- FreeSWITCH-dev - FreeSWITCH Mailing Lists

Urgent Feature Request - RTCP report blocks
by jay binks 19 Jul '11

19 Jul '11

I NEED to get Freeswitch generating RTCP Report Blocks ( with all associated data - packet_loss, jitter etc ) Im happy to consider a bounty on this requirement as I need this done as quickly as humanly possible. in preference to only filling my request, a complete RTCP implementation would be great however currently I simply need RTCP packet generation with Packet Loss Jitter and RTT information. we need this is to fulfill an interop test we are trying to get done, so as you can imagine time is of the essence. please contact me if you wish to work on this, to be eligible for bounty payment the code will need to be accepted into the FS Git tree and pass my testing. ( however I would like to test asap, even before inclusion in git ) I would like your guidance, as to what bounty you think this would require. please contact me for bounty approval. this email does not constitute acceptance of any bounty / work. Sincerely Jay Binks

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Associating a Dialplan with a Session
by Kurtis Heimerl 19 Jul '11

19 Jul '11

Hello FS-Dev! I'm trying to figure out how to get associate a dialplan with a newly created Session. I create the new session when I receive an chat message. Here's the current state of the code: http://pastebin.freeswitch.org/16844 There are (undoubtedly) many things wrong with this code... but I'm focused on getting it to route the SMS through the dialplan right now. It fails in the following way: ... 2011-07-18 15:05:49.315430 [DEBUG] mod_smqueue.c:39 laaa CHANNEL ROUTING 2011-07-18 15:05:49.315430 [DEBUG] switch_core_state_machine.c:77 laaa Standard ROUTING 2011-07-18 15:05:49.315430 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2011-07-18 15:05:49.315430 [DEBUG] switch_channel.c:2641 (laaa) Callstate Change RINGING -> HANGUP 2011-07-18 15:05:49.315430 [NOTICE] switch_core_state_machine.c:143 Hangup laaa [CS_ROUTING] [NO_ROUTE_DESTINATION] 2011-07-18 15:05:49.315430 [DEBUG] switch_channel.c:2657 Send signal laaa [KILL] ... The "No Route, Aborting" debug message SEEMS to indicate either that the extension isn't found, or that the dialplan isn't found. I think I've confirmed it's the dialplan by passing NULL instead of "globals.dialplan" in my own code and getting the same result. This is also confirmed by the line: switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "KURTIS %p\n", (void*) switch_loadable_module_get_dialplan_interface("default")); printing "(nil)" instead of returning a valid pointer. I assume there must be some other naming mechanism for dialplans that I'm not using correctly. I pass "default" right now (as I saw in mod_dingaling.c) but that's not working. Any suggestions would be appreciated!

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Regarding the patch for Multiple grammar ASR support
by Yungwei Chen 18 Jul '11

18 Jul '11

Hi, I am interested in taking advantages of the patch (http://jira.freeswitch.org/browse/FS-2906) for Multiple grammar ASR support from a javascript program. But I am not sure exactly when to call the following function from a javascript program: ... this.addMultipleGrammars = function () { // hardcoded urls are just for my initial tests. this.session.execute("detect_speech", "grammar " + "http://192.168.16.9/pizza_arso.grxml"); this.session.execute("detect_speech", "grammaron " + "http://192.168.16.9/pizza_arso.grxml"); } ... I tried the following in scripts/js_modules/SpeechTools.jm, but this didn't work because the code snippet at the bottom prevented me from having multiple active grammars. ..... while(this.asr.session.ready() && this.collected_index < this.req) { var x; this.needConfirm = false; if (!rv) { this.asr.addMultipleGrammars(); rv = this.react(this.top_sound, this.top_sound); } ... I had to comment out the following piece of code in recog_asr_load_grammar function of /opt/freeswitch-snapshot/src/mod/asr_tts/mod_unimrcp/mod_unimrcp.c to get my expected results. This fix may not be needed. Perhaps you can point me to the right direction. Thanks. ... start_recognize = (char *) switch_core_hash_find(schannel->params, "start-recognize"); if (zstr(start_recognize) || strcasecmp(start_recognize, "false")) { // this disables all grammars /* switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "calling recog_channel_disable_all_grammars()\n"); if (recog_channel_disable_all_grammars(schannel) != SWITCH_STATUS_SUCCESS) { status = SWITCH_STATUS_FALSE; goto done; } */ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "calling recog_channel_enable_grammar()\n"); if (recog_channel_enable_grammar(schannel, name) != SWITCH_STATUS_SUCCESS) { status = SWITCH_STATUS_FALSE; goto done; } switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "calling recog_channel_start()\n"); status = recog_channel_start(schannel); }

