PCMA/8000/1 >> a=rtpmap:0 PCMU/8000/1 >> a=rtpmap:18 G729/8000/1 >> a=fmtp:18 annexb=no >> a=rtpmap:125 CLEARMODE/8000/1 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_service_number=012468601 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_dialed_number=0914392122 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_originate_early_media=true >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sofia_profile_name=external >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_call_id=b5cfd0c9-22e3-122d-2c98-001e684a25c5 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_switch_r_sdp=v=0 >> o=cp10 125371069278 125371069279 IN IP4 172.16.2.42 >> s=SIP Call >> c=IN IP4 172.16.2.42 >> t=0 0 >> m=audio 34972 RTP/AVP 8 0 >> b=AS:64 >> a=rtpmap:8 PCMA/8000/1 >> a=rtpmap:0 PCMU/8000/1 >> a=ptime:20 >> >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_remote_media_ip=172.16.2.42 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_remote_media_port=34972 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_read_codec=PCMA >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_read_rate=8000 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Event-Name=CHANNEL_DATA *<= A-LEG (Inbound)* >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Core-UUID=ca97d306-a840-11de-8ae7-5585e27e6446 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> FreeSWITCH-Hostname=node1 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> FreeSWITCH-IPv4=192.168.102.81 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 FreeSWITCH-IPv6=::1 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Event-Date-Local=2009-09-23 14:59:22 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Event-Date-GMT=Wed, >> 23 Sep 2009 12:59:22 GMT >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Event-Date-Timestamp=1253710762323094 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Event-Calling-File=mod_albatross.c >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Event-Calling-Function=process_hangup >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_write_codec=PCMA >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_write_rate=8000 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_local_media_ip=172.16.3.2 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_local_media_port=42684 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_bridge_channel=sofia/external/016659280 at sip-priv.amis.hr >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_bridge_uuid=d2242912-a840-11de-8ae7-5585e27e6446 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_signal_bond=d2242912-a840-11de-8ae7-5585e27e6446 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_reply_host=172.16.1.20 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_reply_port=5060 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_endpoint_disposition=ANSWER >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_current_application_data=in 012468601, in connected >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_current_application=updateQuizServiceStatus_ch >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_term_status=200 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_term_cause=16 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_user_agent=Cirpack/v4.42d (gw_sip) >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_hangup_disposition=recv_bye >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_hangup_cause=NORMAL_CLEARING >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_hangup_cause_q850=16 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_digits_dialed=none >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_start_stamp=2009-09-23 14:59:03 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_profile_start_stamp=2009-09-23 14:59:03 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_answer_stamp=2009-09-23 14:59:08 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progress_media_stamp=2009-09-23 14:59:05 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_end_stamp=2009-09-23 14:59:22 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_start_epoch=1253710743 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_start_uepoch=1253710743103107 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_profile_start_epoch=1253710743 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_profile_start_uepoch=1253710743103107 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_answer_epoch=1253710748 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_answer_uepoch=1253710748807099 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progress_epoch=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progress_uepoch=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progress_media_epoch=1253710745 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progress_media_uepoch=1253710745451100 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_end_epoch=1253710762 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_end_uepoch=1253710762323094 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_last_app=updateQuizServiceStatus_ch >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_last_arg=in 012468601, in connected >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Event-Calling-Line-Number=2366 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_caller_id="016659280" <016659280> >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_duration=19 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-State=CS_HANGUP >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_billsec=14 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-State-Number=10 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progresssec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-Name=sofia/external/016659280 at sip-priv.amis.hr >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_answersec=5 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progress_mediasec=2 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Unique-ID=d2242912-a840-11de-8ae7-5585e27e6446 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_flow_billsec=19 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Call-Direction=inbound >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_mduration=19220 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Presence-Call-Direction=inbound >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_billmsec=13516 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Answer-State=answered >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progressmsec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-Read-Codec-Name=PCMA >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_answermsec=5704 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-Read-Codec-Rate=8000 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progress_mediamsec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-Write-Codec-Name=PCMA >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_flow_billmsec=19220 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Channel-Write-Codec-Rate=8000 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_uduration=19219987 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_billusec=13515995 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Username=016659280 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progressusec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 Caller-Dialplan=XML >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_answerusec=5703992 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progress_mediausec=2347993 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Caller-ID-Name=016659280 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_flow_billusec=19219987 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Caller-ID-Number=016659280 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_raw_bytes=144996 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_media_bytes=144996 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Network-Addr=172.16.1.20 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_packet_count=843 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Destination-Number=012468601_Kviz >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_media_packet_count=843 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Unique-ID=d2242912-a840-11de-8ae7-5585e27e6446 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_skip_packet_count=1 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Source=mod_sofia >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_jb_packet_count=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Context=public >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_dtmf_packet_count=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Channel-Name=sofia/external/016659280 at sip-priv.amis.hr >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_cng_packet_count=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Screen-Bit=true >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_flush_packet_count=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Privacy-Hide-Name=false >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_raw_bytes=143964 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> Caller-Privacy-Hide-Number=false >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_media_bytes=143964 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_received_ip=172.16.1.20 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_packet_count=837 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_received_port=5060 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_media_packet_count=837 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_via_protocol=udp >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_skip_packet_count=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_from_params=user=phone >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_dtmf_packet_count=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_from_user=016659280 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_cng_packet_count=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_from_uri=016659280 at sip-priv.amis.hr >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_from_host=sip-priv.amis.hr >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_from_user_stripped=016659280 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_from_tag=11746-FX-007c2917-23e2091e3 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sofia_profile_name=external >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_P-Asserted-Identity=016659280 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_cid_type=pid >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2407 siId: (null) >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_req_params=user=phone >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2408 ani: 016659280 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_req_user=012468601 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2409 dnis: 0914392122 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_req_port=5060 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2410 dialed_number: >> 0914392122 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2412 start_stamp: >> 2009-09-23 14:59:03 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_req_uri=012468601 at 172.16.3.2:5060 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2413 answer_stamp: >> 2009-09-23 14:59:08 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2414 progress_stamp: >> (null) >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_req_host=172.16.3.2 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_to_params=user=phone >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2415 >> progress_media_stamp: 2009-09-23 14:59:05 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_to_user=012468601 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2416 end_stamp: >> 2009-09-23 14:59:22 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_to_uri=012468601 at 172.16.3.2 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2420 >> privacy_hide_number: false >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_to_host=172.16.3.2 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_contact_user=nobody >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_contact_port=5060 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_contact_uri=nobody at 172.16.1.20:5060 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_contact_host=172.16.1.20 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_channel_name=sofia/external/016659280 at sip-priv.amis.hr >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_call_id=11746-QJ-007c2916-59db25232 at sip-priv.amis.hr >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_user_agent=Cirpack/v4.42d (gw_sip) >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_via_host=172.16.1.20 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_via_port=5060 