June 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

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[Freeswitch-dev] Conference recording pause/resume problem
by Mariusz Czulada 13 Jul '12

13 Jul '12

Hi, I try to prepare a patch which allows to pause/resume a given recording. I'm in the middle of the road: pause is working, by resume - not :(. I'd like to ask for your help in finding a problem. I'm not sure if I handle member and file flags in the thread sub correctly. BTW. I lost some time playing with member-fnode until found fh in conference_record_thread_run is a proper handler. What is the conference_file_node_t member->fnode used for, if not for handling a recording file? Regards, Mariusz

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[Freeswitch-dev] Issue while creating a session for an incoming call in managed application
by Papineni, Suneel 02 Jul '12

02 Jul '12

Hi, I am having a third party VoIP client which exposes API in .NET to communicate with my application. I want to use this client as an endpoint in Freeswitch (like how it is implemented in "mod_skypopen" module), so that calls coming to this client can be handled at Freeswitch and (if required) bridge calls to clients connected to Freeswitch. To implement this, I have written a managed module in which "ILoadNotificationPlugin" is implemented where config (client's interface config) is loaded. When there is an incoming call (ringing state) to the client, I get handle in managed module and trying to establish a session at freeswitch. While doing this I am getting exception as "Attempted to read or write protected memory. This is often an indication that other memory is corrupt." Here is the code that I am using... Could someone guide me if I am doing this correctly. Note: I am following "mod_skypopen" as an example... public static int new_inbound_channel(private_object pobject) { SWIGTYPE_p_switch_core_session session = null; SWIGTYPE_p_switch_channel channel = null; switch_endpoint_interface intf = new switch_endpoint_interface(); intf.interface_name = "NET_interface"; intf.io_routines = io_routines; intf.state_handler = state_handlers; freeswitch.switch_core_set_globals(); Api fsApi = new Api(); String uuid = fsApi.ExecuteString("create_uuid"); try { session = freeswitch.switch_core_session_request_uuid(intf, switch_call_direction_t.SWITCH_CALL_DIRECTION_INBOUND, 0, null, uuid); //I am getting the above issue here.... if (session != null) { freeswitch.switch_core_session_add_stream(session, null); channel = freeswitch.switch_core_session_get_channel(session); if (channel == null) { freeswitch.switch_core_session_destroy_state(session); return 0; } freeswitch.switch_channel_set_variable_name_printf(channel, "waste", "false"); pobject.caller_profile = freeswitch.switch_caller_profile_new(freeswitch.switch_core_session_get_pool(session), "NETClient", "dialplan", "callid_name", "callid_number", null, null, null, null, "mod_netclient", "context", "destination"); if (pobject.caller_profile != null) { freeswitch.switch_channel_set_name(channel, "name"); freeswitch.switch_channel_set_caller_profile(channel, pobject.caller_profile); } freeswitch.switch_channel_set_state_flag(channel, switch_channel_flag_t.CF_GEN_RINGBACK); switch_status_t session_status = freeswitch.switch_core_session_thread_launch(session); if (session_status != switch_status_t.SWITCH_STATUS_SUCCESS) { freeswitch.switch_core_session_destroy_state(session); return 0; } } } catch (Exception exp) { string expstring = exp.Message; } return 0; } Thanks & Regards Suneel

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[Freeswitch-dev] Support for TLS and SRTP with SPA 3102
by Prasd D 30 Jun '12

30 Jun '12

SPA 3102 has TLS support but SRTP with different type of key exchange I think (called Mikey) instead of SDP. It would be very useful to have support for this as an added module or profile if needed. Thanks, Prasd

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[Freeswitch-dev] Connecting FS to MS UM
by Michael Collins 29 Jun '12

29 Jun '12

Hey, if anyone knows about how this can be done with FreeSWITCH, please contact me off list: http://help.outlook.com/en-us/140/gg702670%28d=loband%29.aspx I know someone looking to provide a solution for his customer. I'd like to get this solution documented on the wiki... Thanks, MC

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[Freeswitch-dev] questions about making a new endpoint
by Seven Du 29 Jun '12

