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pocketsphinx questions
by Seven Du 16 Jul '10

16 Jul '10

Hi, I'm trying pocketsphinx, here are some questions: I'm almost on git head. I followed http://wiki.freeswitch.org/wiki/Mod_pocketsphinx , and can run the pizza test. The recognize not as accurate as expected when I'm using zoiper. Portaudio sounds better.(my background is a little noisy). I also read http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts , lua example, but cannot make it work. 1) Cannot find Makefile when make in grammer/ dir. Goole found it did exists but for some reason deleted, why? and no mklm. 2) the lua example doesn't need a grammer ? It only have a .dic and .lm 3) I didn't see Speech-Type related events on event socket, found a var asr_fire_events in source, but the js example has nothing with this var. Do we need this? Thanks. -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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.wav/.mp3 recording problems - lag
by Bogdan Patrascoiu 16 Jul '10

16 Jul '10

Hello, I recently set up a FS system for one call-center. One of the request was to enable call recording for inbound calls (it's just an inbound callcenter (20 lines) no outbound calls). At first i went with .mp3 enconding but i soon found out that in a "longer" conversation 8-9 minutes the two channels would get very much lagged.T The caller's, leg A, questions would be far ahead of the callees', leg B, answers. I though that this was caused by the .mp3 encoding so i switched back to plain old .wav. The problem is that the lag is still there and comes to "life" totally random. Has anyone ever encountered this before ? Thanks, Bogdan

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suppress ORIGINATOR_CANCEL errors
by Tim St. Pierre 16 Jul '10

16 Jul '10

Hello, Whenever calls are placed from a SIP endpoint, and that user cancels the call before it's answered, I get an error on the console: [ERR] switch_ivr_originate.c:2529 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] This error is correct, the call failed because someone bailed on it, but I don't really need to know this every time, at least I don't think it justifies logging an error. Is there a way to tell FS not to view an originator_cancel as an error, but to consider it "normal"? Any suggestions appreciated. Thanks! -Tim

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No Answer to Hold Request After Simultaneous-Ring Call Answered
by Jerry Richards 15 Jul '10

15 Jul '10

I have a configuration that simultaneously rings multiple extensions using the following XML statement: <action application="bridge" data="user/${dialed_extension}@${domain_name},sofia/internal/1071%${domain_n ame},sofia/internal/1072%${domain_name}"/> However, after a phone other than the last phone (in the list) answers an inbound call, it cannot successfully hold the call. The INVITE only receives 100 Trying from FS and nothing more. See Line #2041 of the following post: http://pastebin.freeswitch.org/13470 The surrounding XML configuration tags are here: http://pastebin.freeswitch.org/13468 Thanks and Best Regards, Jerry

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no dtmf call back during file_string conjunction
by xuyan yang 15 Jul '10

15 Jul '10

hi all, I write a small IVR in javascript to react on user's 1 key input session.streamFile("a.wav!b.wav",dtmf_handle); when user press a key at the end of a.wav and the beginning of b.wav, there is no dtmf call back. if another key is pressed after that, the dtmf_handle is call with a event containing 2 digits. Is this a bug?

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FreeTDM Dialplan help w/ TDM410p
by Joseph L. Casale 15 Jul '10

15 Jul '10

I was using <condition field="source" expression="mod_openzap"> to catch my incoming zap fxo line, but now w/ FreeTDM changing mod_openzap to mod_freetdm as I expected no longer causes a match. Outgoing calls work, but looking at the log while an incoming call is made, it appears fs isn't even seeing it, nothing appears in the log. Any idea how to troubleshoot this? Thanks, jlc freetdm.conf [span zt FreeTDM] number => 1 fxo-channel => 1:1 freetdm.conf.xml <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="1"/> </settings> <analog_spans> <span id="1"> <param name="tonegroup" value="us"/> <param name="digit-timeout" value="2000"/> <param name="max-digits" value="11"/> <param name="dialplan" value="XML"/> <param name="context" value="public"/> <param name="enable-callerid" value="true"/> </span> </analog_spans> </configuration> conf/dialplan/public/01_freetdm_in.xml <include> <extension name="freetdm_in"> <condition field="source" expression="mod_freetdm"> <action application="set" data="domain_name=$${domain}"/> <action application="set" data="transfer_ringback=$${us-ring}"/> <action application="answer"/> <action application="transfer" data="2000 XML default"/> </condition> </extension> </include>