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_max_forwards=31 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_h_P-Access-Network-Info=ADSL;dsl_location="NOA=4;APRI=1;ADD=3851";network-provided >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_switch_r_sdp=v=0 >> o=cp10 125371067255 125371067255 IN IP4 172.16.2.42 >> s=SIP Call >> c=IN IP4 172.16.2.42 >> t=0 0 >> m=audio 34474 RTP/AVP 8 0 18 125 101 >> b=AS:64 >> a=rtpmap:8 PCMA/8000/1 >> a=rtpmap:0 PCMU/8000/1 >> a=rtpmap:18 G729/8000/1 >> a=fmtp:18 annexb=no >> a=rtpmap:125 CLEARMODE/8000/1 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-15 >> a=ptime:20 >> >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_remote_media_ip=172.16.2.42 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_remote_media_port=34474 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_read_codec=PCMA >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_read_rate=8000 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_write_codec=PCMA >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_write_rate=8000 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_service_type_id=1 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_local_media_ip=172.16.3.2 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_local_media_port=27686 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_price_prompt=3.66kn_novo_upozorenje.wav >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_dialed_number=012468601 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_bNum=012468601 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_service_status1=win >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_number_2_connect=0914392122 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_next_number_2_connect=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_next_number_2_display=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_service_instance=130 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_service_id=2 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_quiz_status=win >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_not_working_prompt=2/emisija trenutno nije u tijeku-za telefone.wav >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_service_win_prompt=2/bit cete spojeni u emisiju1.wav >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_service_loose_prompt=2/zovi ponovo.wav >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_outside_call=true >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_endpoint_disposition=ANSWER >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_playback_ms=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_playback_samples=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_bypass_media=false >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_hangup_after_bridge=true >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_nolocal:execute_on_answer=updateQuizServiceStatus_ch in 012468601, >> in connected >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_export_vars=nolocal:execute_on_answer >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_current_application_data=[service_number=012468601,dialed_number=0914392122]sofia/gateway/gw1/0914392122 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_current_application=bridge >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_switch_m_sdp=v=0 >> o=cp10 125371069278 125371069279 IN IP4 172.16.2.42 >> s=SIP Call >> c=IN IP4 172.16.2.42 >> t=0 0 >> m=audio 34972 RTP/AVP 8 0 >> b=AS:64 >> a=rtpmap:8 PCMA/8000/1 >> a=rtpmap:0 PCMU/8000/1 >> a=ptime:20 >> >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_originate_disposition=SUCCESS >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_bridge_channel=sofia/external/0914392122 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_bridge_uuid=de716bd0-a840-11de-8ae7-5585e27e6446 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_signal_bond=de716bd0-a840-11de-8ae7-5585e27e6446 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_hangup_phrase=OK >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_bridge_hangup_cause=NORMAL_CLEARING >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_hangup_cause=NORMAL_CLEARING >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_hangup_cause_q850=16 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_digits_dialed=none >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_last_app=bridge >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_last_arg=[service_number=012468601,dialed_number=0914392122]sofia/gateway/gw1/0914392122 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_caller_id="016659280" <016659280> >> *2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_duration=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_billsec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progresssec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_answersec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progress_mediasec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_flow_billsec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_mduration=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_billmsec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progressmsec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_answermsec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progress_mediamsec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_flow_billmsec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_uduration=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 variable_billusec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progressusec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_answerusec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_progress_mediausec=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_flow_billusec=0* >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_sip_hangup_disposition=send_bye >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_raw_bytes=341764 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_media_bytes=341764 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_packet_count=1987 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_media_packet_count=1987 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_skip_packet_count=10 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_jb_packet_count=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_dtmf_packet_count=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_cng_packet_count=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_in_flush_packet_count=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_raw_bytes=304440 