From victor.velo at videotron.ca Fri Jun 1 00:58:20 2012 From: victor.velo at videotron.ca (victor.velo at videotron.ca) Date: Thu, 31 May 2012 16:58:20 -0400 Subject: [Freeswitch-dev] how to prevent DTMF tones/info from being recorded In-Reply-To: <76c0c0af8d7aa.4fc0a204@videotron.ca> References: <76c0c0af8d7aa.4fc0a204@videotron.ca> Message-ID: <77c0a07da187c.4fc7a32c@videotron.ca> Hello Michael, The source is the hand set, or precisely the soft-phone. I looked at the "preferences" and there was this inband DTMF activated. Dis-activating this option solved the issue actually. Thank you very much! Victor ----- Original Message ----- From: Date: Saturday, May 26, 2012 9:27 am Subject: how to prevent DTMF tones/info from being recorded To: freeswitch-dev at lists.freeswitch.org > Hello, > > I'm using ESL (outbound) to control a Freeswitch server. I need to be able to record only the "sound" from the microphone, excluding any DTMF related info and sound... > > The actual command used for recording is "record". > > Thanks a lot for your help, > > Victor > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120531/2a80c170/attachment.html From msc at freeswitch.org Fri Jun 1 23:18:20 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 1 Jun 2012 12:18:20 -0700 Subject: [Freeswitch-dev] Sending arbitrary NOTIFY messages Message-ID: Has anyone gotten the magic formula down for doing this? I'm trying to send a NOTIFY to a non-registered endpoint but I can't quite get the event headers correct. I found this old commit that Tony did about 3 years ago that suggests it's possible: http://fisheye.freeswitch.org/changelog/freeswitch.git/?showid=1fa1e961e4587475a51dcbadd31765b5a06d3115 If you know what the necessary headers are, or better yet, if you have a working example please let me know. And yes, I will gladly pay the wiki tax on this one. ;) Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120601/48db0107/attachment.html From marketing at cluecon.com Mon Jun 4 20:55:22 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 4 Jun 2012 09:55:22 -0700 Subject: [Freeswitch-dev] FreeSWITCH Weekly News And Notes Message-ID: Happy June to all! We had a busy month of May but last Wednesday (May 30) we took a break from the formal presentations on the FreeSWITCH conference call and instead focused some time and energy on topics of general interest to our community. Feel free to downloadlast week's conference call recording and listen to Travis Cross and myself discuss the pros and cons of licensing for content other than source code, such as the wiki documentation and the sounds we've recorded over the years. Another topic of interest is the original FreeSWITCH book, a.k.a. the "bridge book" because of the cover photograph. Can you believe that this summer it will be two years since the book was released? The FreeSWITCH team and a number of interested community members have been discussing what a second edition would look like - what changes would be made, what new content could be added, etc. We have no definitive plans at the moment, so now is the time to talk about what you would like to see. Do you have an idea on how to make the second edition of the FreeSWITCH book even better? If so, please let us know. Perhaps by ClueCon 2013 we'll have a brand new book on the shelves! Speaking of ClueCon , we would like to let everyone know that the plans for this year's event are coming along nicely. Our speaker list is growing and the schedule is filling out. We still have a few openings, so please get your speaking proposals in to us right away. We are looking forward to seeing everyone again this August, so please registernow, that way we can make plans for ancillary events like the welcome reception. Thanks for being such a great community! Talk to you next week. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0604 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120604/176fa5d2/attachment.html From msc at freeswitch.org Mon Jun 4 23:05:38 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 4 Jun 2012 12:05:38 -0700 Subject: [Freeswitch-dev] FreeSWITCH Conference Call June 6 Message-ID: Hello all! Just an update: We've rescheduled Darren's SIP 101 discussion for June 20th. Also, this Wednesday, June 6th yours truly will be doing an introduction to mod_httapi! We will be keeping it simple this week: just getting your server up and running, trying out the sample code, and maybe we'll do some what-if's from the audience. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_06 Keep in mind that mod_httapi allows for *your* FreeSWITCH server to call *my * web server to get dialplan instructions, so be sure to compile and enable mod_httapi on your systems. Talk to you Wednesday! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120604/81cd1af4/attachment-0001.html From gabe at gundy.org Tue Jun 5 06:14:37 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 4 Jun 2012 20:14:37 -0600 Subject: [Freeswitch-dev] FreeSWITCH Weekly News And Notes In-Reply-To: References: Message-ID: On Mon, Jun 4, 2012 at 10:55 AM, Michael Collins wrote: > Another topic of interest is the original FreeSWITCH book, a.k.a. the > "bridge book" because of the cover photograph. Can you believe that this > summer it will be two years since the book was released? Crazy. Where does time go? Gabe From msc at freeswitch.org Wed Jun 6 21:03:57 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 6 Jun 2012 10:03:57 -0700 Subject: [Freeswitch-dev] FreeSWITCH Conf Call Is On! Message-ID: Come join us! http://wiki.freeswitch.org/wiki/FS_weekly_2012_06_06 We're going to talk about mod_httapi and a few other fun things. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120606/ceda50fc/attachment.html From dujinfang at gmail.com Fri Jun 8 17:36:19 2012 From: dujinfang at gmail.com (Seven Du) Date: Fri, 8 Jun 2012 21:36:19 +0800 Subject: [Freeswitch-dev] video formats, playback and recording features Message-ID: <1B681CAD08CD470B90C0BE3A7388A848@gmail.com> Hi there, As we have mod_fsv, mod_mp4v, mod_vlc, mod_dingaling, mod_rtmp, mod_rtsp, some already support video and some are possible. In addition to p2p videos, I'd like to also get video playback, recording, eavesdroping working. So I did some research and here's some quetions: Basically there's two ways to support video 1) formats. formats support a large range of audios, but not videos. So it is possible to extend to add something like switch_file_read_video/switch_file_write_video etc. And also extend playback, record eavesdrop, fifo, esf_page_group ... to support video, and also need to extend video media_bug ... Video files like mp4 also including audio tracks 2) as the current solutions in modules on the top, provide different tools like play_fsv, play_mp4, etc to add video support The 2) perhaps natural as it goes, and 1) perhaps more elegant but may add complexity and possible resource/performance problems to pure audio usage? Advice ? Is there any interest to support video transcodings ? btw, any working in progress of webrtc in FS ? I saw this http://code.google.com/p/sipml5/ sounds cool. 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) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120608/e6f59b99/attachment.html From no-reply at dropboxmail.com Mon Jun 11 18:44:06 2012 From: no-reply at dropboxmail.com (Dropbox) Date: Mon, 11 Jun 2012 14:44:06 +0000 Subject: [Freeswitch-dev] Andre Mendes invited you to Dropbox Message-ID: Andre Mendes wants you to try Dropbox! Dropbox lets you bring all your photos, docs and videos with you anywhere and share them easily. Get started here: http://www.dropbox.com/link/20.N7kjcPs2RS/NjIxOTQ4NDg5Nzc?src=referrals_bulk9&eh=655a568 - The Dropbox Team ____________________________________________________ To stop receiving invites from Dropbox, please go to http://www.dropbox.com/bl/074747b8b714/freeswitch-dev%40lists.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120611/c9c8dfa3/attachment.html From marketing at cluecon.com Mon Jun 11 21:12:47 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 11 Jun 2012 10:12:47 -0700 Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes Message-ID: Happy Monday to you all! Last Wednesday we had a nice discussion on the weekly conference callabout mod_httapi . It turns out that there is a fair amount of interest in the great module but that there are some questions about how to handle certain scenarios, such as "session tracking." Thanks to Raymond Chandler for being available to answer a lot of those questions. We will be doing a followup discussion that focuses on Raymond's PHP examples for using mod_httapi. These examples do a good job of demonstrating the power and ease of HTTAPI. ClueCon season is upon us and our list of speakers and sponsors continues to grow. Packt Publishingis now on board as a media sponsor and we hope to have some nice items from them to give away at the conference. Keep checking the schedulefor updates as we are confirming new speakers each week. We are also happy to report that the Wyndham has completed its changeover and is officially known as the Hyatt Chicago Magnificent Mile. (As of this writing their web site is undergoing maintenance, so be patient.) We are excited to see what's in store at this revamped hotel and can't wait