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Re: [Freeswitch-dev] Freeswitch and Google PUSH (C2DM)
by Nacho 18 Jul '11

18 Jul '11

Hi again. I'm having a little problem with the Java ESL. I can do some basic things like answer the call, subscribe to events, put music in the caller... Thought I have som questions about API commands. The flow right now in my program is this one: - I begin listenning at port 8084 for ESL events - If I receive an event with the name "channel_data" I begin the iteration - I answer the call with *sendMsg, addcallcommand("execute") and addExecuteAppName("answer");* - I subscribe to all events for debuging, I just use *EslMessage response = sendSyncSingleLineCommand(channel, "event text all");* - I put a tone on the caller with *sendMsg, addcallcommand("execute") and addExecuteAppName("playback");* - Now here is a tricky part, I send a petition with HttpURLConnection to an APP server, this server is the one that controls the C2DM stuff, he answers me when he sends the push and I continue the flow, but still have to wait for the legB to connect to the freeswitch server - I have to control when the legB gets connected to freeswitch, this is something i have to work on, i don't want to be making continuous requests to freeswitch to avoid overloading, so right now I assume legB is connected by default. - It's time to originate the call, so I send *sendMsg, addcallcommand("execute"), addExecuteAppName("set") and addExecuteAppArg("{ignore_early_media='true'}sofia/internal/1002%10.166.108.139"); * . *I'm using a default destiny for testing* - I have just tested that this also works *EslMessage response = sendSyncSingleLineCommand(channel, "api originate {ignore_early_media='true'}sofia/internal/1002%10.166.108.139"); * - And finally I bridge the call with SendMsg, addCallCommand("execute"), addExecuteAppName("bridge"), addExecuteAppArg("sofia/internal/1002%10.166.108.139") and addEventLock(); This works well, but right now my major issue is knowing when the legB gets connected to freeswitch so I can begin originating the call, but I have some questions that I would like to know the answer too: - The execute messages are for APP so they work like in the dialplan <action> field, I saw somewhere that to see the complete list one should look in the dialplan.xml, but I haven't found anything about a complete list :S - Things can be accomplished with API calls too, but I just dont get why I have to use "api" before "originate" to make a call, but not needing it for the events, it's a little confusing to use API commands like this, is there any simpler way to do it? - This makes my last question, is there any way to make freeswitch tell me when someone registers in the server or do I need to be continuosly asking freeswith with an API command if it is connected Thinks are going well, I have to begin working with the android client next to be able to process the C2DM messages, right now im using a default voip client. Sorry for the long post and thanks for your attention. Regards Jose P.S: Again, thanks to davidv for making possible the use of a Java ESL client :) 2011/7/11 Nacho <leucaruth(a)gmail.com> > Cool, thanks for the tip David, I'm using your library and have it working. > > The wiki should advice about the SWIGed library and it's problems with Java > and outbound connections. > > In