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_media_bytes=304440 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_packet_count=1770 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_media_packet_count=1770 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_skip_packet_count=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_dtmf_packet_count=0 >> 2009-09-23 14:59:22.323094 [INFO] mod_albatross.c:2375 >> variable_rtp_audio_out_cng_packet_count=0 >> >> >> >> >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/94250b9e/attachment-0001.html From grae at digilord.net Thu Sep 24 13:12:41 2009 From: grae at digilord.net (digilord) Date: Thu, 24 Sep 2009 13:12:41 -0700 Subject: [Freeswitch-users] Polycom MWI Forgetfulness In-Reply-To: <8173A603-8012-4499-9AFB-C9C39EFA3557@freeswitch.org> References: <68755a9c0909231044ta70d850vab3ce7971992433d@mail.gmail.com> <68755a9c0909231638t5e39cfbkc0d1b48e0891d8d8@mail.gmail.com> <200909241243.52381.chris@cloudtel.com> <8173A603-8012-4499-9AFB-C9C39EFA3557@freeswitch.org> Message-ID: <1253823161.6024.5.camel@digilords-desktop.digilord.net> Brian, I am using the latest firmware. Would a solution be to lower the registration time so that notifies happen more often? I know that would increase the amount of traffic on the network but it would keep the light lit when a user has a message. DigiLord On Thu, 2009-09-24 at 13:32 -0500, Brian West wrote: > It beeps every few min cuz you register and we send you a notify again. > > /b > > On Sep 24, 2009, at 11:43 AM, Chris Burns wrote: > > > This happens with our polycoms as well ... NAT on phone and PBX. > > Still haven't > > had time to look into it so I disabled the sound for new message > > waiting ... > > for now it doesn't keep beeping every few minutes. > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/478dd66e/attachment.bin From siniypin at gmail.com Thu Sep 24 13:20:47 2009 From: siniypin at gmail.com (RobertT) Date: Fri, 25 Sep 2009 00:20:47 +0400 Subject: [Freeswitch-users] instant messaging Message-ID: <2160023e0909241320kfa246eeu437205a7bdfe9df@mail.gmail.com> Hi guys! I'm considering to use SIMPLE protocol for IM in my application, but get a following error trying to send a message from one registered user to another: [ERR] sofia_presence.c:93 Chat proto [sip] from [1001 at xx.xxx.xx.xx] to [1000 at xx.xxx.xx.xx] 1111111111 Invalid Profile xx.xxx.xx.xx Should presence be enabled in order SIMPLE to work? What additional steps do I have to complete in order to make presence work in FS besides setting "manage_presence" param in SIP profile to true? Best regards, Robert. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090925/5c1ac394/attachment.html From dmitry.bely at gmail.com Thu Sep 24 13:36:18 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Fri, 25 Sep 2009 00:36:18 +0400 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <4ABA47FB.2050100@gmail.com> References: <4ABA47FB.2050100@gmail.com> Message-ID: <90823c940909241336h636d2c30wb0de776b7423681d@mail.gmail.com> On Wed, Sep 23, 2009 at 8:08 PM, William King wrote: Another problem: all music packages in the repository (except 48Khz) are empty. - Dmitry Bely From quentusrex at gmail.com Thu Sep 24 13:55:10 2009 From: quentusrex at gmail.com (William King) Date: Thu, 24 Sep 2009 13:55:10 -0700 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <90823c940909241336h636d2c30wb0de776b7423681d@mail.gmail.com> References: <4ABA47FB.2050100@gmail.com> <90823c940909241336h636d2c30wb0de776b7423681d@mail.gmail.com> Message-ID: <4ABBDCAE.6030007@gmail.com> You're saying the binary sounds files are empty? And only the music ones? -William King Dmitry Bely wrote: > On Wed, Sep 23, 2009 at 8:08 PM, William King wrote: > > Another problem: all music packages in the repository (except 48Khz) are empty. > > - Dmitry Bely > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From frank at carmickle.com Thu Sep 24 14:00:17 2009 From: frank at carmickle.com (Frank Carmickle) Date: Thu, 24 Sep 2009 17:00:17 -0400 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <90823c940909241336h636d2c30wb0de776b7423681d@mail.gmail.com> References: <4ABA47FB.2050100@gmail.com> <90823c940909241336h636d2c30wb0de776b7423681d@mail.gmail.com> Message-ID: <20090924210017.GC17256@base.carmickle.com> Hello On Fri, Sep 25, Dmitry Bely wrote: > On Wed, Sep 23, 2009 at 8:08 PM, William King wrote: > > Another problem: all music packages in the repository (except 48Khz) are empty. If your speaking of the source package then they should be. If the binary package freeswitch-sounds-music-8000 is empty then we have problems. It was working the other day. Please let me know. --FC From andrew at hijacked.us Thu Sep 24 14:09:55 2009 From: andrew at hijacked.us (Andrew Thompson) Date: Thu, 24 Sep 2009 17:09:55 -0400 Subject: [Freeswitch-users] skill-based ACD In-Reply-To: <87f2f3b90909240840u74a322d4m33255c46244b17b5@mail.gmail.com> References: <20ad6b920909130801k7ae3fa64u83e2bd5a4e9e7c15@mail.gmail.com> <11372C8B9E603F4FACDE6AB18256DEC695A9EC@srvmtel.office.mtel.nl> <87f2f3b90909171357y40fe0db4kef1d59fb86e9790c@mail.gmail.com> <11372C8B9E603F4FACDE6AB18256DEC601D8BC94@srvmtel.office.mtel.nl> <87f2f3b90909240840u74a322d4m33255c46244b17b5@mail.gmail.com> Message-ID: <20090924210954.GN7677@hijacked.us> On Thu, Sep 24, 2009 at 08:40:16AM -0700, msc wrote: > On Thu, Sep 24, 2009 at 1:09 AM, Remko Kloosterman wrote: > > > Hello Michael, > > > > > > > > Do you still want to follow up on this? I?m having difficulty gathering the > > old stuff in an understandable form. Also, it looks like the open source ACD > > Spice Telephony by Andrew Thompson can do just what you might need. > > > > I had totally forgotten about Andrew's stuff! Unless people want to build > their own 100% community/free/DIY version of a skill-based ACD then I say > let's all play with SpiceCSM and help improve it. I'd certainly appreciate the feedback (and the kick in the ass to improve some of the rough spots and documentation). Andrew From dmitry.bely at gmail.com Thu Sep 24 14:32:11 2009 From: dmitry.bely at gmail.com (Dmitry Bely) Date: Fri, 25 Sep 2009 01:32:11 +0400 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <20090924210017.GC17256@base.carmickle.com> References: <4ABA47FB.2050100@gmail.com> <90823c940909241336h636d2c30wb0de776b7423681d@mail.gmail.com> <20090924210017.GC17256@base.carmickle.com> Message-ID: <90823c940909241432u612ba659uee9cbe8f86299155@mail.gmail.com> On Fri, Sep 25, 2009 at 1:00 AM, Frank Carmickle wrote: > Hello > > On Fri, Sep 25, Dmitry Bely wrote: >> On Wed, Sep 23, 2009 at 8:08 PM, William King wrote: >> >> Another problem: all music packages in the repository (except 48Khz) are empty. > > If your speaking of the source package then they should be. ?If the binary package freeswitch-sounds-music-8000 is empty then we have problems. ?It was working the other day. ?Please let me know. Of course, I meant the binary packages: http://ppa.launchpad.net/pbxbuntu-drivers/ppa/ubuntu/pool/main/f/freeswitch-sounds-music/ Note their sizes. - Dmitry Bely From grae at digilord.net Thu Sep 24 14:40:37 2009 From: grae at digilord.net (digilord) Date: Thu, 24 Sep 2009 14:40:37 -0700 Subject: [Freeswitch-users] Polycom MWI Forgetfulness In-Reply-To: <8173A603-8012-4499-9AFB-C9C39EFA3557@freeswitch.org> References: <68755a9c0909231044ta70d850vab3ce7971992433d@mail.gmail.com> <68755a9c0909231638t5e39cfbkc0d1b48e0891d8d8@mail.gmail.com> <200909241243.52381.chris@cloudtel.com> <8173A603-8012-4499-9AFB-C9C39EFA3557@freeswitch.org> Message-ID: <1253828437.6024.22.camel@digilords-desktop.digilord.net> Here is the requested SIP trace that Anthony wanted. http://pastebin:freeswitch at pastebin.freeswitch.org/10479 This is for ext 102 at 192.168.0.2. On line 168 the phone thinks there are no messages. On line 206 the phone thinks there are no messages. On line 309 the phone thinks there are messages. On line 599 the phone thinks there are no messages. On line 637 the phone thinks there are no messages. -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/75c82e6e/attachment.bin From quentusrex at gmail.com Thu Sep 24 15:58:53 2009 From: quentusrex at gmail.com (William King) Date: Thu, 24 Sep 2009 15:58:53 -0700 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <90823c940909241432u612ba659uee9cbe8f86299155@mail.gmail.com> References: <4ABA47FB.2050100@gmail.com> <90823c940909241336h636d2c30wb0de776b7423681d@mail.gmail.com> <20090924210017.GC17256@base.carmickle.com> <90823c940909241432u612ba659uee9cbe8f86299155@mail.gmail.com> Message-ID: <4ABBF9AD.8050401@gmail.com> Alright. I was able to get the freeswitch project officially on launchpad. So here are the new links: Nightlies: https://launchpad.net/~freeswitch-drivers/+archive/freeswitch-nightly-drivers Official releases plus major bug fixes: https://launchpad.net/~freeswitch-drivers/+archive/ppa -William King Dmitry Bely wrote: > On Fri, Sep 25, 2009 at 1:00 AM, Frank Carmickle wrote: > >> Hello >> >> On Fri, Sep 25, Dmitry Bely wrote: >> >>> On Wed, Sep 23, 2009 at 8:08 PM, William King wrote: >>> >>> Another problem: all music packages in the repository (except 48Khz) are empty. >>> >> If your speaking of the source package then they should be. If the binary package freeswitch-sounds-music-8000 is empty then we have problems. It was working the other day. Please let me know. >> > > Of course, I meant the binary packages: > > http://ppa.launchpad.net/pbxbuntu-drivers/ppa/ubuntu/pool/main/f/freeswitch-sounds-music/ > > Note their sizes. > > - Dmitry Bely > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Thu Sep 24 16:08:02 2009 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 24 Sep 2009 18:08:02 -0500 Subject: [Freeswitch-users] Transfer hangs. In-Reply-To: <74d41a3d0909241204m5a2caaf7q370c8322b253bb24@mail.gmail.com> References: <74d41a3d0909241135y75d4ddaeq24392f7239b46d54@mail.gmail.com> <191c3a030909241150w1a04759dg5d14751dfd49bcf8@mail.gmail.com> <74d41a3d0909241204m5a2caaf7q370c8322b253bb24@mail.gmail.com> Message-ID: <191c3a030909241608r8c9c346jd1c3a61e50615360@mail.gmail.com> in that case, it's probably a delay in the media stream where the app is queued when you press the key try updating to trunk and add the new i flag to the flags param i.e. 1 b ai transfer::ff-transfer XML public On Thu, Sep 24, 2009 at 2:04 PM, Harry Vangberg wrote: > Not exactly, as I said, if the original B-leg doesn't hang up, it will > wait 20 second before transfering to the new extension (check the > timestamps!) - but if the original B leg hangs up, it gets transfered > to the extension immediately. > > Look at this: > > 2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_async.c:2042 > sofia/external/hemmeligt at 129.142.224.250 Processing meta digit '1' > [transfer::ff-transfer XML public] > 2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:813 Send > signal sofia/external/hemmeligt at 129.142.224.250 [BREAK] > 2009-09-24 18:29:48.138326 [DEBUG] switch_ivr_bridge.c:228 > sofia/external/hemmeligt at 129.142.224.250 receive message [UNBRIDGE] > 2009-09-24 18:29:48.138326 [DEBUG] switch_core_session.c:630 Send > signal sofia/external/hemmeligt at 129.142.224.250 [BREAK] > 2009-09-24 18:29:48.278342 [DEBUG] switch_core_session.c:813 Send > signal sofia/external/46934488 [BREAK] > 2009-09-24 18:29:48.298341 [DEBUG] switch_ivr_bridge.c:228 > sofia/external/46934488 receive message [UNBRIDGE] > 2009-09-24 18:29:48.298341 [DEBUG] switch_core_session.c:630 Send > signal sofia/external/46934488 [BREAK] > 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr.c:540 > sofia/external/46934488 Command Execute playback(local_stream://moh) > EXECUTE sofia/external/46934488 playback(local_stream://moh) > 2009-09-24 18:29:48.438320 [WARNING] mod_local_stream.c:318 Unknown > source moh, trying 'default' > 2009-09-24 18:29:48.438320 [ERR] mod_local_stream.c:327 Unknown source > default > 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:231 > sofia/external/46934488 receive message [BRIDGE] > 2009-09-24 18:29:48.438320 [DEBUG] switch_core_session.c:630 Send > signal sofia/external/46934488 [BREAK] > 2009-09-24 18:29:48.438320 [DEBUG] switch_ivr_bridge.c:233 Send signal > sofia/external/hemmeligt at 129.142.224.250 [BREAK] > 2009-09-24 18:30:09.111448 [DEBUG] switch_ivr.c:540 > sofia/external/hemmeligt at 129.142.224.250 Command Execute > transfer(ff-transfer XML public) > EXECUTE sofia/external/hemmeligt at 129.142.224.250 transfer(ff-transfer > XML public) > > >From 18:29:48 to 19:30:09 nothing happens - it's first then it's > transferred to the new extension, and first after that that the new > B-leg will even get called. > > 2009/9/24 Anthony Minessale : > > because it's waiting for the other party to answer > > > > if you want to hear ringback or music while you are waiting > > see: > > http://wiki.freeswitch.org/wiki/Custom_Ring_Back_Tones > > > > specifically transfer_ringback > > > > > > On