- FreeSWITCH-users - FreeSWITCH Mailing Lists

send fax in one leg call
by Madovsky 26 Feb '11

26 Feb '11

I'm trying to send a fax to a one leg call. for example a registered user wants to send a fax to another registered user so I have this in my default dialplan <extension name="internal_fax"> <condition field="${sip_to_uri}" expression="^(\+)(\d{10})@$${domain}$"> <action application="ring_ready"/> <action application="set" data="execute_on_media=tone_detect fax 1100 r +5000 transfer 'receivefax XML features' 2" /> <action application="sleep" data="2000"/> <action application="answer"/> <action application="voicemail" data="default ${domain_name} ${sip_to_user}"/> </condition> </extension> for now I have no other way to send the fax from CLI like this: fs_cli -x "uuid_broadcast <uuid> execute_extension::' sendfax XML features' <aleg=timeout??> <bleg=no bleg??>" thanks Franck

2 4

ESL socket outbound: clean way to close a session?
by Stephen Wilde 25 Feb '11

25 Feb '11

I have problem to clean close an outbound session. I'm using the "esl_oop.h/esl_oop.cpp" files and my application is developed in c++ and at startup I do a "linger" command. My event consuming loop is: while (run) { ESLevent * event = connection.recvEventTimed(sleeptime); ... event processing ... } This application handle an inbound call (legA), originate an outbound call (legB) and bridge the two calls. The problem is present when the legA hangup the call: sometimes is received the content-type="text/disconnect-notice" (without receiving "CHANNEL_HANGUP") and the "recvEventTimed" exits with event=NULL. In this case it's not possible to receive any event or send any other message to Freeswitch because the "connection.execute" fails. In this situation, if I have originated (with '& park') an outbound call, this call remain in parked state and the application cannot hangup it. What I'm doing wrong?

2 2

audio<->T38 gateway problem
by Wesley Akio 25 Feb '11

25 Feb '11

Hi All, I have a FreeSwitch box running FreeSWITCH version: 1.0.head (git-c6e7988 2010-12-09 14-25-39 -0600) under CentOS 5.5. I need freeswitch act as a gateway with G711 in one side an T38 in the other: Caller:G711--->FreeSwitch--->Callee:T38 I tried several variations of dialplans without success and was hoping someone could help me, the call stablishes in G711 and I could not find a way to switch to t38. My current dialplan follows: <condition field="destination_number" expression=".*" break="on-true"> <action application="set" data="fax_enable_t38=true"/> <action application="set" data="execute_on_answer=t38_gateway peer"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="progress_timeout=10"/> <action application="set" data="effective_caller_id_name=0000"/> <action application="set" data="effective_caller_id_number=0000"/> <action application="set" data="execute_on_answer=sched_hangup +9000 alloted_timeout"/> <action application="bridge" data="sofia/testfax/1234(a)1.2.3.4"/> </condition> Thank you, Wesley Akio

2 1

PlayAndGetDigits parameters
by Malay Thakershi 25 Feb '11

25 Feb '11

Recently I upgraded to latest GIT version and found there was an extra parameter to this function. I have two questions if someone can explain: --> 4th parameter * timeout = Number of milliseconds to wait once the file is done playing before you type a digit. >From the name, I alway thought this had to do with user's inactivity at the prompt. So if value is 3000, after 3 seconds of inactivity prompt will either do a retry or move on. But I think from wiki description, it seems # of milliseconds for which data entry will be blocked, am I correct? --> 10th parameter * digit_timeout = Number of milliseconds to wait after DTMF ( Added Here[1]<http://fisheye.freeswitch.org/changelog/freeswitch.git/?cs=cfa30468e89788d2…> ) What does this do? Is it # ms to wait before second digit can be pressed? I have following function in mod_managed code. pStrLivePreObInput = mObjMainSession.PlayAndGetDigits(5, 5, 3, 5000, "", pStrObGreetTextFile.Replace(".txt", ".wav"), @"C:\FreeSWITCH\sounds\en\us\chAsmt\static\error\invalid_number.wav", "^\\d", "", 3000); Does that mean user has to wait 5 seconds before a key entry can be accepted? And there has to be 3 seconds gap between each key? If so, I will need to redo the timing. Please let me know. Malay

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Second transfer
by Santiago Soares 25 Feb '11