a few days I will have news about the my progress. > > Regards, > > Jose > > 2011/7/11 david varnes <david.varnes(a)gmail.com> > >> Jose, >> >> If you want to use Java and ESL there are other alternatives >> than the SWIGed library. >> >> http://wiki.freeswitch.org/wiki/Java_ESL >> >> I maintain the one here http://wiki.freeswitch.org/wiki/Java_ESL_Client >> >> I have a few pending changes that I want to commit for a new >> release, but the existing release is stable. >> >> hope this helps >> davidv >> >> On 11 July 2011 00:12, Nacho <leucaruth(a)gmail.com> wrote: >> > I just noticed short before you answered :). >> > I have been reading it and I found it very interesting, but I have found >> a >> > problem in Java. >> > The constructor for an Outbound connection requires the File Descriptor >> (an >> > Int parameter) of the listening socket to create the ESL object, but as >> far >> > as I know, Java doesn't allow to get the int value of a File Descriptor >> from >> > a Socket. This would be veary easy to achieve in C, but I have no found >> any >> > means to get this to work with Java in Linux (Windows is out of the >> > question). >> > I tried to see the source code of the ESL client to see how it uses the >> int >> > value, but as I suspected is C code wrapped with SWIG. I have thought >> about >> > other ways to work with it, like using the FileDescriptor class in java >> and >> > Reflection, but I have not found any way to get the desired int value, >> so >> > i'm stuck about what to do right now. >> > Any ideas? if the constructor requires the int value of the File >> Descriptor >> > means that there is a way to get it, doesn't it? >> > Regards, >> > Jose >> > >> > 2011/7/7 Kristian Kielhofner <kris(a)kriskinc.com> >> >> >> >> http://wiki.freeswitch.org/wiki/Event_Socket_Library >> >> >> >> On Thu, Jul 7, 2011 at 7:35 AM, Nacho <leucaruth(a)gmail.com> wrote: >> >> > Hi again >> >> > Im doing as Kristian suggested, at first is an easier solution that >> >> > making a >> >> > module. >> >> > I created a plan that redirects calls to a local port where I will do >> >> > the >> >> > testings with the server that will work with that call. >> >> > The call is parked and the connection is stablished as I can see with >> >> > netcat, so I was thinking about making a program with C or Java that >> >> > will >> >> > become the custom server. The problem is that I don't find info about >> >> > the >> >> > structure of the packages I'll receive from Freeswitch, how to deal >> with >> >> > them or even how can I send orders to freeswitch. >> >> > Any suggestions about where I could find some info about it would be >> >> > kindly >> >> > appreciated. >> >> > Regards, >> >> > Jose. >> >> > 2011/7/6 Robert-iPhone <rhuddleston(a)gmail.com> >> >> >> >> >> >> I have interest too! Considering mwi / voicemail push for apple >> >> >> >> >> >> Sent from my iPhone >> >> >> >> >> >> On Jul 5, 2011, at 5:49 PM, Anthony Minessale >> >> >> <anthony.minessale(a)gmail.com> wrote: >> >> >> >> >> >> > it can be left out of FS code but still part of contrib if an >> >> >> > external >> >> >> > solution is developed. >> >> >> > >> >> >> > >> >> >> > On Tue, Jul 5, 2011 at 1:20 PM, Kristian Kielhofner >> >> >> > <kris(a)kriskinc.com> >> >> >> > wrote: >> >> >> >> Jose, >> >> >> >> >> >> >> >> Whether it's C2DM or APNS (Apple) this functionality