Thu, Sep 24, 2009 at 1:35 PM, Harry Vangberg > wrote: > >> > >> Hello > >> > >> My setup is this (I've simplified everything, because a lot of my > >> logic isn't necesarry for showcasing this): A calls in, transfer is > >> bound as meta app, B is bridged. When the meta app is processed, the > >> call is transfered to a new extension, which rebridges A. But! After > >> triggering the meta app, it hangs 20 seconds, until transfering to the > >> new extension, unless the B-leg hangs up manually. > >> > >> It should be noted that I've set dtmf-type=sip-info, as I would like > >> to bypass media?if there's a better solution to get DTMF events while > >> bypassing media, please say so, as I know the SIP INFO solution is > >> kinda havoced. > >> > >> This is my dialplan: > >> > >> > >> > >> > >> > >> > >> > >> > >> >> data="sofia/gateway/gw1.fonet.dk/46934488" /> > >> > >> > >> > >> > >> > >> >> data="sofia/gateway/gw1.fonet.dk/31354228" /> > >> > >> > >> ... > >> > >> > >> > >> A full trace of a session with A calling in, B answering, B triggering > >> meta app, waiting for transfer, and finally bridge to C is attached. > >> > >> This is using freeswitch-trunk at 14962 > >> > >> _______________________________________________ > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at 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_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > iax:guest at conference.freeswitch.org/888 > > googletalk:conf+888 at conference.freeswitch.org > > pstn:213-799-1400 > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at 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_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org iax:guest at conference.freeswitch.org/888 googletalk:conf+888 at conference.freeswitch.org pstn:213-799-1400 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/00565673/attachment-0001.html From stevecrozz at gmail.com Thu Sep 24 16:08:11 2009 From: stevecrozz at gmail.com (Stephen Crosby) Date: Thu, 24 Sep 2009 16:08:11 -0700 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <4ABBF9AD.8050401@gmail.com> References: <4ABA47FB.2050100@gmail.com> <90823c940909241336h636d2c30wb0de776b7423681d@mail.gmail.com> <20090924210017.GC17256@base.carmickle.com> <90823c940909241432u612ba659uee9cbe8f86299155@mail.gmail.com> <4ABBF9AD.8050401@gmail.com> Message-ID: <11990ade0909241608n65991dccq37ee88b4bbb29b8@mail.gmail.com> Thanks William, This is very helpful. --Stephen On Thu, Sep 24, 2009 at 3:58 PM, William King wrote: > Alright. I was able to get the freeswitch project officially on > launchpad. So here are the new links: > > Nightlies: > > https://launchpad.net/~freeswitch-drivers/+archive/freeswitch-nightly-drivers > > Official releases plus major bug fixes: > https://launchpad.net/~freeswitch-drivers/+archive/ppa > > -William King > > Dmitry Bely wrote: > > On Fri, Sep 25, 2009 at 1:00 AM, Frank Carmickle > wrote: > > > >> Hello > >> > >> On Fri, Sep 25, Dmitry Bely wrote: > >> > >>> On Wed, Sep 23, 2009 at 8:08 PM, William King > wrote: > >>> > >>> Another problem: all music packages in the repository (except 48Khz) > are empty. > >>> > >> If your speaking of the source package then they should be. If the > binary package freeswitch-sounds-music-8000 is empty then we have problems. > It was working the other day. Please let me know. > >> > > > > Of course, I meant the binary packages: > > > > > http://ppa.launchpad.net/pbxbuntu-drivers/ppa/ubuntu/pool/main/f/freeswitch-sounds-music/ > > > > Note their sizes. > > > > - Dmitry Bely > > > > _______________________________________________ > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090924/7780d9a2/attachment.html From mike at jerris.com Thu Sep 24 21:26:22 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 25 Sep 2009 00:26:22 -0400 Subject: [Freeswitch-users] Subscribing to events in managed C# / .NET In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702DCEC6B3C@mse17be1.mse17.exchange.ms> References: <367751820909030726s75e9a32bhaefb68759afbaca2@mail.gmail.com> <6E8D2069C08AA84A83D336E996AE4C6702DB2B598E@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B59B3@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DB2B5B8E@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DCEC6B31@mse17be1.mse17.exchange.ms> <6E8D2069C08AA84A83D336E996AE4C6702DCEC6B3C@mse17be1.mse17.exchange.ms> Message-ID: There are a few other things I can think would be nice additions to mod_managed. Maybe an event handler that does not require a thread to be sitting and waiting for events trying in a loop would be nice, instead something that is triggered each time there is a certain event class triggered. Also, there has been some interest in doing full endpoint modules in mod_managed. exposing all the state handlers in .net like ways and having that all work would be quite interesting, but probably requires someone specific actually ready to write a module like that to be worthwhile. Mike On Sep 24, 2009, at 4:01 AM, Michael Giagnocavo wrote: > Great ? hopefully we?ll meet on IRC or the conference sometime on > Friday. Email me when you?re on. > > A few questions I have: > > Clarity ? I agree with you there, and thanks! > > Testability ? is this even remotely practical? Looking at our FS > code plugins, there?s simply no way any amount of test environment > code would get us to anything testable. We make tons of direct P/ > Invoke calls, and the whole model for what variables are set when, > the state machine progression, etc. does not seem like something > that we can hope to possibly model right. And it?s subject to many > external influences (all the modules you have loaded in FS). Logging > is a pretty simple case, sure, we can make it not call FS for > testing. But in a real app, it just seems that there are way too > many dependencies, no? Maybe others who have apps written can chime > in? > > Modularity ? I agree there are two parts. But, I think they are > pretty tightly coupled. The FS interface into unmanaged code is done > via unmanaged code and is really clear: App, Api, ApiBackground. The > other ways I can think of are FS-specific, such as XML binding > interface and so on. But those are things we should just add to the > mod_managed core and be done with. I?m thinking maybe we are talking > about different things? Can you provide some user stories that we > want to cover with a pluggable loader/executor/etc.? Thanks for > putting up with me! > > -Michael > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Josh