to see everyone there this August! -- Michael S Collins ClueCon Team http://www.cluecon.com?cc12-0611 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120611/84b62ca5/attachment.html From manieq at wp.eu Tue Jun 12 05:38:53 2012 From: manieq at wp.eu (Mariusz Czulada) Date: Tue, 12 Jun 2012 03:38:53 +0200 Subject: [Freeswitch-dev] Conference CDRs via http Message-ID: <4fd69dad04abd6.16863237@wp.pl> 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 '....') - 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 From anthony.minessale at gmail.com Tue Jun 12 06:31:57 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 11 Jun 2012 21:31:57 -0500 Subject: [Freeswitch-dev] Conference CDRs via http In-Reply-To: <4fd69dad04abd6.16863237@wp.pl> References: <4fd69dad04abd6.16863237@wp.pl> Message-ID: B is the right choice. The size of the body has no limit. How bout, if a certian header exists in the event with a local file path, then the the mod opens the referenced file for delivery; Otherwise it uses the body if it exists. On Jun 11, 2012 8:39 PM, "Mariusz Czulada" wrote: > 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 ' subclass="conference">....') > - 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 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120611/3ec41eb1/attachment-0001.html From SPapineni at enghouse.com Wed Jun 13 19:29:06 2012 From: SPapineni at enghouse.com (Papineni, Suneel) Date: Wed, 13 Jun 2012 15:29:06 +0000 Subject: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application Message-ID: <9438D04074E0DE45A49CD7609982127246742C4D@CORP-MAIL-002.edge.local> 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120613/0157bac5/attachment.html From mgg at giagnocavo.net Wed Jun 13 19:33:55 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Wed, 13 Jun 2012 15:33:55 +0000 Subject: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application In-Reply-To: <9438D04074E0DE45A49CD7609982127246742C4D@CORP-MAIL-002.edge.local> References: <9438D04074E0DE45A49CD7609982127246742C4D@CORP-MAIL-002.edge.local> Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B6BA9B72@BY2PRD0710MB390.namprd07.prod.outlook.com> I suggest writing the same code in C, just to troubleshoot and make sure you're calling the switch_core_session_request_uuid function properly. That exception means "access violation" aka segfault. You can also turn on the debugger and step into the native code and see where it's failing. (I'm not sure if you break on exception if it'll show you.) Basically, you have to troubleshoot it like a normal C app. -Michael From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Papineni, Suneel Sent: Wednesday, June 13, 2012 9:29 AM To: freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120613/4901a831/attachment-0001.html From peter.olsson at visionutveckling.se Wed Jun 13 20:58:09 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 13 Jun 2012 16:58:09 +0000 Subject: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B6BA9B72@BY2PRD0710MB390.namprd07.prod.outlook.com> References: <9438D04074E0DE45A49CD7609982127246742C4D@CORP-MAIL-002.edge.local>, <63B00DD1DA6A364E9F64A3A0BD2FE7B6BA9B72@BY2PRD0710MB390.namprd07.prod.outlook.com> Message-ID: <1FFF97C269757C458224B7C895F35F15106C51@cantor.std.visionutv.se> Seems strange to create a new endpoint interface on each new incoming call. That interface is something you usually create using switch_loadable_module_create_module_interface(), and that is something you do when the module is loaded (that interface is valid until the modules is unloaded again). I've never used the managed stuff myself, so I say as Michael, first try using C to get basic stuff working, and then try to port that to managed code. /Peter ________________________________ Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f?r Michael Giagnocavo [mgg at giagnocavo.net] Skickat: den 13 juni 2012 17:33 Till: freeswitch-dev at lists.freeswitch.org ?mne: Re: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application I suggest writing the same code in C, just to troubleshoot and make sure you?re calling the switch_core_session_request_uuid function properly. That exception means ?access violation? aka segfault. You can also turn on the debugger and step into the native code and see where it?s failing. (I?m not sure if you break on exception if it?ll show you.) Basically, you have to troubleshoot it like a normal C app. -Michael From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Papineni, Suneel Sent: Wednesday, June 13, 2012 9:29 AM To: freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application 