is best >> left >> >> >> >> out of FreeSWITCH. FreeSWITCH has plenty of other existing means >> >> >> >> (XML_CURL, socket, etc) to drive dynamic call functionality and >> >> >> >> interact with other technologies. >> >> >> >> >> >> >> >> How do you ask? An imcoming call to FreeSWITCH could hit static >> >> >> >> dialplan and execute a socket connection to some other custom >> >> >> >> server. >> >> >> >> This server could tell FreeSWITCH to do something (play media, >> >> >> >> ringback, messages, whatever) while it sends a PUSH (via C2DM or >> >> >> >> APNS, >> >> >> >> for example). >> >> >> >> >> >> >> >> One could implement such a socket program easily. It can be >> >> >> >> extended and maintained separately as new push technologies >> become >> >> >> >> available. >> >> >> >> >> >> >> >> On Mon, Jul 4, 2011 at 7:48 AM, Nacho <leucaruth(a)gmail.com> >> wrote: >> >> >> >>> Hello all. >> >> >> >>> My name is Jose. This is my first post here. I joined the list >> >> >> >>> because >> >> >> >>> I was >> >> >> >>> wondering about developing a feature for Freeswitch and I would >> >> >> >>> like >> >> >> >>> to know >> >> >> >>> your opinion. >> >> >> >>> The idea is about making a non-persistent client for cell phones >> >> >> >>> that >> >> >> >>> works >> >> >> >>> with PUSH technology (C2DM and Android for example). These >> clients >> >> >> >>> wouldn't >> >> >> >>> be connected to the Freeswitch server at first, but if there is >> an >> >> >> >>> incoming >> >> >> >>> call, the Freeswitch server would send a PUSH message to these >> >> >> >>> clients, the >> >> >> >>> client would process it and, if accepted by the user, the client >> >> >> >>> would >> >> >> >>> awake, connect to Freeswitch and then receive the the call >> invite >> >> >> >>> message >> >> >> >>> and accept it automatically. >> >> >> >>> I don't know if this idea is plausible due to the real time >> >> >> >>> restrictions we >> >> >> >>> have to face in phone calls. If the delay introduced by C2DM >> >> >> >>> delivery >> >> >> >>> is >> >> >> >>> high, the waiting time for the caller is probably something that >> he >> >> >> >>> isn't >> >> >> >>> willing to accept in order to get his call answered. >> >> >> >>> Another question is about the battery life time it would save >> >> >> >>> because >> >> >> >>> the >> >> >> >>> application woulnd't need to be answering the "still alive" ACKS >> >> >> >>> from >> >> >> >>> Freeswitch. >> >> >> >>> I'm new in freeswitch developing and still learning about it, >> but >> >> >> >>> I'm >> >> >> >>> really >> >> >> >>> interested in this, so any help would be kindly appreciated. >> >> >> >>> -- >> >> >> >>> Aquellos que hablan son esclavos de sus palabras y los que >> callan >> >> >> >>> dueños de >> >> >> >>> su silencio. >> >> >> >>> >> >> >> >>> _______________________________________________ >> >> >> >>> Join us at ClueCon 2011, Aug 9-11, Chicago >> >> >> >>> http://www.cluecon.com 877-7-4ACLUE >> >> >> >>> >> >> >> >>> FreeSWITCH-dev mailing list >> >> >> >>> FreeSWITCH-dev(a)lists.freeswitch.org >> >> >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> >> >> >>> >> >> >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-dev >> >> >> >>> http://www.freeswitch.org >> >> >> >>> >> >> >> >>> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> >> Kristian Kielhofner >> >> >> >> >> >> >> >> _______________________________________________ >> >> >> >> Join us at ClueCon 2011, Aug 9-11, Chicago >> >> >> >> http://www.cluecon.com 877-7-4ACLUE >> >> >> >> >> >> >> >> FreeSWITCH-dev mailing list >> >> >> >> FreeSWITCH-dev(a)lists.freeswitch.org >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> >> >> >> >> >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-dev >> >> >> >> http://www.freeswitch.org >> >> >> >> >> >> >> > >> >> >> > >> >> >> > >> >> >> > -- >> >> >> > Anthony Minessale II >> >> >> > >> >> >> > FreeSWITCH http://www.freeswitch.org/ >> >> >> > ClueCon http://www.cluecon.com/ >> >> >> > Twitter: http://twitter.com/FreeSWITCH_wire >> >> >> > >> >> >> > AIM: anthm >> >> >> > MSN:anthony_minessale@hotmail.com >> >> >> > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com >> >> >> > IRC: irc.freenode.net #freeswitch >> >> >> > >> >> >> > FreeSWITCH Developer Conference >> >> >> > sip:888@conference.freeswitch.org >> >> >> > googletalk:conf+888@conference.freeswitch.org >> >> >> > pstn:+19193869900 >> >> >> > >> >> >> > _______________________________________________ >> >> >> > Join us at ClueCon 2011, Aug 9-11, Chicago >> >> >> > http://www.cluecon.com 877-7-4ACLUE >> >> >> > >> >> >> > FreeSWITCH-dev mailing list >> >> >> > FreeSWITCH-dev(a)lists.freeswitch.org >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> >> >> > >> >> >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-dev >> >> >> > http://www.freeswitch.org >> >> >> >> >> >> _______________________________________________ >> >> >> Join us at ClueCon 2011, Aug 9-11, Chicago >> >> >> http://www.cluecon.com 877-7-4ACLUE >> >> >> >> >> >> FreeSWITCH-dev mailing list >> >> >> FreeSWITCH-dev(a)lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-dev >> >> >> http://www.freeswitch.org >> >> > >> >> > >> >> > >> >> > -- >> >> > Aquellos que hablan son esclavos de sus palabras y los que callan >> dueños >> >> > de >> >> > su silencio. >> >> > >> >> > _______________________________________________ >> >> > Join us at ClueCon 2011, Aug 9-11, Chicago >> >> > http://www.cluecon.com 877-7-4ACLUE >> >> > >> >> > FreeSWITCH-dev mailing list >> >> > FreeSWITCH-dev(a)lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-dev >> >> > http://www.freeswitch.org >> >> > >> >> > >> >> >> >> >> >> >> >> -- >> >> Kristian Kielhofner >> >> >> >> _______________________________________________ >> >> Join us at ClueCon 2011, Aug 9-11, Chicago >> >> http://www.cluecon.com 877-7-4ACLUE >> >> >> >> FreeSWITCH-dev mailing list >> >> FreeSWITCH-dev(a)lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> >> http://www.freeswitch.org >> > >> > >> > >> > -- >> > Aquellos que hablan son esclavos de sus palabras y los que callan dueños >> de >> > su silencio. >> > >> > _______________________________________________ >> > Join us at ClueCon 2011, Aug 9-11, Chicago >> > http://www.cluecon.com 877-7-4ACLUE >> > >> > FreeSWITCH-dev mailing list >> > FreeSWITCH-dev(a)lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev >> > http://www.freeswitch.org >> > >> > >> >> >> >> -- >> david varnes >> >> e: david.varnes(a)gmail.com >> p: +61 404 925 633 >> > > > > -- > Aquellos que hablan son esclavos de sus palabras y los que callan dueños de > su silencio. > -- Aquellos que hablan son esclavos de sus palabras y los que callan dueños de su silencio.

