May 2013 - freeswitch-dev@lists.freeswitch.org

[Freeswitch-dev] FreeSWITCH for WebRTC
by Richard Screene 24 Jun '13

24 Jun '13

Is there any news on when WebRTC support will be added to FreeSWITCH? It would be awesome if we could host a video conference using FreeSWITCH. Regards, Richard

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[Freeswitch-dev] Hey Guys Today 3PM EST Friday FreeForAll
by Ken Rice 31 May '13

31 May '13

Hey Guys don¹t forget to join us at 3PM EST (that¹s like NOON PST) for the FridayFreeForAll For those of you Voice Only, sip:888@conference.freeswitch.org as usual For those that want to join with Video, sip:888@mcu.freeswitch.org For those that have the option, try the video Bridge lets see if we can find a breaking point with load! K -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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[Freeswitch-dev] mod_event_socket
by Georgi Stefanov 30 May '13

30 May '13

Hi All there :-) Recently I have played with mod_event_socket It works great and makes me happy, but recent days I have noticed FS send to my application CHANNEL_ANSWER after I hangup the call - something like it is blocked and after I hangup it is unblocked. It doesn't do it every time - in fact it is very rare. I want to ask: Does anybody else have noticed something similar ? I have only one call which start playing about 20 files

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[Freeswitch-dev] FreeSWITCH Conference Call Today
by Michael Collins 29 May '13

29 May '13

Hello all, Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_29 We have Martin from VoIPMonitor.org who will be sharing with us some information about the project, both the OSS piece and the commercial GUI front-end. Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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[Freeswitch-dev] FreeSWITCH Weekly News and Notes
by Michael Collins 28 May '13

28 May '13

Happy Tuesday to you all! It is a short week for us here in the U.S. as yesterday was a national holiday. The first order of business: I am happy to let you know that Packt Publishing <http://www.packtpub.com> has informed us that the new FreeSWITCH 1.2 book <http://www.packtpub.com/freeswitch-1-2/book> is available as of May 24th! Congrats to the FreeSWITCH team for getting yet another book published. This is the third one we've done with Packt. Also, many thanks to all those who spent time answering questions and helping us with the technical reviewing process. With the new book completed we focus our attention on other things, not the least of which is ClueCon 2013 <http://www.cluecon.com>. Starting this week, those who've submitted talk proposals will be receiving the official acceptance of their presentations, including the day and time for the presentation. If you have not already submitted a talk proposal please do so right away as we have a limited number of available speaking slots. Our weekly conference calls have been lively with discussions on various topics suggested by those calling in. On last week's call<http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_22>we had a particularly nice discussion about mod_skinny <http://wiki.freeswitch.org/wiki/Mod_skinny>. A relatively new community member, Nathan Neulinger, has really done a nice job of picking up the torch for mod_skinny. If you have a need to use Cisco phones with FreeSWITCH using SCCP<http://en.wikipedia.org/wiki/Skinny_Call_Control_Protocol>then definitely listen <http://torrents.freeswitch.org/conf_call_2013-05-22.torrent> to last week's discussion. This week <http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_29> we have Martin from the VoIPMonitor.org project. We look forward to learning more about VoIPMonitor and how it can help us with keeping tabs on our VoIP servers. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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[Freeswitch-dev] ANNOUNCEMENT: ClueCon Hotel Registration Available Online
by Michael Collins 22 May '13

22 May '13

The ClueCon team would like to announce that the hotel for ClueCon 2013 may now be booked online. Please visit www.ClueCon.com and click the "Book Your Room Online" button. Once you are at the Hyatt site you can enter in your check in and checkout dates and click the Search button. You will be presented with your choice of "standard king" or "standard double" rooms. Keep in mind that the standard double rooms go fast, so don't delay! Be sure to register <http://www.cluecon.com/register/> for ClueCon right away so that we can plan our various activities for when we're all in Chicago. See you in August! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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[Freeswitch-dev] FreeSWITCH Conference Call Today
by Michael Collins 22 May '13

22 May '13

Hello all. Our community conference call is coming up shortly. The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_22 We don't have a feature presentation today, but we do have several items to discuss so please join us at 1PM EDT, 10AM PDT. Thanks! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Re: [Freeswitch-dev] Same headers in a SIP message
by Brian West 22 May '13

22 May '13

Opening a Jira doesn't automagically get what you want done, you should probably put a bounty on it or take this chance to learn how to modify the source and contribute patches. Its how I got started way back when first started using Asterisk. /b On May 14, 2013, at 7:06 AM, Kevin Mathy <k.mathy(a)hexanet.fr> wrote: > Hi Brian, > > I've already opened a JIRA, and I'm waiting for improvement from other devs... Concerning my providers, the fact is not that they'll refuse messages with unknown headers, but that they'll not interpret those headers, so they'll miss some informations (here, divert informations) > > But, I agree with you for the solution to beat them with RFC3261 :-) > > Bien cordialement, > Best Regards, > > Kevin MATHY

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[Freeswitch-dev] FreeSWITCH Weekly News and Notes
by Michael Collins 20 May '13