29 Jun '12

Hi, I'm working on a mod_rtsp endpoint the idea is to stream sip video to a rtsp client such as VLC and also a sip client could "call" a rtsp camera later. I copied some codes from mod_sofia and it is too long to understand for me so I'd like to get some inputs either highlight me how to do or point out where to read from mod_sofia. 1) VLC "calls" in as a-leg and bridge to a sip client with video(or play_fsv), I made it sort of working now, but - How to disable the read_frame and read_video frame io_routine ? I set them to NULL and it fails on read, so I made a fake read function there, kind of ugly. - I hard coded to H264 and I can see video on VLC, I'd like to do like sip later negotiation then when b-leg sip client offers SDP and offer to a-leg when DESCRIBE - looks like switch_io complains when no audio codec, is it possible to do video only calls? or is it possible to set audio and video rtp to a different path ? 2) Call a rtsp camera would be easy later 3) Maybe a client can call in and record? I don't have such a client. I would like to share the code later if others interested on this. Thanks, Seven. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com)

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[Freeswitch-dev] FreeSWITCH Conf Call Today
by Michael Collins 27 Jun '12

27 Jun '12

Hello gang, This week's conference call will be starting in less than an hour. Here's the agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_27 Talk to you soon! -Michael

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[Freeswitch-dev] stream a file multicast
by Xijing Dai 27 Jun '12

27 Jun '12

hey all, Just wonder is there any progress on on stream a file multicast? I saw some topics on it back to 2010. Currently, I need the similar functionalities. Cheers Xijing Dai

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[Freeswitch-dev] I can not see sendonly/recvonly from SDP
by Xijing Dai 26 Jun '12

26 Jun '12

hey I am using mod_esf to do the multicasting. In bypass_media mode, it will send recvonly SDP to other side. I use wireshark to catch the sip/sdp packets, but I can not see freeswitch sent recvonly inside SDP. After that, I try to set NDLB-sendrecv-in-session to be true (specified here: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#NDLB-sendrecv-in-… ). It still does not send recvonly to the other side. Did I miss anything? I also want to allow softphone to be multicasting listener, is there any extension module to do it?? Cheers Xijing Dai

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[Freeswitch-dev] FreeSWITCH Weekly News and Notes
by Michael Collins 25 Jun '12

25 Jun '12

Welcome to the last week of June! Things quieted down a bit last week, but we still had some interesting items for your consideration. Darren Schreiber from the 2600hz<http://www.2600hz.com?cc12-0625>project joined us last week on our conference call <http://wiki.freeswitch.org/wiki/Weekly_Conference_Call?cc12-0625> to present "SIP 101" - a gentle introduction to the SIP protocol for those doing VoIP. It was a nice discussion and we plan to do a follow up, tentatively scheduled for July 18. The audio is available in the usual place<http://torrents.freeswitch.org/conf_call_2012-06-20.torrent?cc12-0625>and the slides are available here <http://wiki.freeswitch.org/files/SIP_101_Presentation.ppt?cc12-0625>. For this week's<http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_27?cc12-0625>conference call we are going to have an open discussion about topics of interest to our community members. Our ClueCon schedule <http://www.cluecon.com/schedule?cc12-0625> is now full! Please visit the schedule to see a list of speakers and talk titles. We have a few presentations whose titles are yet to be announced, so check back often to stay updated. We are pleased to have Brad Pitt joining us this year! (No, not that one <http://en.wikipedia.org/wiki/Brad_pitt> - we have this one <http://www.cluecon.com/speaker/108?cc12-0625>.) Bradley J Pitt is a long-time veteran of the technology and telecom industries and recently joined Barracuda Networks <http://www.cudatel.com?cc12-0625>. His discussion will touch upon both technology and business aspects of the IP PBX world. As a reminder, you still have about one week before the "bits shift" and you go from 16 down to 8 tickets for your ClueCon registration<http://www.cluecon.com/register?cc12-0625>. Register right away so that you have the best chance possible to win one of the 10 Android tablets that will be given away this year! See you in Chicago! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0625

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