25 Feb '11

Hello, I'm using att_xfer to make call transfer, like that: A calls B B calls C B hangup and A is bridged do C The thing is, the way I am trying to do it, C is able to transfer the call again. I don't want that C be able to transfer the call again. The call should be transfered only once. Is it possible? This is my dialplan: <include> <extension name="test_extension"> <condition field="destination_number" expression="^numberB$"> <action application="set" data="bind_meta_key=*"/> <action application="bind_meta_app" data="3 b s execute_extension::transferencia XML public"/> <action application="bridge" data="sofia/gateway/test/loginB"/> </condition> </extension> <extension name="transferencia"> <condition field="destination_number" expression="^transferencia$"> <action application="read" data="3 3 base256/16000/sound.wav attxfer_callthis 30000 #"/> <action application="execute_extension" data="transferencia2"/> </condition> </extension> <extension name="transferencia2"> <condition field="${attxfer_callthis}" expression="^numberC$"> <action application="att_xfer" data="sofia/gateway/test/loginC"/> </condition> <condition field="${attxfer_callthis}" expression="^numberD$"> <action application="att_xfer" data="sofia/gateway/test/loginD"/> </condition> </extension> </include> Thank you, Santiago Soares

3 7

Using 16 KHz sounds
by Malay Thakershi 25 Feb '11

25 Feb '11

Hello, I use Cepstral in my mod_managed FS application. I mainly use Session.Speak and PlayAndGetDigits in my code to play WAV / audio text. When I started using FS and got a stable program running, I used Cepstral Allison 8 KHz voice. But frequently I get choppy type of sound. Earlier it was acceptable but now some callers seem to have difficulty understanding the call audio. Would it help if I get 16 KHz sounds / Cepstral license? What are changes I would need to make? Thank you for any help. Malay Thakershi

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ESL in Delphi/Pascal
by Dave Bracken 25 Feb '11

25 Feb '11

Can anyone help me to figure out how to use the ESL in Delphi? We have tried compiling the files in VS2008, and it produces a .lib file and some .obj files, but none of which can be linked into delphi. I do not know hardly anything about C, so i am kinda of flying blind here. Any help would be very greatly appreciated. DelphiGuy

2 1

Receive order guarantee for bgapi command replies over ESL?
by Mauritz Løvgren 25 Feb '11

25 Feb '11

Hi, Are replies for bgapi commands sent back to an inbound socket connection in the same order as the commands were sent? Example: sending 10 bgapi messages from client to freeswitch socket with no delay inbetween. Will the response for those command messages be sent back by freeswitch in the exact same order as their commands were received, or should one wait for each reply before sending a new bgapi command to be sure one gets a reply for the correct command? If there was an option of providing a UUID (or sequenceId) for the command, it would be easier to match the reply with the request upon receival, but this doesn’t seem to be possible with the current implementation? Regards, Mauritz Lovgren Systems Architect IPLink Inc.

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one leg call and channel uuid
by Madovsky 25 Feb '11

25 Feb '11

when there is one leg call, if I want to use uuid_broadcast or uuid_transfer, should I use the uuid of leg A ? Thanks