20 May '13

Hello! The FreeSWITCH team is busy on several fronts but the most exciting news at the moment is that Packt Publishing has told us the new book<http://www.packtpub.com/freeswitch-1-2/book>is officially off to production! If all goes well it should be available in the next few weeks. (It looks like they changed the cover art - once we know for sure what's on the cover we'll give it a nickname like we did with the "bridge" book.) Last week we had an impromptu discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_15>on various FreeSWITCH topics. We have rescheduled the VoIPMonitor.org conference call presentation for May 29th. Also, we have tentatively scheduled the HOMER presentation for June 5. That means we will have another open discussion<http://wiki.freeswitch.org/wiki/FS_weekly_2013_05_22>for this Wednesday's conference call. As a tip, if you would like to browse around and see what has been committed in FreeSWITCH lately you can use the Fisheye site<http://fisheye.freeswitch.org/>. Here you can browse the source code and look at recent commits to see what has changed. If you prefer to look at the commits without the source code browser then try out the HTML front-end <http://fisheye.freeswitch.org/>for our FreeSWITCH git repository. We also have repositories for contributions <http://git.freeswitch.org/git/freeswitch-contrib/> and sample configuration <http://git.freeswitch.org/git/freeswitch-sample-configs/>sets. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE

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[Freeswitch-dev] VAD Events stop Firing after Bridgine
by Rony Beck 20 May '13

20 May '13

Hi All, I have a problem with Voice Activity Detection. I am writing a module which will bridge a call to a conference when it detects voice activity on the line. When voice activity isn't detected for a period, the call is parked until voice activity is detected again. It seems that before the call is bridged to the conference, Voice Activity Detection works fine and events with the "Action" of "start-talking" or "stop-talking" are fired. Once the call is bridged, these events are no longer fired. Even If I manually park the call from fs_cli using "uuid_park", the events with "start-talking"/"stop-talking" are never fired again. I have been scouring the net looking for a means of getting these events to fire again. The sip profiles involved all have VAD enabled in both directions. In code, I start the conference call like so: sprintf(szCommand,"{TALK, NOTALK}sofia/externalx/confuser@%s:5080",szConfAddress); switch_ivr_originate( NULL,&pPersistentSession,cause,szCommand, iTimeout, NULL,NULL,NULL,NULL,NULL, SOF_FORKED_DIAL,NULL ) != SWITCH_STATUS_SUCCESS; pMainSessionUUID = switch_core_session_get_uuid( pPersistentSession ); pPersistentChannel = switch_core_session_get_channel( pPersistentSession ); pAppName = switch_core_session_strdup( pPersistentSession, "conference" ); sprintf(szCommand,"%s@default",pExtension); pAppOpt = switch_core_session_strdup( pPersistentSession, szCommand); pPersistentCallExtension = switch_caller_extension_new(pPersistentSession, pAppName, pAppOpt); switch_caller_extension_add_application( pPersistentSession, pPersistentCallExtension, pAppName, pAppOpt ); switch_channel_set_caller_extension( pPersistentChannel, pPersistentCallExtension ); switch_channel_set_state( pPersistentChannel, CS_EXECUTE ); This is just calling another freeswitch box which answers the call and keeps the conference open. I then make a second call to the user's sip phone in much the same way. sprintf(szCommand,"{TALK, NOTALK}user/1001"); switch_ivr_originate( NULL, &pUserSession,&cause, szCommand, iTimeout,NULL,NULL,NULL,NULL,NULL, SOF_FORKED_DIAL,NULL ); pUserSessionUUID = switch_core_session_get_uuid( pUserSession ); pUserChannel = switch_core_session_get_channel( pUserSession ); pUserAppName = switch_core_session_strdup( pUserSession, "conference" ); sprintf(szCommand,"%s@default",USER_EXTENSION); pUserAppOpt = switch_core_session_strdup( pUserSession, szCommand); pUserCallerExtension = switch_caller_extension_new(pUserSession, pUserAppName, pUserAppOpt); switch_caller_extension_add_application( pUserSession, pUserCallerExtension, pUserAppName, pUserAppOpt ); switch_channel_set_caller_extension( pUserChannel, pUserCallerExtension ); switch_channel_set_state( pUserChannel, CS_EXECUTE ); At this point the events are firing fine and my event handler is reacting to them. On the first "start-talking" event I get from the user, I bridge the calls like so: switch_ivr_uuid_bridge (pMainSessionUUID,pUserSessionUUID); At this point, the events for "start-talking" or "stop-talking" are no longer fired and are never fired again even if I break the bridging. I tried commenting out the bridging line just to be sure that the events are fired without the briding and it looks fine. I get all the events continuously. So it seems to break as soon as I bridge. To capture the events, I had expected that I should bind to the events like this: switch_event_bind("START_TALK_EVENT", SWITCH_EVENT_TALK, SWITCH_EVENT_SUBCLASS_ANY, detect_talk , NULL); switch_event_bind("END_TALK_EVENT", SWITCH_EVENT_NOTALK, SWITCH_EVENT_SUBCLASS_ANY, detect_talk , NULL); This never worked. What did work was this: switch_event_bind("TALK_EVENT", SWITCH_EVENT_CUSTOM, "conference::maintenance", detect_talk , NULL); My event handler would just look for an event where the "Action" was set to "start-talking" or "stop-talking". Is there something I am missing here? I don't find enough information on the wiki to help me work out exactly how VAD events should be listened for and handled, so it is entirely likely I am doing it wrong. Can someone help me here or event paste a working code sample? Many thanks for your time, Ronnie

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