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Module_Exists for SpiderMonkey and PocketSpinx failing the dialplan at Execution state
by ankIT WALiA 06 Jul '11

06 Jul '11

Hi all, I tried to execute the pizza example for PocketSphinx for speech recognition. I added the grammar,enabled mod_pocketsphinx and spidermonkey, added sound files and js files as per the wiki. http://wiki.freeswitch.org/wiki/Mod_pocketsphinx#Install_.26_Configure When, I call to the extension, though it passes the Regex, showing true for the conditions. But could not execute the javascript application and subsequently the js file. The dialplan looks like * <extension name="pizza_demo"> <condition field="destination_number" expression="^(74992)$"/> <condition field="${module_exists(mod_spidermonkey)}" expression="true"/> <condition field="${module_exists(mod_pocketsphinx)}" expression="true"> <action application="javascript" data="ps_pizza.js"/> </condition> </extension>* If I remove the condition of module exist, the js file gets executed and the call goes on. But, It does not work with Module exists condition. I have pasted the logs http://pastebin.freeswitch.org/16680. I could not even find anything concrete on the logs. Thanks Ankit -- Life is like a rose its upto u feel it as its fragrance or thorns

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FSClient FreesSWITCH Softphone Updated
by Mitch Capper 06 Jul '11

06 Jul '11

A new binary build of FSClient is out it has several changes from the original build published and should fix any language issues people had with the original installer. It also adds support for call recording, TLS, and a few other improvements. In addition it has been documented in the wiki at: http://wiki.freeswitch.com/wiki/FSClient with features, usage, information to compile it yourself, developer/plugin information etc. Binary installer can be found at: http://files.freeswitch.org/windows/installer/x86/FSClient.zip it will upgrade your settings automatically and you need not uninstall the old version. It also includes a patched fs_cli per patch http://jira.freeswitch.org/browse/FS-3188 which can be used with fs_logger.pl (freeswitch log collector more information can be found at: http://fisheye.freeswitch.org/browse/~raw,r=HEAD/freeswitch-contrib/mitchca… or the perl app itself: http://fisheye.freeswitch.org/browse/~raw,r=HEAD/freeswitch-contrib/mitchca… or a compiled binary at http://mitchcapper.com/fs_logger.exe ). Feedback on both always welcome! ~Mitch

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curl.run data field post file
by Daniel Knoll 06 Jul '11

06 Jul '11

Hi Group, I figured the right syntax of curl.run out. and now I have a question. How can I post the Content of a File into the "data" Field of curl.run() If i set the Option to "@/tmp/filename.flac" or "filename=(a)/tmp/filename.flac" the Content-Length is allways the number of the Characters in the "data" Field. my code line is: curl.run("POST", "http://www.google.com/speech-api/v1/recognize?xjerr=1&client=chromium&lang=…", "filename=(a)/tmp/90.flac", my_callback, "my arg\n","","","audio/x-flac; rate=16000"); The sniffing of HTTP Traffic look like this: interface: eth0 (xxx.xxx.xxx.xxx/255.255.255.192) filter: (ip or ip6) and ( port 80 ) #### T xxx.xxx.xxx.xxx:45159 -> 209.85.149.105:80 [AP] POST /speech-api/v1/recognize?xjerr=1&client=chromium&lang=de-DEc HTTP/1.1..User-Agent: freeswitch-spidermonkey-curl/1.0..Host: www.google.com..Accept: */*..Content-Type: aud io/x-flac; rate=16000..Content-Length: 22....filename=(a)/tmp/90.flac ## T 209.85.149.105:80 -> xxx.xxx.xxx.xxx:45159 [AP] HTTP/1.1 200 OK..Content-Type: application/json; charset=utf-8..Content-Disposition: at tachment..Date: Wed, 06 Jul 2011 11:23:35 GMT..Expires: Wed, 06 Jul 2011 11:23:35 GMT.. Cache-Control: private, max-age=0..X-Content-Type-Options: nosniff..X-Frame-Options: SA MEORIGIN..X-XSS-Protection: 1; mode=block..Server: GSE..Transfer-Encoding: chunked....4 7..{"status":5,"id":"ec8d60cb80800296297afe166dee826c-1","hypotheses":[]}...0.... ####exit 11 received, 0 dropped any ideas? Thanks for helping me. Daniel

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Trouble with bind_digit_action and exec:execute_extension
by Michael Collins 06 Jul '11