2 2

Re: [Freeswitch-users] Message Playback Stops
by Michael De Lorenzo 24 Feb '11

24 Feb '11

Is there some place where I can get more detailed log information on this? I thought that streamFile was a blocking operation (the next statement wouldn't be executed until playback was completed), is that not the case? Is there a difference in terms of performance with Freeswitch for Lua versus SpiderMonkey? Am I misreading the log? > > > *2011-02-21 17:23:09.077014 [NOTICE] switch_cpp.cpp:1181 Playing file: > /usr/local/freeswitch/* > *recordings/messages/16c0f890_**c35e33c0_777973.wav* > 2011-02-21 17:23:09.077014 [DEBUG] switch_ivr_play_say.c:1186 Codec > Activated L16@8000hz 1 channels 20ms > 2011-02-21 17:23:19.417631 [DEBUG] switch_ivr_play_say.c:1515 done playing > file > 2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 !!!!! Finished > playing the file !!!!! > 2011-02-21 17:23:19.417631 [INFO] switch_cpp.cpp:1181 Was VM detected? no > 2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 Played the message > at least once and checked for VM, we should be exiting the loop > > The messages like "finished playing" are log commands that are only > available in the script after it's answered. > > > > > On Wed, Feb 23, 2011 at 11:48 AM, Anthony Minessale < > anthony.minessale(a)gmail.com> wrote: > >> Your logs also don't show anything else like the call being answered >> or hungup or any other information. >> >> you probably need to add {ignore_early_media=true} to your dial string >> so the message does not play while the call is not answered. >> >> if you are calling sofia/internal/foo(a)bar.com change it to >> {ignore_early_media=true}sofia/internal/foo(a)bar.com >> >> >> >> >> On Wed, Feb 23, 2011 at 10:39 AM, Michael De Lorenzo >> <delorenzodesign(a)gmail.com> wrote: >> > Maybe failure isn't the correct word. Here's what happens: >> > >> > Call gets placed, recipient phone rings >> > Call answered >> > Check for VM >> > If not VM, play the message back after short delay <-- the message >> doesn't >> > play, the call hangs up (step 5), but the logs indicate that the message >> > played >> > Hangup >> > >> > As the log shows, it doesn't indicate that the message playback fails. >> > >> > On Wed, Feb 23, 2011 at 11:14 AM, Anthony Minessale >> > <anthony.minessale(a)gmail.com> wrote: >> >> >> >> I am asking for description of what happens when it fails, How do you >> >> know it's failing? >> >> If it fails it would be logging errors. >> >> >> >> >> >> >> >> >> >> On Wed, Feb 23, 2011 at 12:00 AM, Michael De Lorenzo >> >> <delorenzodesign(a)gmail.com> wrote: >> >> > I'm not sure where to check, but there's entries like this in >> >> > freeswitch.log.2011-02-21-23-40-11.1 >> >> > >> >> > 2011-02-21 17:22:53.157390 [NOTICE] mod_logfile.c:158 New log >> started. >> >> > 2011-02-21 17:22:53.157390 [NOTICE] switch_cpp.cpp:1181 Played the >> >> > message >> >> > at least once and checked for VM, we should be exiting the loop. >> >> > 2011-02-21 17:22:53.157390 [NOTICE] switch_cpp.cpp:1181 ************* >> >> > SETTING SESSION VARIABLES *************** >> >> > 2011-02-21 17:22:53.157390 [NOTICE] switch_cpp.cpp:1181 Session >> >> > Variable[profile_id]: 1 >> >> > 2011-02-21 17:22:53.157390 [NOTICE] switch_cpp.cpp:1181 ************* >> >> > DONE >> >> > SETTING SESSION VARIABLES *********** >> >> > 2011-02-21 17:22:53.377395 [NOTICE] switch_cpp.cpp:1181 Playing file: >> >> > >> /usr/local/freeswitch/recordings/messages/16c0f890_c35e33c0_777973.wav >> >> > 2011-02-21 17:22:53.377395 [DEBUG] switch_ivr_play_say.c:1186 Codec >> >> > Activated L16@8000hz 1 channels 20ms >> >> > 2011-02-21 17:23:03.717511 [DEBUG] switch_ivr_play_say.c:1515 done >> >> > playing >> >> > file >> >> > 2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 !!!!! >> Finished >> >> > playing the file !!!!! >> >> > 2011-02-21 17:23:03.717511 [INFO] switch_cpp.cpp:1181 Was VM >> detected? >> >> > no >> >> > 2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 Played the >> >> > message >> >> > at least once and checked for VM, we should be exiting the loop. >> >> > 2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 ************* >> >> > SETTING SESSION VARIABLES *************** >> >> > 2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 Session >> >> > Variable[profile_id]: 1 >> >> > 2011-02-21 17:23:03.717511 [NOTICE] switch_cpp.cpp:1181 ************* >> >> > DONE >> >> > SETTING SESSION VARIABLES *********** >> >> > 