- FreeSWITCH-users - FreeSWITCH Mailing Lists

outbound calling
by Boris Krivonog 25 Jul '08

25 Jul '08

Hi all! I'm new to FreeSwitch and have a newby question: what is the best/preferred way to create a new call from some external application? For example: I have an application A (i.e. stand alone Java application) which will invoke a RMI/web service or whatever service which should reside within FreeSwitch and will, when called, create a new call leg, play an audio message to this call leg, ... I'm currently browsing some samples on how to write a Java application that resides within FreeSwitch using mod_java. Is this the right way to do it? The problem is that I want to create a call leg from scratch and not to handle an incoming call and do something with it; all samples only process an incoming call... Thanks in advance, Boris

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ASR - TTS - MRCP
by Klaus Teller 25 Jul '08

25 Jul '08

Hi Folks, Great jobs. Congratulations. I have a question related to the ASR and TTS aspects of Freeswitch. I know Freeswitchsupport OpenMRCP. So, are these ASR and TTS additions somehow integrated through OpenMRCP? Klaus. -- GMX Kostenlose Spiele: Einfach online spielen und Spaß haben mit Pastry Passion! http://games.entertainment.gmx.net/de/entertainment/games/free/puzzle/61691…

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Dhcp and ip address changes
by Simon Capper 25 Jul '08

25 Jul '08

Is there a way to handle the IP address of the server freeswitch is on changing (due to dhcp address change) other than restarting it (or reloading the SIP profile). Simon

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best way todo speech recognition WAS mod_pocketsphinx
by Cesar Bermudez 24 Jul '08

24 Jul '08

What is the best option to get the speech recognition? what advices do you have, or samples, or application that you created using this? On Thu, Jul 24, 2008 at 6:21 PM, Cesar Bermudez <cesar.bermudez(a)gmail.com> wrote: > thx Brian, and sorry for the double post. > > > On Thu, Jul 24, 2008 at 6:18 PM, Brian West <brian(a)freeswitch.org> wrote: > >> In the src directory, Open up modules.conf uncomment asr_tts/ >> mod_pocketsphinx, save it. >> >> Type make mod_pocketsphinx-install >> >> done. >> >> /b >> >> >> On Jul 24, 2008, at 4:11 PM, Cesar Bermudez wrote: >> >> > hi all, >> > how is the way to build FreeSWITCH and enable mod_pocketsphinx? >> > thx. >> > _______________________________________________ >> > 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 >> >> Brian West >> sip:brian@freeswitch.org <sip%3Abrian(a)freeswitch.org> >> >> >> >> >> _______________________________________________ >> 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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Zaptel configuration error
by Hilda Farrad 24 Jul '08

24 Jul '08

We are trying to get Sangoma A200/A104 cards to run zaptel and we are receiving this error: ZT_SPANCONFIG failed on span 1: Invalid argument (22) after the zaptel configuration command (ztcfg -vv). Our zaptel.conf file looks as follows: # Autogenerated by /usr/local/sbin/sangoma/setup-sangoma -- do not hand edit # Zaptel Channels Configurations (zaptel.conf) # loadzone=us defaultzone=us #Sangoma A104 port 1 [slot:2 bus:3 span:1] <wanpipe1> span=1,0,0,ccs,hdb3,crc4 bchan=1-15,17-31 hardhdlc=16 #Sangoma A104 port 2 [slot:2 bus:3 span:2] <wanpipe2> span=2,0,0,ccs,hdb3,crc4 bchan=32-46,48-62 hardhdlc=47 #Sangoma A104 port 3 [slot:2 bus:3 span:3] <wanpipe3> span=3,0,0,ccs,hdb3,crc4 bchan=63-77,79-93 hardhdlc=78 #Sangoma A104 port 4 [slot:2 bus:3 span:4] <wanpipe4> span=4,0,0,ccs,hdb3,crc4 bchan=94-108,110-124 hardhdlc=109 #Sangoma A200 [slot:1 bus:3 span:5] <wanpipe5> fxoks=125 fxoks=126 fxsks=127 fxsks=128 We configured our files according to the documentation on the openzap wiki page for zaptel configuration and openzap.conf, openzap.conf.xml. We are using ubuntu and installed wanpipe3.3.9 and zaptel 1.4.11. We are pretty new to freeswitch so any help would be great. ~Hilda

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From domain
by Alex Gusak 24 Jul '08