06 Jul '11

Hello, I know that some have recently reported having trouble with bind_digit_action and exec:execute_extension. I have not been able to reproduce any of the reported symptoms. I've produced a simple test dialplan that you can drop into conf/dialplan/default/ and just reloadxml for a quick test. I've attached it and also pasted it here for reference. If you've been having BDA trouble please try this out and report back. Be sure to use pastebin.freeswitch.org for reporting your logs. -MC Dialplan: <extension name="BDA Test"> <condition field="destination_number" expression="^77(\d\d\d\d)$"> <action application="answer"/> <action application="bind_digit_action" data="test1,*1,exec:execute_extension,BDA_TARGET_1 XML default"/> <action application="bind_digit_action" data="test1,*2,exec:execute_extension,BDA_TARGET_2 XML default"/> <action application="bind_digit_action" data="test1,*3,exec:execute_extension,BDA_TARGET_3 XML default"/> <action application="bind_digit_action" data="test1,*4,exec:execute_extension,BDA_TARGET_4 XML default"/> <action application="bind_digit_action" data="test1,*5,exec:execute_extension,BDA_TARGET_5 XML default"/> <action application="bind_digit_action" data="test1,*6,exec:execute_extension,BDA_TARGET_6 XML default"/> <action application="digit_action_set_realm" data="test1"/> <action application="playback" data="${hold_music}"/> </condition> </extension> <extension name="BDA Target"> <condition field="destination_number" expression="^BDA_TARGET_(\d)$"> <action application="say" data="en number pronounced $1"/> </condition> </extension> Just dial 77xxxx and then press *1, *2, etc. You will hear the number voiced, i.e. *1 will say "one", *2 will say "two" and so forth.

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Calling an extension on a remote freeswitch box
by Yungwei Chen 06 Jul '11

06 Jul '11

Hi, I'm trying to place a call to an extension on a remote freeswitch box. I tried the following command, but it doesn't work. originate {ignore_early_media=true}sofia/internal/77777(a)192.168.1.20 test Here's the error message. 2011-07-05 18:06:46.670942 [ERR] sofia_reg.c:1912 Cannot locate any authentication credentials to complete an authentication request for realm '"192.168.1.18"' Am I using the right dial string? If so, how can I fix the error? Thanks.

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GTalk mod_dingaling read and send text messages
by Sebastien Peterson 05 Jul '11

05 Jul '11

Hi freeswitch-users, I am a new FreeSwitch user. I am trying to send and read Gtalk text messages from the dialplan. Is that something I can do and how? Could not find examples from the wiki. Once I read the message, I need to place it into a variable. In advance thanks ! Seb

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NAT Traversal
by Carmelo Fischetti - Digi Unit 05 Jul '11

05 Jul '11

Hi all, I'm experiencing a NAT traversal problem connecting some softphones to a FreeSWITCH server. At office the softphone registers with TLS-NAT, while at home it registers with TLS and actually it's unreacheable (it can place calls but it cannot receive them), even if the registration parameters (sofia status profile internal reg) contain both the public (IP: x.y.z.w) and the private (Contact: "user" <sip:1010@192.168.1.100:6010;transport=TLS;line=3c4e8d673906c5d>) IP addresses. In both cases the softphone is behind a router, the server is behind a firewall and runs the ddclient service (there is a DynDNS name assigned to it). Am I missing some configuration detail at server side? At the moment I've modified the default configuration files in the following way. The following lines in vars.xml: <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=stun:stun.freeswitch.org"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=stun:stun.freeswitch.org"/> were substituted with: <X-PRE-PROCESS cmd="set" data="domain=myhost.dyndns-server.com"/> <X-PRE-PROCESS cmd="set" data="external_rtp_ip=host:myhost.dyndns-server.com"/> <X-PRE-PROCESS cmd="set" data="external_sip_ip=host:myhost.dyndns-server.com"/> The following lines in sip_profiles/internal.xml: <param name="ext-rtp-ip" value="auto-nat"/> <param name="ext-sip-ip" value="auto-nat"/> were substituted with: <param name="ext-rtp-ip" value="host:myhost.dyndns-server.com"/> <param name="ext-sip-ip" value="host:myhost.dyndns-server.com"/> and I've also added the line: <param name="sip-force-contact" value="NDLB-connectile-dysfunction"/> but I actually don't understand what it means (I've found it in some posts about NAT traversal in FreeSWITCH). Thanks in advance. Carmelo

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pl HELP! how to change conference's caller-id-number at run-time?
by ran zhang 05 Jul '11

05 Jul '11

Hi All: I want when the bridging conference is inviting someone to start the conference, the caller is shown as the real caller id number rather than whats stored in 'caller-id-number' in conference.conf.xml. I have tried to change the 'effective_caller_id_number', it only affects when its a bridging call.

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originate a call to an extension and then connect to another extension
by Yungwei Chen 05 Jul '11

05 Jul '11

Hi, I am wondering if it's possible to originate a call to an extension and then connect to another extension. If so, how does the the dialstring look like? Thanks.

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Outbound socket, bridge command - how to specify leg_timeout?
by Mateusz Bartczak 05 Jul '11

05 Jul '11

Hi all I use outbound socket to handle call, I'm bridging call to external voip provider. This works very nice but I also need to specify call timeout (once answered). I've tried following but with no success: sendmsg call-command: execute execute-app-name: bridge execute-app-arg: {ignore_early_media=true,leg_timeout=10}sofia/gateway/trunk/600 event-lock: true sendmsg call-command: execute execute-app-name: bridge execute-app-arg: {ignore_early_media=true}[leg_timeout=10]sofia/gateway/trunk/600 event-lock: true sendmsg call-command: execute execute-app-name: bridge execute-app-arg: [leg_timeout=10]sofia/gateway/trunk/600 event-lock: true Leg_timeout seems to be ignored and call continues after 10 seconds. What's the correct form of setting it for calls bridged via outbound socket handler? Is the leg_timeout a good choice to limit maximum answered call length?

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