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parallel execution in dialplan
by Achim Stamm 15 Jul '10

15 Jul '10

Hello! I have following problem: I want that 2 applications are executed parallel from my dialplan. Actually the dialplan is executed step by step, that means that application "bridge" is started and phone 1011 is ringing. But application "test" will not start at the same time. This is a snipit from my dialplan: ----------------------------------- <extension name="TEST-Anrufer"> <condition field="destination_number" expression="70"> <action application="info"/> <action application="bridge" data="user/1011(a)192.168.1.30"/> <action application="test" data="EProcessCall,EXT50,70"/> </condition> </extension> ---------------------------------- Is there a solution that both applications are executed parallel and start at same time ? regards Achim Stamm -- Achim Stamm, Dipl.-Inform. (FH) Lyncker & Theis GmbH Wilhelmstr. 16 65185 Wiesbaden Germany Fon +49 611/9006951 Fax +49 611/9406125 Handelsregister: HRB 23156 Amtsgericht Wiesbaden Steuernummer: 04323897052 USt-IdNr.: DE255806399 Geschäftsführer: Filip Lyncker, Armin Theis

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Display update support on Cisco SPA 509g?
by Steven Ward 15 Jul '10

15 Jul '10

Hello list, Does anyone know if the Cisco SPA 509g supports display updates? Is there any good way to check this? I run a uuid_display <uuid> <display> on a channel that the SPA is on, but I'm not even seeing freeswitch send an UPDATE to the phone. This works on my Aastras and Polycoms, so I'm wondering if there is any way to get display updates on the SPA, or if it's simply unsupported by the phone. Thanks! - Steve

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Re: [Freeswitch-users] Variable in Bridge string
by Anthony Minessale 15 Jul '10

15 Jul '10

Its a security feature to prevent injection. I think I can get it to work, try latest GIT otherwise create an new object in your lua script and run distributor in the script. then build the string. or use session:execute("set", "var=val") On Thu, Jul 15, 2010 at 7:20 AM, Ahmed Ajmal <ahmed.ajmal(a)breezecom.ae>wrote: > Hi > > > > I have a strange problem creating a bridge string from LUA. The LUA script > returns a bridge string which looks like this: > > > > sofia/gateway/${distributor(my_dist)}/1773XXXXXXX > > > > I pass this bridge string to the bridge app in the dialplan: > > <action application="bridge" data="${bridge_string}"/> > > > > On executing the dialplan I get the following error: > > *Invalid data* (${bridge_string} *contains a variable* > > * * > > *I have tried executing the bridge app from within the lua script it works > but I would prefer to have the dialplan process it. I am on Freeswitch > 1.0.6.* > > * * > > *Thanks* > > *Ahmed* > > > > > > > > > > _______________________________________________ > 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 <MSN%3Aanthony_minessale(a)hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale(a)gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888(a)conference.freeswitch.org> pstn:+19193869900

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Variable in Bridge string
by Ahmed Ajmal 15 Jul '10

15 Jul '10

Hi I have a strange problem creating a bridge string from LUA. The LUA script returns a bridge string which looks like this: sofia/gateway/${distributor(my_dist)}/1773XXXXXXX I pass this bridge string to the bridge app in the dialplan: <action application="bridge" data="${bridge_string}"/> On executing the dialplan I get the following error: Invalid data (${bridge_string} contains a variable I have tried executing the bridge app from within the lua script it works but I would prefer to have the dialplan process it. I am on Freeswitch 1.0.6. Thanks Ahmed

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