24 Jul '08

We have sheme: +-----+ +----+ +-----+ | GW1 |-->| FS |-->| GW2 | +-----+ +----+ +-----+ <gateway name="GW1" ip="X.X.X.X"> <param name="caller-id-in-from" value="true"/> <param name="realm" value="domain.net"/> <param name="from-domain" value="domain.net"/> <param name="username" value="1"/> <param name="password" value="1"/> <param name="register" value="false"/> </gateway> <gateway name="GW2" ip="X.X.X.X"> <param name="caller-id-in-from" value="true"/> <param name="realm" value="domain.net"/> <param name="from-domain" value="domain.net"/> <param name="username" value="1"/> <param name="password" value="1"/> <param name="register" value="false"/> </gateway> When call from GW1 to GW2 FreeSWITCH recv INVITE from GW1: From: "Alex"<sip:alex_gusak@domain.net>;transport=UDP;tag=7552bb34 To: <sip:XXXXXX@domain.net>;transport=UDP and send INVITE to GW2: From: "Alex" <sip:alex_gusak@X.X.X.X>;tag=BK71NZ3yZvptK To: <sip:XXXXXX@domain.net> where X.X.X.X IP address FreeSWITCH. How to make that pass in the header From: the domain name domain.net? -- Alex Gusak

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Custom sip headers
by Simon Capper 24 Jul '08

24 Jul '08

Is there a way to add custom headers to the sip messages sent from mod_sofia? Simon

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lua, sound playback and interruption on DTMF entry
by Alex Kinch 24 Jul '08

24 Jul '08

Hi all, I've got a lua script that plays a sound file then listens for DTMF. What I'd like to do is interrupt the sound file playback and process the DTMF if the user starts entering digits whilst the sound file is playing. At the moment they have to wait til the sound file has finished before entering digits. Any suggestions? Here's the script snippet.. ession:streamFile("/usr/local/freeswitch/sounds/dk/gsm/vm-enter-num-to- call.gsm"); session:streamFile("/usr/local/freeswitch/sounds/dk/gsm/vm-then- pound.gsm"); digits = session:getDigits(20, "#", 3000); Thanks, Alex

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problems with dingaling
by Alex Gusak 24 Jul '08

24 Jul '08

We have some problems with dingaling. When call from SIP to GTALK, the connection is established, but clients do not hear each other. During connection comes message: <iq type="error" to="alex.gusak(a)gmail.com/gtalk37B4CA2D" id="304" from="Gennadiy.Lunkov(a)gmail.com/Talk.v105176AFA53"> <session type="transport-info" id="4768150905" initiator="alex.gusak(a)gmail.com/gtalk37B4CA2D" xmlns="http://www.google.com/session"> <transport xmlns="http://www.google.com/transport/p2p"> <info_element name="caller_id_number" value="alex_gusak" xmlns="http://www.freeswitch.org/jie"></info_element> <info_element name="caller_id_name" value="Alex" xmlns="http://www.freeswitch.org/jie"></info_element> <candidate name="rtp" address="X.X.X.X" port="17762" username="XXXXXXX" password="XXXXXXXX" preference="1.0" protocol="udp" type="local" network="0" generation="0"></candidate> </transport> </session> <error code="501" type="cancel"> <feature-not-implemented xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"></feature-not-implemented> </error> </iq> Jingle profile looks like this: <profile type="client"> <param name="name" value="gmail.com"/> <param name="login" value="alex.gusak(a)gmail.com/gtalk"/> <param name="password" value="XXXXXXXXXX"/> <param name="server" value="talk.google.com"/> <param name="context" value="from_google"/> <param name="dialplan" value="XML"/> <param name="message" value="FreeSwitch"/> <param name="rtp-ip" value="X.X.X.X"/> <param name="auto-login" value="true"/> <param name="auto-reply" value="Press"/> <param name="use-rtp-timer" value="true"/> <param name="sasl" value="plain"/> <param name="tls" value="true"/> <param name="vad" value="both"/> <param name="exten" value="100"/> </profile> Dialplan: <extension name="out"> <condition field="source" expression="mod_sofia"/> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="dingaling/gmail.com/gennadiy.lunkov@gmail.com"/> </condition> </extension> FreeSWITCH has public IP. Where could be the problem? -- Alex Gusak

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execute_on_answer problem
by Erol Akarsu 24 Jul '08