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FS-2875 SIP REGISTER Fetch (without Contact header)
by Leon de Rooij 15 Jul '11

15 Jul '11

Hi all, Can someone please take a look at http://jira.freeswitch.org/browse/FS-2875 ? Especially the last few comments and last patch which is here: http://jira.freeswitch.org/secure/attachment/14120/sofia_reg_enable_registe… With this it is possible to fetch bindings by sending a sip register without contact headers. A regular register also responds with all bindings that are in the db now (see traces below). Thanks, Leon --- Traces (I left out headers that are not important for this): * UAC does REGISTER fetch (without contact) when there are no bindings in the db yet: REGISTER sip:172.16.44.8 SIP/2.0 Via: SIP/2.0/UDP 172.16.42.150:5060;rport;branch=z9hG4bKD33480AB9580F76B From: <sip:1000@172.16.44.8>;tag=4041070207 To: <sip:1000@172.16.44.8> Call-ID: 967655ECA0FEA994(a)192.168.178.1 CSeq: 2 REGISTER SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.42.150:5060;rport=5060;branch=z9hG4bKD33480AB9580F76B From: <sip:1000@172.16.44.8>;tag=4041070207 To: <sip:1000@172.16.44.8>;tag=48QmcmccUc50B Call-ID: 967655ECA0FEA994(a)192.168.178.1 CSeq: 2 REGISTER Date: Fri, 15 Jul 2011 10:39:07 GMT * UAC does REGISTER (with contact) when there are no bindings in the db yet: REGISTER sip:172.16.44.8 SIP/2.0 Via: SIP/2.0/UDP 172.16.42.150:5060;rport;branch=z9hG4bKB8AA233F4F985F7C From: <sip:1000@172.16.44.8>;tag=4041070207 To: <sip:1000@172.16.44.8> Call-ID: 967655ECA0FEA994(a)192.168.178.1 CSeq: 3 REGISTER Contact: <sip:1000@172.16.42.150;uniq=87E52E228E505C8B8957ADA870829> Expires: 1800 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.42.150:5060;rport=5060;branch=z9hG4bKB8AA233F4F985F7C From: <sip:1000@172.16.44.8>;tag=4041070207 To: <sip:1000@172.16.44.8>;tag=5HHDeFXFrNUKQ Call-ID: 967655ECA0FEA994(a)192.168.178.1 CSeq: 3 REGISTER Contact: <sip:1000@172.16.42.150;uniq=87E52E228E505C8B8957ADA870829>;expires=1800 Date: Fri, 15 Jul 2011 10:39:07 GMT * UAC does REGISTER (with contact) when there is already one binding in the db: REGISTER sip:172.16.44.8 SIP/2.0 Via: SIP/2.0/UDP 172.16.42.126:10818;branch=z9hG4bK-d8754z-a7869a477c0f317c-1---d8754z-;rport Contact: <sip:1000@172.16.42.126:10818;rinstance=f2aff35495d275bc> To: "1000"<sip:1000@172.16.44.8> From: "1000"<sip:1000@172.16.44.8>;tag=cf168522 Call-ID: MTAyZDJkMTUyN2ViYTE1NmFkNjFiNDc1ZDkwNWQwYTc. CSeq: 2 REGISTER Expires: 3600 SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.42.126:10818;branch=z9hG4bK-d8754z-a7869a477c0f317c-1---d8754z-;rport=10818 From: "1000"<sip:1000@172.16.44.8>;tag=cf168522 To: "1000" <sip:1000@172.16.44.8>;tag=ZrQtFUm2rpjtB Call-ID: MTAyZDJkMTUyN2ViYTE1NmFkNjFiNDc1ZDkwNWQwYTc. CSeq: 2 REGISTER Contact: <sip:1000@172.16.42.150;uniq=87E52E228E505C8B8957ADA870829>;expires=1787 Contact: <sip:1000@172.16.42.126:10818;rinstance=f2aff35495d275bc>;expires=3600 Date: Fri, 15 Jul 2011 11:08:10 GMT

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FreeSWITCH Conference Call Today
by Michael Collins 13 Jul '11

13 Jul '11

Hello all, Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_07_13 We have a few things to discuss as a community, plus a cool video to share. We are especially interested in your thoughts on the subject of FreeSWITCH "config sets" like what's here: http://svn.freeswitch.org/svn/configs/ Please be ready to share your config ideas with the group. Talk to you soon! -Michael

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mod_rtmp
by Stephen Dame 13 Jul '11

13 Jul '11

Great module. mod_rtmp , I almost have it working. I'm connecting to freeswitch thru mod_rtmp using the flex client. but unclear to how the username password works for authentication. Doesn't look like anyplace in rtmp.conf.xml or the .html, or .mxml to set? Using fs_cli - It nets connects fine. looks like digits are received, but not sure how to configure or use default username or password so it can login to make a call. Also question on rtmp://myserver/phone should the profile in rtmp.conf.xml match the rtmp name /phone? Any tips appreciated. Trying to get this working with BigBluebutton. Will document a tutorial when finished. Regards, Stephen