Rivers > Sent: Thursday, September 24, 2009 12:32 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Subscribing to events in managed > C# / .NET > > > > On Wed, Sep 23, 2009 at 7:31 PM, Michael Giagnocavo > wrote: > Right off the bat: there can be tons of cleanup and refactoring, no > doubt about that. Much of the current code is to satisfy my needs in > production, which it does very well. > > The current base doesn't have anything wrong with it for sure, in > fact, I learned a good bit about PInvoke. AppDomains, and In-Process > Remoting in the last week. > > My refactoring had the following goals (in no particular order) > - Testability - I'd really like to see a decent unit test suite on > the more module so that we can change it with confidence. Also, it's > been drilled into me that a testable design is a good design. > - Clarity - Where possible, I extracted blocks of code that served > a particular purpose so that purpose could be self-documenting in > the method calls rather than mixed in. > - Modularity - I wanted to make it easy to remove or add > alternative behavior to the managed.dll. > > I?m a bit hesitant to go too far from the FreeSWITCH core as far as > architecture goes. For instance, I?m not quite sure why?d we have > our own managed logging subsystem that allows them to plug in other > things that aren?t part of FS. Either they should use the FS logging > system, or use their own such as log4net. Or perhaps I don?t see why > we?d want this behavior. > > I completely agree, with the following caveats: > 1) I'd like to see things testable. It's very hard to do isolation > testing with classes making direct calls out to a static Log class > that in turn pinvokes out to unmanaged code. > 2) I'd like to allow folk to make changes to the default behavior > (optimally) without recompiling managed.dll. > > One thing at issue here is that there are two principal purposes for > managed.dll. The first is to provide an interface into unmanaged > code. The second is a module/plugin extensibility framework. The > first purpose should absolutely provide the thinnest layer possible. > The second purpose is very likely to need a lot of change and > adaptation as people come up with development models that they would > like to follow in using freeswitch. The extensibility framework > should be mostly managed code, coded to interfaces for mock-ability > and testabiliy. It should also be able to just push it out of the > way and hook your own extensibilty framework in instead. > Going away from the core as far as adding .NET specific features > (like look at the static ManagedSession.Originate that takes hangup > delegates, or the ?nice? wrapper for Log (Write and WeiteLine, with > an enum instead of a string) are keeping close to the core, just > adding a tiny bit of API cleanup. FreeSWITCH exposes a lot of > strings, and while maybe that?s important for some languages, .NET > users are going to expect stronger typing. But I don?t think these > types of things get people away from FreeSWITCH much. > > No disagreement here. I would like to see these things made > available by interface rather than concrete implementation. It's > currently not possible to test a plugin without loading it into FS. > That precludes automated testing, and leaves a pretty big round-trip > to test a tweak. I'm a sloppy coder too, so I'm always introducing > interesting regressions, and that's why I like doing my testing > without having to bring up a full process :) > Things like making a published SOAP interface for FS seem not really > related to mod_managed. They can easily be done as 3rd party > plugins, or convince the core FS team that exposing via SOAP via > mod_managed is the way to go. Also keep in mind that the majority of > users are on Linux, so that rules out WCF and some other fun stuff > that only works on the CLR ? I?d say it all has to work on Mono. > > This kind of stuff is definitely beyond the scope of mod_managed. > Although there is a slippery slope since we're building in an > extensibility model. I don't think a WCF host, or a winforms host, > or any of that should be baked in. Rather, I think we should provide > the hooks for adding such a thing. If somebody wants to build ESL > via WCF, why should they need to leave managed code? If the module > system is general enough, then such a thing should just be a module. > (BTW, I think WCF-Mono is getting there http://www.mono-project.com/WCF_Development > ) > Absolutely, everything in mod_managed and managed.dll should run on > mono and the CLR. However, there shouldn't be any reason that a Win- > only developer can't build a complete FS application framework that > plugs in and only runs on Windows. > As for all the rest of it, can we talk interactively, perhaps with > other users interested in mod_managed? Reading over your email, I > think I?m not understanding many of the use cases that are being > fixed. > > I'd be very glad to get a discussion going. I definitely haven't > covered all of the issues here. > > -Josh > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090925/8948c710/attachment-0001.html From mike at jerris.com Thu Sep 24 21:27:43 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 25 Sep 2009 00:27:43 -0400 Subject: [Freeswitch-users] User Creation with DB in Freeswitch In-Reply-To: <59daa2cd0909240402r1fe089e3r52414a2ac41c255a@mail.gmail.com> References: <1b2118200909201313jfceee74o33e6ad13f4a0086e@mail.gmail.com> <566E110B-72CC-428E-AEE8-94AB600587B0@enterux.com> <347D3032-F778-46F4-AECE-94DECD27E849@freeswitch.org> <1b2118200909231137w79ea33afq6050133fb50b32a0@mail.gmail.com> <903da5680909232236v21cd9f47xa2aaa83749a16fab@mail.gmail.com> <59daa2cd0909240402r1fe089e3r52414a2ac41c255a@mail.gmail.com> Message-ID: There are a number of examples out there such as: http://svn.freeswitch.org/svn/freeswitch/trunk/contrib/intralanman/PHP/fs_curl/ Mike On Sep 24, 2009, at 7:02 AM, Costa Zikalala wrote: > Hi Gabe > > Thanks for you response to this question. > Do you perhaps have a link to an example (or just further detail) to > what you've descibed below. > I guess one would also use a similar setup to generate dialplans > from web forms. > > Thanks again, > Costa > > > > 2009/9/24 Gabriel Gunderson > On Wed, Sep 23, 2009 at 12:37 PM, Anil Kumar S. R. > wrote: > > I didn't get much help for my problem with XML CURL. What I meant > to say is, > > suppose I want to have some 10000 users on freeswitch. Do we have > to create > > some many xml files in the directory or is there some way in which > the users > > can be put in the db ? > > That's the whole point. You serve up the XML from *your* web server > using whatever technologies that *you* would like on the back-end. > You'll want to make that XML reply dynamic. > > Use php, perl, python, c#, java or whatever other language *you* want. > Pull the data from MySQL, flat-files, PostgreSQL, MSSQL, LDAP or > whatever *you* want. Just serve up the right bit of configuration to > FreeSWITCH and you're done. > > Good luck. Spend more time in the docs. Others have posted the > links. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090925/452d70b9/attachment.html From mike at jerris.com Thu Sep 24 21:40:01 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 25 Sep 2009 00:40:01 -0400 Subject: [Freeswitch-users] checking subscribed/subscribers to events In-Reply-To: References: Message-ID: <3AAB8360-DC8A-426C-B6C4-3F7A3E968BD6@jerris.com> If you need to be able to do granular permissions like that you would either need to extend mod_event_socket or write a proxy that handled that. Mike On Sep 24, 2009, at 5:23 AM, Alberto Escudero wrote: > Hi, > > Is there any simple way to know: > > who is subscribed to certain events via ESL? > check which events i have subscribed during a ESL session? > control which events can one user subscribe? > disable the subscription of certain events and not all at the same > time? From mike at jerris.com Thu Sep 24 21:41:58 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 25 Sep 2009 00:41:58 -0400 Subject: [Freeswitch-users] Freeswitch on embedded device: interesting or not? In-Reply-To: References: Message-ID: I know of at least one person who has had good luck with small applications on arm, in fact there are good working instructions for how to cross for arm on the wiki that are known to work. Mike On Sep 24, 2009, at 5:34 AM, Cavalera Claudio Luigi wrote: > Hello guys, > lately I've been trying to compile Freeswitch for a MIPS architecture. > With the help of the community I've understood that my target > architecture was wrong because of limitations in the SDK toolchain's. > I'm not writing now to get help but to start (I hope) a discussion. > I would like to understand your points of view about the general > idea of > porting FS on embedded devices. > I'm not hardware expert at all; someone says that porting FS to any > appliance which is not x86 based is a loss of time, because ARM and > MIPS > processors just lack computational power, this could be true, but > maybe > it depends on what you expect FS to do on such an embedded > architecture. > We are now all used to the amazing performance of FS on multicores > 64bit > cpus but still the one line description of FS is: > "FreeSWITCH is an open source telephony platform designed to > facilitate > the creation of voice and chat driven products scaling from a soft- > phone > up to a soft-switch". > Therefore a scaled down FS could be done, do you think is interesting? > I'm not speaking here from a technical point of view, I know others > have > already compiled FS for ARM and MIPS and their experience is on the > wiki. > Would you consider a scaled down FS only for x86 architectures (e.g. > the > pfSense package or Atom)? > > Regards, > Claudio > From mike at jerris.com Thu Sep 24 21:37:59 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 25 Sep 2009 00:37:59 -0400 Subject: [Freeswitch-users] Status of ubuntu/debian packages. In-Reply-To: <4ABBB10E.1080900@gmail.com> References: <4ABA47FB.2050100@gmail.com> <90823c940909231251k5f85e3dambb7506bdc62e62ac@mail.gmail.com> <20090923202524.GR30343@base.carmickle.com> <90823c940909231507ie10dc30x17442386f401c375@mail.gmail.com> <4ABA9DC9.6090808@gmail.com> <90823c940909241029w1bc6c302obf602f94a5654362@mail.gmail.com> <4ABBB10E.1080900@gmail.com> Message-ID: I can confirm you should not need the swig dependency at all for anything. Mike On Sep 24, 2009, at 1:49 PM, William King wrote: > Hmm... That is interesting... swig is needed I believe only for the > mod_perl or the esl modules. I'll find out more information and put it > on the correct package. > > I will also update the mod_skypiax config files in the *.install > files. > > -William King > > Dmitry Bely wrote: >> On Thu, Sep 24, 2009 at 2:14 AM, William King >> wrote: >> >>> Sure, post it here and I'll add it in the next build in a few hours. >>> >> >> See attached file. >> >> Unfortunately mod_skypiax author did not placed config files >> (skypiax.conf.xml, skypiax.X.conf) into conf/autoload_configs, so >> they >> are not included into freeswitch-config and should be added manually. >> >> BTW, why swig is a dependency for the source package? I recall >> Brian's >> post where he insists that swig is never needed to build Freeswitch. >> >> >>> -William King >>> >>> Dmitry Bely wrote: >>> >>>> On Thu, Sep 24, 2009 at 12:25 AM, Frank Carmickle >>> > wrote: >>>> >>>> >>>>> On Wed, Sep 23, Dmitry Bely wrote: >>>>> >>>>> >>>>>> Can you enable mod_skypiax in your debian package? >>>>>> >>>>>> >>>>> We will be enabling as much as we can cleanly build on debian/ >>>>> ubuntu. There will be a lot more to come. We will be breaking >>>>> the mods and end points in to different packages so that you can >>>>> install what you like. If you have something you would like to >>>>> see in the package let us know. Also patches are welcome. >>>>> >>>>> >>>> Well, mod_skypiax just requires trivial one-line addition to >>>> debian/rules and debian/freeswitch.install. It builds OK. If the >>>> patch >>>> is required I can post it here. >>>> >> >> - Dmitry Bely >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From mike at jerris.com Thu Sep 24 21:44:17 2009 From: mike at jerris.com (Michael Jerris) Date: Fri, 25 Sep 2009 00:44:17 -0400 Subject: [Freeswitch-users] mod_cdr_csv missing timestamps in A-LEG In-Reply-To: <65d96fc80909240310q31cf2cbby1ed82a66d6555ce5@mail.gmail.com> References: <65d96fc80909240310q31cf2cbby1ed82a66d6555ce5@mail.gmail.com> Message-ID: Can you get these same values in xml-cdr? I don't think csv was ever intended to work with different cdrs for a and b leg, it was more intended as a more familiar interface for those coming over from asterisk. Mike On Sep 24, 2009, at 6:10 AM, Tihomir Culjaga wrote: > hello, > > i'm on latest trunk and for some reason i cannot get timestamps > dumped in my cdrs. I use mod_cdr_csv with default settings plus i > enabled to get both a and b legs dumped. > > > cdr_csv.conf.xml: > > > > > > > > > > > > > > > >