2011-02-21 17:23:03.937661 [NOTICE] switch_cpp.cpp:1181 Playing file: >> >> > >> /usr/local/freeswitch/recordings/messages/16c0f890_c35e33c0_777973.wav >> >> > 2011-02-21 17:23:03.937661 [DEBUG] switch_ivr_play_say.c:1186 Codec >> >> > Activated L16@8000hz 1 channels 20ms >> >> > 2011-02-21 17:23:08.237260 [DEBUG] switch_ivr_play_say.c:1515 done >> >> > playing >> >> > file >> >> > 2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 !!!!! >> Finished >> >> > playing the file !!!!! >> >> > 2011-02-21 17:23:08.237260 [INFO] switch_cpp.cpp:1181 Was VM >> detected? >> >> > no >> >> > 2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 Played the >> >> > message >> >> > at least once and checked for VM, we should be exiting the loop. >> >> > 2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 ************* >> >> > SETTING SESSION VARIABLES *************** >> >> > 2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 Session >> >> > Variable[profile_id]: 1 >> >> > 2011-02-21 17:23:08.237260 [NOTICE] switch_cpp.cpp:1181 ************* >> >> > DONE >> >> > SETTING SESSION VARIABLES *********** >> >> > 2011-02-21 17:23:08.437885 [NOTICE] switch_cpp.cpp:1181 Playing file: >> >> > >> /usr/local/freeswitch/recordings/messages/16c0f890_c35e33c0_777973.wav >> >> > 2011-02-21 17:23:08.437885 [DEBUG] switch_ivr_play_say.c:1186 Codec >> >> > Activated L16@8000hz 1 channels 20ms >> >> > 2011-02-21 17:23:08.857270 [DEBUG] switch_ivr_play_say.c:1515 done >> >> > playing >> >> > file >> >> > 2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 !!!!! >> Finished >> >> > playing the file !!!!! >> >> > 2011-02-21 17:23:08.857270 [INFO] switch_cpp.cpp:1181 Was VM >> detected? >> >> > no >> >> > 2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 Played the >> >> > message >> >> > at least once and checked for VM, we should be exiting the loop. >> >> > 2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 ************* >> >> > SETTING SESSION VARIABLES *************** >> >> > 2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 Session >> >> > Variable[profile_id]: 1 >> >> > 2011-02-21 17:23:08.857270 [NOTICE] switch_cpp.cpp:1181 ************* >> >> > DONE >> >> > SETTING SESSION VARIABLES *********** >> >> > 2011-02-21 17:23:09.077014 [NOTICE] switch_cpp.cpp:1181 Playing file: >> >> > >> /usr/local/freeswitch/recordings/messages/16c0f890_c35e33c0_777973.wav >> >> > 2011-02-21 17:23:09.077014 [DEBUG] switch_ivr_play_say.c:1186 Codec >> >> > Activated L16@8000hz 1 channels 20ms >> >> > 2011-02-21 17:23:19.417631 [DEBUG] switch_ivr_play_say.c:1515 done >> >> > playing >> >> > file >> >> > 2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 !!!!! >> Finished >> >> > playing the file !!!!! >> >> > 2011-02-21 17:23:19.417631 [INFO] switch_cpp.cpp:1181 Was VM >> detected? >> >> > no >> >> > 2011-02-21 17:23:19.417631 [NOTICE] switch_cpp.cpp:1181 Played the >> >> > message >> >> > at least once and checked for VM, we should be exiting the loop. >> >> > >> >> > I also have the Freeswitch log from the date too, but I'm not sure >> what >> >> > I >> >> > should post here that would be useful. >> >> > >> >> > >> >> > >> >> > On Tue, Feb 22, 2011 at 8:20 PM, Anthony Minessale >> >> > <anthony.minessale(a)gmail.com> wrote: >> >> >> >> >> >> do you have any console logs of it failing or any error messages >> etc? >> >> >> >> >> >> >> >> >> On Tue, Feb 22, 2011 at 1:22 AM, Michael De Lorenzo >> >> >> <delorenzodesign(a)gmail.com> wrote: >> >> >> > Hello, >> >> >> > >> >> >> > I have a Freeswitch installation that is intended to make many >> calls >> >> >> > (thousands) and playback a single wav file. The calls are >> >> >> > successfully >> >> >> > processed (the recipient's phone rings), but the call almost >> >> >> > immediately >> >> >> > disconnects, after about 1s. Everything seems to work fine if I'm >> >> >> > only >> >> >> > pushing one or two calls through the Freeswitch instance, but as >> soon >> >> >> > as >> >> >> > I >> >> >> > turn up the call rate (I'm still only doing about 50 concurrent >> >> >> > sessions) >> >> >> > the playback begins to fail. >> >> >> > >> >> >> > I've watched the calls go out from the console and nothing looks >> out >> >> >> > of >> >> >> > the >> >> >> > ordinary, except that the calls are disconnected with NORMAL >> CLEARING >> >> >> > prior >> >> >> > to completion. >> >> >> > >> >> >> > Here's the Lua script I'm using... >> >> >> > >> >> >> > profile_id = argv[1]; >> >> >> > account_code = argv[2]; >> >> >> > client_id = argv[3]; >> >> >> > caller_id_name = argv[4]; >> >> >> > caller_id = argv[5]; >> >> >> > dial_id = argv[6]; >> >> >> > number_to_call = argv[7]; >> >> >> > message_to_play = argv[8]; >> >> >> > max_retries = argv[9]; >> >> >> > retry_interval = argv[10]; >> >> >> > >> >> >> > local human_detected = false >> >> >> > local voicemail_detected = false; >> >> >> > local message_played = false; >> >> >> > >> >> >> > recordings_directory = >> "/usr/local/freeswitch/recordings/messages/"; >> >> >> > >> >> >> > function setDialVariables(set_as_session_variables) >> >> >> > local s = "profile_id=" .. profile_id; >> >> >> > s = s .. ",account_code=" .. account_code; >> >> >> > s = s .. ",client_id=" .. client_id; >> >> >> > s = s .. ",caller_id_name=" .. caller_id_name; >> >> >> > s = s .. ",caller_id=" .. caller_id; >> >> >> > s = s .. ",dial_id=" .. dial_id; >> >> >> > s = s .. ",number_to_call=" .. number_to_call; >> >> >> > s = s .. ",message_to_play=" .. message_to_play; >> >> >> > >> >> >> > freeswitch.consoleLog("notice", s .. "\n"); >> >> >> > >> >> >> > return s >> >> >> > end >> >> >> > >> >> >> > function printSessionVariables() >> >> >> > freeswitch.consoleLog("notice", "******* PRINTING SESSION >> >> >> > VARIABLES >> >> >> > **********\n"); >> >> >> > -- ommitted >> >> >> > freeswitch.consoleLog("notice", >> >> >> > "**********************************************\n"); >> >> >> > end >> >> >> > >> >> >> > function onInput(s, type, obj, arg) >> >> >> > if(type == "event" and voicemail_detected == false) then >> >> >> > freeswitch.consoleLog("debug","************ VOICE >> >> >> > MAIL/ANSWERING >> >> >> > MACHINE DETECTED *************\n"); >> >> >> > voicemail_detected = true; >> >> >> > return "break"; >> >> >> > end >> >> >> > return true; >> >> >> > end >> >> >> > >> >> >> > function playbackMessage(sleepTime) >> >> >> > message_played = false; >> >> >> > session:sleep(sleepTime); >> >> >> > -- play a file >> >> >> > message_file = recordings_directory .. message_to_play; >> >> >> > freeswitch.consoleLog("notice", "Playing file: " .. >> message_file >> >> >> > .. >> >> >> > "\n"); >> >> >> > session:streamFile(message_file); >> >> >> > freeswitch.consoleLog("notice", "!!!!! Finished playing the >> file >> >> >> > !!!!!\n"); >> >> >> > message_played = true; >> >> >> > end >> >> >> > >> >> >> > session = freeswitch.Session("{" .. setDialVariables(false) .. >> >> >> > ",ignore_early_media=true,origination_caller_id_name=" .. >> >> >> > caller_id_name >> >> >> > .. >> >> >> > ",origination_caller_id_number=+1" .. caller_id .. >> >> >> > "}sofia/gateway/gateway_" >> >> >> > .. profile_id .. "/" .. number_to_call); >> >> >> > >> >> >> > while(session:ready()) do >> >> >> > setDialVariables(true) >> >> >> > session:answer(); >> >> >> > >> >> >> > -- session:execute("continue_on_fail","true"); >> >> >> > session:setInputCallback("onInput","true"); >> >> >> > session:execute("avmd","start"); >> >> >> > >> >> >> > playbackMessage(200); >> >> >> > >> >> >> > vm_status = voicemail_detected == true and "yes" or "no" >> >> >> > freeswitch.consoleLog("info", "Was VM detected? " .. vm_status >> .. >> >> >> > "\n"); >> >> >> > if(voicemail_detected) then >> >> >> > return "break"; >> >> >> > end >> >> >> > >> >> >> > freeswitch.consoleLog("notice", "Played the message at least >> once >> >> >> > and >> >> >> > checked for VM, we should be exiting the loop.\n") >> >> >> > end >> >> >> > >> >> >> > if (voicemail_detected) then >> >> >> > freeswitch.consoleLog("info", "Playback for voicemail.\n"); >> >> >> > session:execute("avmd","stop"); >> >> >> > playbackMessage(5000); >> >> >> > end >> >> >> > >> >> >> > freeswitch.consoleLog("info", "All finished, hanging up the >> >> >> > session.\n"); >> >> >> > session:hangup(); >> >> >> > >> >> >> > Any help would be greatly appreciated. >> >> >> > >> >> >> > Thank you, >> >> >> > >> >> >> > Michael >> >> >> > >> >> >> > _______________________________________________ >> >> >> > 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 >> >> >> > >> >> >> > >> >> >> >> >> >> >> >> >> >> >> >> -- >> >> >> 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 >> >> > >> >> > >> >> >> >> >> >> >> >> -- >> >> 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 >> > >> > >> >> >> >> -- >> 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 >> > >

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