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mod_skypopen
by Debasish Chandra 13 Jul '11

13 Jul '11

Hi, I am trying to integrate FreeSWITCH with Skype. I have installed skype client using install.pl under mod_skypopen dir. The skype client is not able to connect when am running client using start_skype_clients.sh. However it connect when I open skype101 in skype-clients-symlinks-dir from desktop. I am using OS: Red Hat Enterprise Linux Server release 5.3 (Tikanga). Below is the start_skype_clients.sh script -------------------------------------------------------------------------------------------------------------------------------------------------------------------------- #!/bin/sh #Unload possible ALSA sound modules that would conflict with our OSS fake module rmmod snd_pcm_oss rmmod snd_mixer_oss rmmod snd_seq_oss sleep 1 #Create the inode our fake sound driver will use mknod /dev/dsp c 14 3 #Load our OSS fake module insmod /usr/local/freeswitch/skypopen/skypopen-sound-driver-dir/skypopen.ko #start the fake X server on the given port /usr/bin/Xvfb :101 -ac -nolisten tcp -screen 0 640x480x8 & sleep 3 # start a Skype client instance that will connect to the X server above, and will login to the Skype network using the 'username password' you send to it on stdin. su root -c "/bin/echo 'username passwd'| DISPLAY=:101 /usr/local/freeswitch/skypopen/skype-clients-symlinks-dir/skype101 --dbpath=/usr/local/freeswitch/skypopen/skype-clients-configuration-dir/skype101 --pipelogin &" sleep 7 exit 0 -------------------------------------------------------------------------------------------------------------------------------------------------------------------------- Can you please help to find out where I am doing wrong? Best Regards, Debasish

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ClueCon Assistance Needed: Sponsoring a VIP
by Michael Collins 12 Jul '11

12 Jul '11

Hello all! If you or your company is in a position to sponsor the travel of a VIP to ClueCon please contact me off list and I will give you the details. We've already had one group step up and sponsor Philip Zimmermann (thank you, Travis!) and we are looking for another one. Thanks, Michael

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windows build fails, on XP, Visual Studio Exprss 2008: libaprutil - 9 error(s)
by Giovanni Maruzzelli 12 Jul '11

12 Jul '11

All other code is compiled successfully ------ Build started: Project: libaprutil, Configuration: Debug Win32 ------ Performing Pre-Build Event... 0 File(s) copied Linking... Creating library Debug/libaprutil-1.lib and object Debug/libaprutil-1.exp apr_xml.obj : error LNK2019: unresolved external symbol _XML_SetCharacterDataHandler referenced in function _apr_xml_parser_create@4 apr_xml.obj : error LNK2019: unresolved external symbol _XML_SetElementHandler referenced in function _apr_xml_parser_create@4 apr_xml.obj : error LNK2019: unresolved external symbol _XML_SetUserData referenced in function _apr_xml_parser_create@4 apr_xml.obj : error LNK2019: unresolved external symbol _XML_ParserCreate referenced in function _apr_xml_parser_create@4 apr_xml.obj : error LNK2019: unresolved external symbol _XML_ParserFree referenced in function _cleanup_parser apr_xml.obj : error LNK2019: unresolved external symbol _XML_GetErrorCode referenced in function _do_parse apr_xml.obj : error LNK2019: unresolved external symbol _XML_Parse referenced in function _do_parse apr_xml.obj : error LNK2019: unresolved external symbol _XML_ErrorString referenced in function _apr_xml_parser_geterror@12 C:\freeswitch\Debug\libaprutil.dll : fatal error LNK1120: 8 unresolved externals Build log was saved at "file://c:\freeswitch\libs\win32\apr-util\Debug\BuildLog.htm" libaprutil - 9 error(s), 0 warning(s) ========== Build: 0 succeeded, 1 failed, 1 up-to-date, 0 skipped ========== -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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