September 2012 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] Conference CDRs via http
by Mariusz Czulada 04 Dec '12

04 Dec '12

Hi all, I was thinking about sending CDRs via HTTP in a same or similar way mod_xml_cdr does. I consider implementing this (unless someone else is working on it) but first wanted to discuss with you the best approach. A. All in mod_conference This would require to copy many fragments of code from mod_xml_cdr into mod_conference. Also same configuration parameters used in xml_cdr must be processed and used for sending data. The advantage for this solution is that everything related to this mechanism is included in this module. Drawback: if something in mod_xml_cdr requires fixing or extending, probably the same changes should by applied in related parts of mod_conference. B. mod_conference builds, mod_xml_cdr sends data For this solution I'll give more details. 1. New event type should be added (like SWITCH_EVENT_CDR) 2. When a module (in this case: mod_conference) wants to store CDR via HTTP it must fire an event of that type and: - "Event-Subclass" set (like "conference") - Only common headers are needed, plus... - "Content-Type" and "Content-Length" must be set - CDR data must be build as XML in a module and added as an event content. 3. mod_xml_cdr will listen to this event type. 4. For each event subclass which mod_xml_cdr must must react, configuration file will contain a set of params same as for generic channel CDRs. 5. If subclass matches configuration, mod_xml_cdr reads data from event content and sends them according to configuration. Changes in mod_conference: - one new parameter for each profile (like "cdr-via-event=yes|no") - if "yes" then xml must be build even if "cdr-log-dir" is unset - if "yes", then an event must be fired as described above Changes in mod_xml_cdr: - extra parameters from configuration to be parsed (like '<settings subclass="conference">....</settings>') - bind to SWITCH_EVENT_CDR - if 'Event-Subclass' matches configuration, a content of the event will be sent via HTTP (probably most of 'my_on_report' routine bellow 'try to post it to the web server' comment will be reused) A [small] drawback is that it makes an indirect module dependency, but we already have such situations (like mod_shout needed to record a conference in mp3). Advantages are: - one can create an external tool for handling this type of event (to store it directly in db or send it with other protocols) - this mechanism can be easily reused in other modules if needed; maybe in mod_callcenter, maybe in other components. No further changes in mod_xml_cdr should be needed. What is unknown to me is a maximum size of event content. Conference CDR XMLs can be quite big - will it be a problem to send it this way? I think the second solution is better and more universal but I'd like to hear your opinions about this case. Regards, Mariusz

3 3

[Freeswitch-dev] Linking C++ .so Into Custom Freeswitch Module
by Jon Lederman 30 Sep '12

30 Sep '12

Hi, I am writing a custom module called mod_aw for freeswitch that uses a dynamic library called aw.so containing a c++ class. I edited the makefile for my module to: BASE=../../../.. LOCAL_LIBADD+=./lib/aw.so include $(BASE)/build/modmake.rules I am able to build the module and also modified modules.conf to include the new module. When I start freeswitch, I don't see a message that the module was loaded. When I type: reload mod_aw I receive the following error: 2012-09-28 04:49:52.738493 [CRIT] switch_loadable_module.c:1310 Error Loading module /usr/local/freeswitch/mod/mod_aw.so **/usr/local/freeswitch/mod/mod_aw.so: undefined symbol: _ZN10CAWETalker9ConfigureEPKc** -ERR unloading module [No such module!] +OK Reloading XML -ERR loading module [module load file routine returned an error] I know the library works as I have tried it in another program outside of freeswitch. I am not sure what I am doing wrong. I am wondering whether I need to provide extern "C" linkage for the c++ library. However, I only have the .h file and .so file for the library. Or, do I need to add something to my Makefile. Any help on how to configure this would be greatly appreciated. Thank you in advance. -Jon

4 7

[Freeswitch-dev] Why are Windows users so badly served - and what can I do to help?
by Niels Thomsen 27 Sep '12

27 Sep '12

To me and my software engineering colleagues, one of the great things about FS is that it is available on Windows. But I often feel a bit left out... J For instance, the download site doesn't appear to have the stable builds (e.g. 1.2.3) available for download pre-built. Only the unstable head of git built weekly. Also, the pre-built installer doesn't allow me to select an install directory, and is unaware of modern Windows install hierarchies. This means you either have to run as administrator or copy the installed files out of there after install. Eurgh. I can build it myself of course, and have done so. But the build has literally hundreds of warnings, so can I trust it? My day-time job is as a software engineer in a 'no warnings' environment, where code cannot be checked in if it generates warnings, so that's a bit of a culture shock. I haven't tried building the linux build to see if this is a VS2010 issue. But rather than moan about it, I'd like to help. I'd like to sort out the installer, for instance, so it is brought up to date and can install on the latest OSs without kludges. But I'd also like to see it being used to generate stable Windows pre-builds at the end of it - which would presumably be done by someone looking after the download site? So where do I start? Who do I speak to? - Niels

