May 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Disable early media (how to)?
by Vlasis Hatzistavrou (KTI) 13 May '09

13 May '09

Hello, I am trying to understand how to disable early media in both directions during the call negotiation. In the following scenario: Originator SIP Client -- SIP --> FS with ignore_early_media=true, FS starts sending RTP audio to the originator right after the "200 OK" is sent from FS to originator. This is correct. However, FS sends SDP information in the Session Progress (183) message before the "200 OK", and the originator client sends RTP before the call is answered. The strange thing is that SDP is sent again in the "200" message although it is already sent in the 183 message (I don't know if this redundant behavior is acceptable in SIP, I have a mostly H323 background) So, in this way early media is disabled only for the audio from FS to the client, but not from the client to FS. My question is: how can I prevent FS from sending SDP in the 183 message and send it only in the 200 message? I searched the mailing list archives but didn't come up with anything. Any help is appreciated. -- Best regards, Vlasis Hatzistavrou.

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Cluecon 2009 - Hotel Special!
by Michael Collins 12 May '09

12 May '09

Good news! For those of you who like to get a good deal on a hotel room we've got the inside scoop on a special for the Wyndham Chicago. I just did a search on expedia.com for a hotel with these specs: Hotel only Check-in: 8/3/2009, Check-Out: 8/6/2009 1 room, 2 adults City: Chicago Hotel Name: "Wyndham Chicago" The results were very nice: the non-refundable rate per night is only $140!!! Get to expedia.com NOW and get signed up! We look forward to seeing you in Chicago this August. -Michael http://www.cluecon.com

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SDP Passthrough, INVITE messages.
by Juan Manuel Vicente 12 May '09

12 May '09

Hi, I'm trying to use the Freeswitch as a proxy (I know that is not designed for that, but I really need to do it in this way), here is my config: Endpoint 1<-> FS A-<->FS B<->FS A<->Endpoint 2 * Both Endpoints are registered in FS A how is acting as a proxy and registrar. * FS B only sends back the Invite to FS A in order to reach Endpoint 2. * Both FS have a public IP * FS A Only handles SIP messages * FS B Handles RTP (Also SIP) My objetive is to keep the signaling in FS A and the RTP in FS B so basically FS A will work as a registrar. So far I've been able to succesfully do it if both endpoint are not nated, how ever I do need to do it in a Natted sceneario too, for what I have been sniffing the problem is that in the INVITE, the SDP is sending the internal IP instead of the external. I've tried to change the switch_r_sdp and switch_l_sdp but I'm not quite sure if I'm doing the correct config of the switch (late_codec_negotiation) If anyone could give a tip or a sample of how can I change the INVITE messages I will appreciate. Thanks in advance Regards

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ANNOUNCEMENT: www.freeswitch.org site updated
by Michael Collins 12 May '09

12 May '09

FYI, We'd like to let the community know that we've moved the main site, www.freeswitch.org, to a new server and have updated it to the latest Drupal version. We are still calibrating the server so if the site seems slower than usual please let me know off list and we'll take a look. Also, because this upgrade was so major we were forced to wipe out all the users from the previous Drupal install. Please feel free to sign up again if you so desire. You only need to sign up if you wish to post comments. One other new feature is the "Share This" button. (You need JavaScript enabled to see this.) It makes it as easy as possible to share stories with Digg, SU, Deli.ico.us, etc. If there are any stories you've not already dugg then please try out the Share This button to see how easy it is to digg stuff now. As always, if you have any questions or comments then please email me off list and I'll address them quickly. -Michael http://www.cluecon.com

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mod_event_socket 183 early media
by İbrahim TUNALI 12 May '09

12 May '09

Hi, I am trying to build a SBC but a kind of different described on wiki. I immediately send incoming leg A to a socket app and connect bridge with my socket python application. I will handle all business logic on python side. My dialplan is so simple; <extension name="SBCSocket"> <condition field="destination_number" expression="^1000$"> <action application="socket" data="127.0.0.1:1905 async full"/> </condition> </extension> Freeswitch send calls to parking and 183 early media through mod_sofia. This is unacceptable on my SBC scenario. Because I not receive 183 or 180 from leg B yet. I want to relay early media or ringing when I received from B leg I realize this is not a bug. It is a feature but i need to know is this possible. I would appreciate all suggestions and clues. Regards, ibrahim

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Re: [Freeswitch-users] Freeswitch Java Socket Interface API
by Ceolter 12 May '09

12 May '09

Hi Klaus, I was wondering if you are still working on the Java Socket Interface for Freeswitch. I'd be interested in using it if it exists. Thanks, Niall. Klaus L'Imbecile wrote: > > Hi Folks, > > Just to let you know that we are working on a library for connecting to > the Freeswitch via the socket interface. We plan to release it under LGPL > as soon as it's somewhat robust. > > > And for those of you who are in the USA, please don't forget to go vote. > > Thanks, > Klaus. > -- > "Feel free" - 10 GB Mailbox, 100 FreeSMS/Monat ... > Jetzt GMX TopMail testen: http://www.gmx.net/de/go/topmail > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- View this message in context: http://www.nabble.com/Freeswitch-Java-Socket-Interface-API-tp20303263p23462… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Re: [Freeswitch-users] get call durantion
by Diego Toro 12 May '09

12 May '09

Hi, I need get call duration after bridge application using mod_managed, my code: Session.Execute("bridge", sbNewOutBoundNum); Debug("billsec :" + _Session.GetVariable("billsec")); Debug("duration :" + _Session.GetVariable("duration")); The bridge is ok, but the variable value duration and billsec is zero (0). Diego --- On Sun, 5/10/09, Michael S Collins <msc(a)freeswitch.org> wrote: Do you mean from the CDR? I recommend XML CDRs because they give tons of information. If you are talking about gathering this stuff midcall then you'll need to supply more information about your setup. -MC Sent from my iPhone On May 10, 2009, at 6:17 PM, Diego Toro <dftoro(a)yahoo.com> wrote: Hi, How can I get call durantion after bridge application ? I tried with billsec and duration but I don't get any value. Thank you _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ Freeswitch-users mailing list Freeswitch-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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rdnis variable from Lua
by Cliff Wells 12 May '09

12 May '09

Hi, I can see the RDN in the log file, but don't know how to retrieve it from a Lua script. Regards, Cliff

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How to enable debug in dingaling?
by Mark Campbell-Smith 11 May '09

11 May '09

Hi! I need to enable debug mode in dingaling as I can't see that freeswitch is coming online in gtalk. I have changed the following: changed the loglevel to debug in console.conf.xml changed the debug level to 1 in dingaling.conf to 1 I do not see any xmpp logs in the console or in freeswitch.log file. All I can see in the window is: 2009-05-12 17:56:35 [DEBUG] mod_dingaling.c:1854 init_profile() Started Thread for myfreeswitchname@gmail.com/gtalk@XML 2009-05-12 17:56:35 [CONSOLE] switch_loadable_module.c:857 switch_loadable_module_load_file() Successfully Loaded [mod_dingaling] 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:141 switch_loadable_module_process() Adding Endpoint 'dingaling' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_debug' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_pres' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_logout' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:259 switch_loadable_module_process() Adding API Function 'dl_login' 2009-05-12 17:56:35 [NOTICE] switch_loadable_module.c:354 switch_loadable_module_process() Adding Chat interface 'jingle' Where is the XMPP traces? Thanks

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Cluecon 2009 News
by Michael Collins 11 May '09

11 May '09

FYI, for those of you keeping up on ClueCon 2009 please visit the latest blog entry: http://cluecon.com/node/29 Thanks, Michael

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