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 !DSPAM:4fd8b07c32761759318328! From SPapineni at enghouse.com Wed Jun 13 21:03:42 2012 From: SPapineni at enghouse.com (Papineni, Suneel) Date: Wed, 13 Jun 2012 17:03:42 +0000 Subject: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B6BA9B72@BY2PRD0710MB390.namprd07.prod.outlook.com> References: <9438D04074E0DE45A49CD7609982127246742C4D@CORP-MAIL-002.edge.local> <63B00DD1DA6A364E9F64A3A0BD2FE7B6BA9B72@BY2PRD0710MB390.namprd07.prod.outlook.com> Message-ID: <9438D04074E0DE45A49CD7609982127246742CF7@CORP-MAIL-002.edge.local> Hi Michael, Thanks for your quick reply. I tried same function in C code and it worked fine. Also as you said, I tried to debug and the problem appears to be at "SWIGEXPORT void * SWIGSTDCALL CSharp_switch_core_session_request_uuid(void * jarg1, int jarg2, unsigned long jarg3, void * jarg4, char * jarg5){ }" function present in "freeswitch_wrap.2010.cxx" file in "mod_managed" project. In this function exception is thrown from "return jresult;" line. In this jarg4 shown as and other parameters are having values (in debug mode). I was unable to debug line by line in this function. I was unable to figure out why it is throwing this error at this function. Will it comes if interface does not exist or interface config is not loaded properly? Thanks & Regards Suneel From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Giagnocavo Sent: 13 June 2012 16:34 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application I suggest writing the same code in C, just to troubleshoot and make sure you're calling the switch_core_session_request_uuid function properly. That exception means "access violation" aka segfault. You can also turn on the debugger and step into the native code and see where it's failing. (I'm not sure if you break on exception if it'll show you.) Basically, you have to troubleshoot it like a normal C app. -Michael From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Papineni, Suneel Sent: Wednesday, June 13, 2012 9:29 AM To: freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120613/4685017d/attachment-0001.html From SPapineni at enghouse.com Wed Jun 13 21:09:26 2012 From: SPapineni at enghouse.com (Papineni, Suneel) Date: Wed, 13 Jun 2012 17:09:26 +0000 Subject: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application In-Reply-To: <1FFF97C269757C458224B7C895F35F15106C51@cantor.std.visionutv.se> References: <9438D04074E0DE45A49CD7609982127246742C4D@CORP-MAIL-002.edge.local>, <63B00DD1DA6A364E9F64A3A0BD2FE7B6BA9B72@BY2PRD0710MB390.namprd07.prod.outlook.com> <1FFF97C269757C458224B7C895F35F15106C51@cantor.std.visionutv.se> Message-ID: <9438D04074E0DE45A49CD7609982127246742D17@CORP-MAIL-002.edge.local> Hi Peter, Yes, you are right. According to the code specified here for each call there is a new interface is created. I put this here only to display less code in the mail. In fact in my application I have only one interface which is created in configLoad function. Thanks & Regards Suneel -----Original Message----- From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: 13 June 2012 17:58 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application Seems strange to create a new endpoint interface on each new incoming call. That interface is something you usually create using switch_loadable_module_create_module_interface(), and that is something you do when the module is loaded (that interface is valid until the modules is unloaded again). I've never used the managed stuff myself, so I say as Michael, first try using C to get basic stuff working, and then try to port that to managed code. /Peter ________________________________ Fr?n: freeswitch-dev-bounces at lists.freeswitch.org [freeswitch-dev-bounces at lists.freeswitch.org] f?r Michael Giagnocavo [mgg at giagnocavo.net] Skickat: den 13 juni 2012 17:33 Till: freeswitch-dev at lists.freeswitch.org ?mne: Re: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application I suggest writing the same code in C, just to troubleshoot and make sure you're calling the switch_core_session_request_uuid function properly. That exception means "access violation" aka segfault. You can also turn on the debugger and step into the native code and see where it's failing. (I'm not sure if you break on exception if it'll show you.) Basically, you have to troubleshoot it like a normal C app. -Michael From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Papineni, Suneel Sent: Wednesday, June 13, 2012 9:29 AM To: freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application 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 !DSPAM:4fd8b07c32761759318328! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 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 mgg at giagnocavo.net Wed Jun 13 21:13:27 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Wed, 13 Jun 2012 17:13:27 +0000 Subject: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application In-Reply-To: <9438D04074E0DE45A49CD7609982127246742CF7@CORP-MAIL-002.edge.local> References: <9438D04074E0DE45A49CD7609982127246742C4D@CORP-MAIL-002.edge.local> <63B00DD1DA6A364E9F64A3A0BD2FE7B6BA9B72@BY2PRD0710MB390.namprd07.prod.outlook.com> <9438D04074E0DE45A49CD7609982127246742CF7@CORP-MAIL-002.edge.local> Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B6BA9D1D@BY2PRD0710MB390.namprd07.prod.outlook.com> You may need to switch to assembly debugging to really find out where it's AV'ing. It's also possible that the SWIG glue code is incorrect, so keep that in mind when you're looking at the codegen. -Michael From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Papineni, Suneel Sent: Wednesday, June 13, 2012 11:04 AM To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application Hi Michael, Thanks for your quick reply. I tried same function in C code and it worked fine. Also as you said, I tried to debug and the problem appears to be at "SWIGEXPORT void * SWIGSTDCALL CSharp_switch_core_session_request_uuid(void * jarg1, int jarg2, unsigned long jarg3, void * jarg4, char * jarg5){ }" function present in "freeswitch_wrap.2010.cxx" file in "mod_managed" project. In this function exception is thrown from "return jresult;" line. In this jarg4 shown as and other parameters are having values (in debug mode). I was unable to debug line by line in this function. I was unable to figure out why it is throwing this error at this function. Will it comes if interface does not exist or interface config is not loaded properly? Thanks & Regards Suneel From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Michael Giagnocavo Sent: 13 June 2012 16:34 To: freeswitch-dev at lists.freeswitch.org Subject: Re: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application I suggest writing the same code in C, just to troubleshoot and make sure you're calling the switch_core_session_request_uuid function properly. That exception means "access violation" aka segfault. You can also turn on the debugger and step into the native code and see where it's failing. (I'm not sure if you break on exception if it'll show you.) Basically, you have to troubleshoot it like a normal C app. -Michael From: freeswitch-dev-bounces at lists.freeswitch.org [mailto:freeswitch-dev-bounces at lists.freeswitch.org] On Behalf Of Papineni, Suneel Sent: Wednesday, June 13, 2012 9:29 AM To: freeswitch-dev at lists.freeswitch.org Subject: [Freeswitch-dev] Issue while creating a session for an incoming call in managed application 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120613/a8497e9c/attachment-0001.html From marketing at cluecon.com Mon Jun 18 20:40:04 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 18 Jun 2012 09:40:04 -0700 Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes Message-ID: We hope you're having a nice Monday today. The past week was a busy one both for the FreeSWITCHproject and for ClueCon . Last week's community conference call was a nice discussion about some of the things that will be happening at ClueCon this year. We also had a visit from Diana Cionoiu of the YATE project. This coming Wednesday we will have Darren Schreiber, co-author of both FreeSWITCH books and co-founder of the 2600hz project , on our call to give his world-famous "SIP 101" presentation. We definitely look forward to that discussion. We also wanted to highlight an interesting blog postby our friend and community member Kristian Kielhofner. Its provocative title is: "Everything you wish you didn't need to know about VoIP." Over the years Kristian has collected a lot of knowledge about interoperability - and lack thereof - between various VoIP devices and servers. Whether you're a VoIP novice or veteran we think you will appreciate seeing this knowledge written down for the benefit of all. In ClueCon news are happy to announce two training session and a birds of a feather (BOF) meetup for this year's event. Paid attendees of ClueCon will have their choice of FreeSWITCH or OpenSIPS training. The FreeSWITCH training will be conducted by the aforementioned Darren Schreiber. The OpenSIPS training will be conducted by none other than Bogdan-Andrei Iancu, lead developer of the OpenSIPS project. The training sessions will take place on Monday, August 6. On the evening of Wednesday, August 8 we will be having a VoIP security ("VoIPSec") meetup for those interested in discussing this subject with others who share an interest. Among those who will be present are Phil Zimmermann of PGP and ZRTPfame. Be sure to register right away so that you can get your extra chances to win. Anyone who registers before July 4th of this year will receive 10 entries in the great ClueCon giveaway. We have a lot of interesting items to give away this year, so stay tuned. And while you are at the ClueCon site to register , but sure to review the schedule as we've added a number of new speakers in the past week. See you in August! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120618/48dc910a/attachment.html From jbr at consiglia.dk Tue Jun 19 18:07:16 2012 From: jbr at consiglia.dk (Jon Bruel) Date: Tue, 19 Jun 2012 16:07:16 +0200 Subject: [Freeswitch-dev] sendevent NOTIFY with uuid - the tag from RINGING response is not copied to NOTIFY Message-ID: I have tested how I can control a SIP-phone receiving a call to go off-hook. This is possible by sending a NOTIFY with the SIP Event-header set to talk. My problem is that the NOTIFY message does not include exactly the same SIP To-header information as the FS received as a RINGING response to the INVITE: The tag from the RINGING response, which the phone adds to the SIP To-header, is not included in the NOTIFY. Snom-phones - and possibly other SIP phones as well, uses this tag in the SIP To-header to pinpoint the exact call to be answered. The information is actually present before the NOTIFY is sent. I have tested it in the dialplan using the info application just before firing off the sendevent NOTIFY. Is there any reason for this behaviour, which I should be aware of before dabbling with the code? I have checked the mod_sofia.c code and have found the code which sends the NOTIFY (with the uuid, event and message type as input parameters): if (uuid && ct && es) { switch_core_session_t *session; private_object_t *tech_pvt; //Jon changed: SIPTAG_SUBSCRIPTION_STATE_STR("terminated;reason=noresource") to SIPTAG_SUBSCRIPTION_STATE_STR("active") if ((session = switch_core_session_locate(uuid))) { if ((tech_pvt = switch_core_session_get_private(session))) { nua_notify(tech_pvt->nh, NUTAG_NEWSUB(1), SIPTAG_SUBSCRIPTION_STATE_STR("active"), SIPTAG_EVENT_STR(es), SIPTAG_CONTENT_TYPE_STR(ct), TAG_IF(!zstr(body), SIPTAG_PAYLOAD_STR(body)), TAG_END()); } switch_core_session_rwunlock(session); } } But in order to change the behaviour, I may need to look somewhere else. Some hints as to where to look would be appreciated! Regards Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120619/a7ff4294/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Jon Bruel.vcf Type: text/x-vcard Size: 174 bytes Desc: Jon Bruel.vcf Url : http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120619/a7ff4294/attachment.vcf From msc at freeswitch.org Wed Jun 20 20:06:42 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 20 Jun 2012 09:06:42 -0700 Subject: [Freeswitch-dev] FreeSWITCH Conf Call Today: SIP 101 Message-ID: Hey all, Don't forget that at 1pm EDT, 10am PDT we have the FreeSWITCH community conference call. Darren Schreiber will be doing the SIP 101 discussion. We look forward to having you all with us. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120620/7c25d1e4/attachment.html From msc at freeswitch.org Thu Jun 21 19:51:28 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 21 Jun 2012 08:51:28 -0700 Subject: [Freeswitch-dev] New wiki page: Language_Files Message-ID: Hello all, I've created a new wiki page: http://wiki.freeswitch.org/wiki/Language_Files I've wanted this for some time and Brian's OCD finally took over and I just *had* to create this page. :) I've added languages that I know about or have heard about. *This page is not complete, and it needs your help.* If you know anything about any language files then please please go to that page and add what you know. If you have any issues with editing the wiki then let me know and I will be happy to assist. In fact, if need be I will do a 20-minute tutorial on wiki editing on an upcoming community conference call. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120621/873bf2c9/attachment.html From victor.velo at videotron.ca Thu Jun 21 22:00:31 2012 From: victor.velo at videotron.ca (victor.velo at videotron.ca) Date: Thu, 21 Jun 2012 14:00:31 -0400 Subject: [Freeswitch-dev] bandtel SIP TRUNK config In-Reply-To: References: Message-ID: <7660ed5b2abc0a.4fe328ff@videotron.ca> Hello, I need some help to validate my bandtel sip trunking config... I get the following info from bandtel: Your SIP User Name: 201XXXXXXX Your SIP Password: XXXXXX Your ANI: 207XXXXXX Your Toll Free/s: N/A Your DID/s: 207XXXXX (prefix with0317) --- DNS Server Addresses: >65.175.129.149 >66.237.65.90 Registration must be sent to: registrar.bandtel.com ?Outbound Calls: Outbound calls must be sent to the following proxy: Primary: proxy1.bandtel.com; Secondary: proxy2.bandtel.com my setting: ....... cat /etc/resolv.conf nameserver 65.175.129.149 nameserver 66.237.65.90 .................. cat 01_bandtel.xml ....................... cat 01_bandtel_did.xml ............. I don't get any error in the log file and I don't get anything by calling the bandtel DID.... Do I miss some settings or params somewhere???? Thank for the helps, Victor Velo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120621/0aefdf0b/attachment-0001.html From msc at freeswitch.org Mon Jun 25 06:40:22 2012 From: msc at freeswitch.org (Michael Collins) Date: Sun, 24 Jun 2012 19:40:22 -0700 Subject: [Freeswitch-dev] Are you using NDLB-force-rport = "safe" ? Message-ID: Hi gang, A friend of mine asked how many people are using this. Are you? It's mostly for Polycoms and non-polys to have force rport in a typical hosted environment. If you have been using this setting in production please give us some feedback. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120624/f27121a0/attachment.html From marketing at cluecon.com Mon Jun 25 20:09:54 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 25 Jun 2012 09:09:54 -0700 Subject: [Freeswitch-dev] FreeSWITCH Weekly News and Notes Message-ID: 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 2600hzproject joined us last week on our conference call 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 placeand the slides are available here . For this week'sconference call we are going to have an open discussion about topics of interest to our community members. Our ClueCon schedule 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 - we have this one .) Bradley J Pitt is a long-time veteran of the technology and telecom industries and recently joined Barracuda Networks . 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. 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120625/4487a535/attachment.html From dxj19831029 at gmail.com Tue Jun 26 11:05:14 2012 From: dxj19831029 at gmail.com (Xijing Dai) Date: Tue, 26 Jun 2012 15:05:14 +0800 Subject: [Freeswitch-dev] I can not see sendonly/recvonly from SDP Message-ID: 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-session ). 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120626/2bff4486/attachment.html From prasd.d.b at gmail.com Tue Jun 26 12:50:59 2012 From: prasd.d.b at gmail.com (Prasd D) Date: Tue, 26 Jun 2012 01:50:59 -0700 Subject: [Freeswitch-dev] Support for TLS and SRTP with SPA 3102 Message-ID: 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 From brian at freeswitch.org Tue Jun 26 20:36:46 2012 From: brian at freeswitch.org (Brian West) Date: Tue, 26 Jun 2012 11:36:46 -0500 Subject: [Freeswitch-dev] Support for TLS and SRTP with SPA 3102 In-Reply-To: References: Message-ID: <8EE85443-9363-4452-B289-9195548007F1@freeswitch.org> Mikey support would need to be added to FreeSWITCH as it currently doesn't support this. I'm sure that would be in mod_sofia. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 UK: +44 20 3298 4900 On Jun 26, 2012, at 3:50 AM, Prasd D wrote: > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120626/9bdb2a90/attachment-0001.html From dxj19831029 at gmail.com Wed Jun 27 12:21:16 2012 From: dxj19831029 at gmail.com (Xijing Dai) Date: Wed, 27 Jun 2012 16:21:16 +0800 Subject: [Freeswitch-dev] stream a file multicast Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120627/39156b51/attachment.html From msc at freeswitch.org Wed Jun 27 20:07:32 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 27 Jun 2012 09:07:32 -0700 Subject: [Freeswitch-dev] FreeSWITCH Conf Call Today Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120627/cdaee5eb/attachment.html From dujinfang at gmail.com Fri Jun 29 19:23:03 2012 From: dujinfang at gmail.com (Seven Du) Date: Fri, 29 Jun 2012 23:23:03 +0800 Subject: [Freeswitch-dev] questions about making a new endpoint In-Reply-To: <001756FC14C6441684351BED7517EC9A@gmail.com> References: <8ACC4EE243024FDCB048F25852C5DE8E@gmail.com> <4F5E2D34.7070201@quentustech.com> <001756FC14C6441684351BED7517EC9A@gmail.com> Message-ID: <2AC94A9435C843BD92AFCFD213ED8C55@gmail.com> Hi William, Just want to let you know that I finally got VLC working on my Mac, instead of compile the src I linked directly to the VLC lib which works fine unless it caused me a lot of trouble for debugging. And I had