4 4

[Freeswitch-dev] :how to let the version 2.1.3 as well as the latest (from git) work for the 0911 and 5001 dialplan? there is no problem with version 1.06
by zhi sun 27 Sep '12

27 Sep '12

in the default diaplan, the 0911 conference call that make a out call to a group doesn't work for the latest and version 2.1.3. the out call always failed because of CHAN_NOT_IMPLEMENTED, but it works fine for version 1.06 the same problem happens with the 5001 in dialplan (default.xml) it is very easy to reproduce this issue: i just get the correct version, make, make install, make cd-sounds-install. the log with problem looks like below: =========================================== 2012-09-26 15:37:10.973469 [NOTICE] switch_ivr.c:1748 Transfer sofia/internal/1000(a)192.168.0.100 to XML[1010@default] 2012-09-26 15:37:10.973469 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/0000000000(a)192.168.0.100 [BREAK] 2012-09-26 15:37:10.973469 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/0000000000(a)192.168.0.100 [BREAK] 2012-09-26 15:37:10.973469 [DEBUG] switch_core_session.c:905 Send signal sofia/internal/1010(a)192.168.0.100 [BREAK] 2012-09-26 15:37:10.993469 [DEBUG] switch_channel.c:2950 (sofia/internal/ 1010(a)192.168.0.100) Callstate Change RINGING -> HANGUP 2012-09-26 15:37:10.993469 [NOTICE] switch_ivr_originate.c:3330 Hangup sofia/internal/1010(a)192.168.0.100 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2012-09-26 15:37:10.993469 [DEBUG] switch_channel.c:2973 Send signal sofia/internal/1010(a)192.168.0.100 [KILL] 2012-09-26 15:37:10.993469 [DEBUG] switch_core_session.c:1210 Send signal sofia/internal/1010(a)192.168.0.100 [BREAK] 2012-09-26 15:37:10.993469 [DEBUG] switch_ivr_originate.c:3506 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL] 2012-09-26 15:37:10.993469 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1010(a)192.168.0.100) Running State Change CS_HANGUP 2012-09-26 15:37:10.993469 [ERR] mod_conference.c:6626 Cannot create outgoing channel, cause: ORIGINATOR_CANCEL 2012-09-26 15:37:10.993469 [DEBUG] switch_channel.c:2950 (sofia/internal/ 1000(a)192.168.0.100) Callstate Change ACTIVE -> HANGUP 2012-09-26 15:37:10.993469 [NOTICE] mod_conference.c:6629 Hangup sofia/internal/1000(a)192.168.0.100 [CS_ROUTING] [ORIGINATOR_CANCEL] 2012-09-26 15:37:10.993469 [DEBUG] switch_channel.c:2973 Send signal sofia/internal/1000(a)192.168.0.100 [KILL] 2012-09-26 15:37:10.993469 [DEBUG] switch_core_state_machine.c:638 (sofia/internal/1010(a)192.168.0.100) State HANGUP 2012-09-26 15:37:10.993469 [DEBUG] switch_core_session.c:1210 Send signal sofia/internal/1000(a)192.168.0.100 [BREAK] ============================================ i am new to freeswitch, Is there anything i missed to let the version 2.1.3 as well as the latest (from git) work for the 0911 and 5001 dialplan? both of them make an out call from a conference. thanks -zhisun

3 7

[Freeswitch-dev] FreeSWITCH Conf Call Today
by Michael Collins 26 Sep '12

26 Sep '12

Don't forget about today's call! http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_26 We get to have a deeper technical conversation with Scott and Daniel regarding the repro SIP proxy. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

1 0

[Freeswitch-dev] leg_timeout not used if multi-contact phone on that leg.
by Don Dawson 25 Sep '12

25 Sep '12

We discovered that the leg_timeout will work fine as long as the SIP device has only one contact. If it has multiple contacts then it is ignored. The bridge ends in 60 seconds so maybe is using originate_timeout which defaults to 60(we do not use it in the dialplan). As a workaround for now, we may need to specify the originate_timeout to shorten the bridge timeout as 10 rings to hit voicemail is not desired. This was discovered on a box that was built a while back. I tested on a recent version from git date of 09/06/12 and it has the same problem. I will get the very latest GIT version shortly to test with.