24 Jul '08

Hi , I have started using freeswitch and impressed with it, how easy to set up and run quick examples. It is awesome! What I do here is I bridge the call between IVR and a remote phone. Then remot eparty pick up the phone and, IVR plays a prompt. I want the remote user listen all prompt content. If we executed these 2 action inside my dialmap extension , playback action don't wait until remote end pick up the phone. Therefore, the user misses the beginning part of prompt. <action application="answer"/> <action application="playback" data="/home/eakarsu/kurumcell.g729"/> Now I started using LUA scripts. When I execute " originate sofia/gateway/voipwise/001804XXXYYYY(a)sip.voipwise.com 1238" from FS command line, I am getting error and calls dropped. I have attached log file where 'nua_r_cancel: unknown event 32: 200 Ok" debug statement is inserted. My dialmap extension and kurumcell.lua script is here. call is dropped. I apprecciate if you can guide me on this. <extension name="kurumcell_test"> <condition field="destination_number" expression="^1238$"> <action application="set" data="execute_on_answer=kurumcell.lua"/> </condition> </extension> kurumcell.lua: -- answer the call session:answer(); -- play a file session:streamFile("/home/eakarsu/kurumcell.g729"); -- hangup session:hangup(); 2008-07-23 20:52:25 [DEBUG] switch_core_state_machine.c:140 switch_core_standard_on_execute() sofia/external/0018042489925(a)sip.voipwise.com Execute set(execute_on_answer=kurumcell.lua) 2008-07-23 20:52:25 [DEBUG] mod_dptools.c:715 set_function() SET [execute_on_answer]=[kurumcell.lua] 2008-07-23 20:52:25 [NOTICE] switch_core_state_machine.c:157 switch_core_standard_on_execute() Hangup sofia/external/0018042489925(a)sip.voipwise.com [CS_EXECUTE] [NORMAL_CLEARING] 2008-07-23 20:52:25 [DEBUG] switch_channel.c:1360 switch_channel_perform_hangup() Kill sofia/external/0018042489925(a)sip.voipwise.com [KILL] 2008-07-23 20:52:25 [DEBUG] switch_core_session.c:676 switch_core_session_signal_state_change() Kill sofia/external/0018042489925(a)sip.voipwise.com [BREAK] 2008-07-23 20:52:25 [DEBUG] switch_core_state_machine.c:430 switch_core_session_run() (sofia/external/0018042489925(a)sip.voipwise.com) State EXECUTE going to sleep 2008-07-23 20:52:25 [DEBUG] switch_core_state_machine.c:365 switch_core_session_run() sofia/external/0018042489925(a)sip.voipwise.com Running State Change CS_HANGUP 2008-07-23 20:52:25 [DEBUG] switch_core_state_machine.c:393 switch_core_session_run() (sofia/external/0018042489925(a)sip.voipwise.com) State HANGUP 2008-07-23 20:52:25 [DEBUG] mod_sofia.c:264 sofia_on_hangup() Channel sofia/external/0018042489925(a)sip.voipwise.com hanging up, cause: NORMAL_CLEARING 2008-07-23 20:52:25 [DEBUG] mod_sofia.c:300 sofia_on_hangup() Sending CANCEL to sofia/external/0018042489925(a)sip.voipwise.com 2008-07-23 20:52:25 [DEBUG] switch_core_state_machine.c:46 switch_core_standard_on_hangup() Standard HANGUP sofia/external/0018042489925(a)sip.voipwise.com, cause: NORMAL_CLEARING 2008-07-23 20:52:25 [DEBUG] switch_core_state_machine.c:393 switch_core_session_run() (sofia/external/0018042489925(a)sip.voipwise.com) State HANGUP going to sleep 2008-07-23 20:52:25 [DEBUG] switch_core_session.c:735 switch_core_session_thread() Session 6 (sofia/external/0018042489925(a)sip.voipwise.com) Locked, Waiting on external entities 2008-07-23 20:52:25 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 6 (sofia/external/0018042489925(a)sip.voipwise.com) Ended 2008-07-23 20:52:25 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/external/0018042489925(a)sip.voipwise.com [CS_HANGUP] 2008-07-23 20:52:25 [DEBUG] sofia.c:319 sofia_event_callback() nua_r_cancel: unknown event 32: 200 Ok 2008-07-23 20:52:25 [DEBUG] sofia.c:190 sofia_event_callback() event [nua_r_invite] status [487][Request terminated] session: n/a 2008-07-23 20:52:25 [DEBUG] sofia.c:190 sofia_event_callback() event [nua_i_state] status [487][Request terminated] session: n/a

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