something new. After Hundreds of crashes I finally made video playing working, it's an app like play_fsv and because vlc decode video into YUV I have to encode back into h264 with libx264. though it wasting resources it gives a nice feature so I can play a 1080p video and resize into CIF so an ordinary SIP client can play. I had little experience on video before this and when I experience hundreds or crashes I really understand how Anthony named FreeSWITCH 1.0 - Phoenix. I successfully played a 1080p Mpeg2TS/RTP stream feed by an hardware encoder, and some .mp4 files. Others untested but it should play anything that VLC can. I tried the get the original h264 data without transcoding to save resources, it seems no existing dummy decoder/filter, it maybe easy or hard to made one. http://mailman.videolan.org/pipermail/vlc-devel/2012-June/088960.html The code is a mess at the time and when I clean down I would commit to jira for review. Thanks, Seven. On Tuesday, March 13, 2012 at 9:20 AM, Seven Du wrote: > Cool. I noticed mod_vlc the only problem is that I have problem to compile the VLC lib when I last try so I made a simple rtsp implementation. > > > I'd like to not re-invent the wheel and merge the endpoint function into mod_vlc. May I directly commit into git? Can you highlight some points on my questions Anthony ? Or maybe you have better inputs when you see the code ;) > > > On Tuesday, March 13, 2012 at 1:07 AM, William King wrote: > > > Seven, > > > > Currently the mod_vlc module supports acting as a client for rtsp > > streams. I'm in the middle of adding server support so that mod_vlc can > > stream out to clients the audio and video of a call. > > > > -William > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-dev mailing list > > FreeSWITCH-dev at lists.freeswitch.org (mailto: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 > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120629/b97ecfad/attachment.html From msc at freeswitch.org Sat Jun 30 01:11:28 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 29 Jun 2012 14:11:28 -0700 Subject: [Freeswitch-dev] Connecting FS to MS UM Message-ID: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-dev/attachments/20120629/e667de21/attachment.html From prasd.d.b at gmail.com Sat Jun 30 14:22:28 2012 From: prasd.d.b at gmail.com (Prasd D) Date: Sat, 30 Jun 2012 03:22:28 -0700 Subject: [Freeswitch-dev] Support for TLS and SRTP with SPA 3102 In-Reply-To: <8EE85443-9363-4452-B289-9195548007F1@freeswitch.org> References: <8EE85443-9363-4452-B289-9195548007F1@freeswitch.org> Message-ID: Would be great to add to mod_sofia. Can you please add it or put it in motion ? On 6/26/12, Brian West wrote: > Mikey support would need to be added to FreeSWITCH as it currently doesn't > support this. I'm sure that would be in mod_sofia. > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > iNUM: +883 5100 1420 9266 > UK: +44 20 3298 4900 > > > > > On Jun 26, 2012, at 3:50 AM, Prasd D wrote: > >> 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 > > -- Thanks, Prasd From jmesquita at freeswitch.org Sat Jun 30 21:25:43 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Sat, 30 Jun 2012 14:25:43 -0300 Subject: [Freeswitch-dev] Support for TLS and SRTP with SPA 3102 In-Reply-To: References: <8EE85443-9363-4452-B289-9195548007F1@freeswitch.org> Message-ID: <9B026B0707F4418991405F61514C8F08@freeswitch.org> Prasd, Maybe you could donate a couple of those devices to the project as well as put a bounty for it, don't ya think? Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Saturday, June 30, 2012 at 7:22 AM, Prasd D wrote: > Would be great to add to mod_sofia. Can you please add it or put it in motion ? > > On 6/26/12, Brian West wrote: > > Mikey support would need to be added to FreeSWITCH as it currently doesn't > > support this. I'm sure that would be in mod_sofia. > > -- > > Brian West > > brian at freeswitch.org (mailto:brian at freeswitch.org) > > FreeSWITCH Solutions, LLC > > PO BOX PO BOX 2531 > > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > > T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST > > iNUM: +883 5100 1420 9266 > > UK: +44 20 3298 4900 > > > > > > > > > > On Jun 26, 2012, at 3:50 AM, Prasd D wrote: > > > > > 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 > > > > > > > > > > > -- > Thanks, > Prasd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-dev mailing list > FreeSWITCH-dev at lists.freeswitch.org (mailto: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 > > -------------- next part -------------- An HTML attachment was scrubbed... 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