1 0

[Freeswitch-dev] FreeSWITCH Weekly News and Notes
by Michael Collins 24 Sep '12

24 Sep '12

Welcome to the last Monday of September 2012! We've had quite the interesting week. Perhaps the most interesting item the team dealt with was a vulnerability in the Sofia SIP stack that would cause a segmentation fault while processing a specially crafted SIP message. Just to show you how nimble the FreeSWITCH developers are, from the time the vulnerability was reported it took less than a day to fix, test, and roll a new version of FreeSWITCH. We encourage everyone on 1.2 to get updated to version 1.2.3 <http://files.freeswitch.org/freeswitch-1.2.3.tar.bz2> as soon as possible. (The fix is also in the 1.3 development branch as of last Wednesday, September 19.) We tip our hats to Anthony and the rest of the dev team<http://wiki.freeswitch.org/wiki/FreeSwitch_FAQ#Q:_So_where_do_I_find_wishli…>for their hard work on our behalf. Last week's conference call<http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_19>was also very informative. We received an introduction to the repro SIP proxy <http://resiprocate.org/About_Repro> software. We look forward to this coming Wednesday <http://wiki.freeswitch.org/wiki/FS_weekly_2012_09_26>where Scott Godin and Daniel Pocock will continue the discussion and will get deeper into how to set up the proxy and use it with FreeSWITCH. If you haven't already tried to install repro please do so. Daniel has a nice tutorial<http://www.opentelecoms.org/using-repro-with-asterisk-or-freeswitch>over at OpenTelecoms.org - be sure to check it out and bring your questions on Wednesday. Finally, we'd like to draw your attention to this blog post<http://blog.krisk.org/2012/09/freeswitch-stable-branch-sponsored-by.html>by long time FreeSWITCH and open source telephony supporter Kristian Kielhofner. Kristian reports that his company, Star2Star Communications<http://www.star2star.com/>, is sponsoring the FreeSWITCH stable branch by giving direct financial support to the project. This allows for a full-time team member to work on things like the stable branch and packaging as well as community interaction and documentation. We appreciate those who support FreeSWITCH and open source telephony! Have a good week and we'll see you again in October. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

1 0

[Freeswitch-dev] ./configure & libcodec2
by Piotr Sobolewski 24 Sep '12

24 Sep '12

Hello branch v1.2stable from git System is Debian squeez The ./configure ends with what's below. === configuring in libs/libcodec2 (/usr/local/src/1.2.3-2/freeswitch/libs/libcodec2) configure: running /bin/sh ./configure.gnu --disable-option-checking '--prefix=/usr/local/freeswitch' 'CONFIGURE_CFLAGS=-g -O2' 'CONFIGURE_CXXFLAGS=-g -O2' 'CONFIGURE_LDFLAGS=' --cache-file=/dev/null --srcdir=. ./configure: line 2059: syntax error near unexpected token `libcodec2,1.0' ./configure: line 2059: `AM_INIT_AUTOMAKE(libcodec2,1.0)' configure: error: ./configure.gnu failed for libs/libcodec2 Is it known problem? -- Piotr Sobolewski sobolewski(a)gmail.com

1 0

[Freeswitch-dev] switch_channel_add_state_handler not work under ubuntu
by Xijing Dai 22 Sep '12

22 Sep '12

hey i have release 1.2.1 code base, the state handler is called in Windows, but not in Ubuntu 12.04 here is the code: static switch_state_handler_table_t channel_event_handlers = { > /*.on_init */ NULL, > /*.on_routing */ NULL, > /*.on_execute */ NULL, > /*.on_hangup */ channel_on_hangup, > /*.on_exchange_media */ NULL, > /*.on_soft_execute */ NULL, > /*.on_consume_media */ NULL, > /*.on_hibernate */ NULL, > /*.on_reset */ NULL, > /*.on_park */ NULL, > /*.on_reporting */ NULL, > /*.on_destroy */ NULL > }; > ...... > if (switch_channel_add_state_handler(hook_channel, > &channel_event_handlers) < 0) > { > switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(session), > SWITCH_LOG_ERROR, "state handler added failed\n"); > } > ....... > > Anyone knows why?

3 7

[Freeswitch-dev] mod_callcenter => mod_huntgroup.
by Ali Işık 21 Sep '12

21 Sep '12

Hello, devs, I am thinking about developing a mod_huntgroup. I am new to FreeSWITCH and would appreciate any help starting up. At first glance, mod_callcenter seems to have some relevant features. I am trying to decide whether it would make a good starting point. In particular, I would like to know how mod_callcenter is viewed by you guys -- is it widely used, for instance, and actively maintained? Does it have a place in your roadmap? Cheers, -- Ali Isik

5 13