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Re: [Freeswitch-users] Calling web services, JMS or EJB from freeswitch
by Erol Akarsu 29 Jul '08

29 Jul '08

Mike, Can you elaborate some and give some pointers to examples? Where has you used json/rest type calls? Thanks ----- Original Message ---- From: Michael Jerris <mike(a)jerris.com> To: freeswitch-users(a)lists.freeswitch.org Sent: Monday, July 28, 2008 12:00:42 PM Subject: Re: [Freeswitch-users] Calling web services, JMS or EJB from freeswitch I have generally just used json/rest type calls, otherwise you would have to use a lua addin or some code in js. We do have exposed the curl library so all of the http socket handling should be done, just no specifics of whatever protocol you want. Mike On Jul 28, 2008, at 11:57 AM, Brian West wrote: > I think most if not all of this type of functionality is there in both > lua and javascript. I highly recommend lua. > > /b > > On Jul 28, 2008, at 10:52 AM, Erol Akarsu wrote: > >> >> >> I am interested in calling web services,JMS or EJB from inside >> freeswitch, using lua or javascript. >> Is this functionality existing? >> If not, how hard to add this function? >> If it is not so difficult, then I can volunteer. >> >> Best Regards >> >> Erol Akarsu > > Brian West > sip:brian@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 _______________________________________________ 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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Mod RSS Err for reading news
by unknown 28 Jul '08

28 Jul '08

Hi there, I need your help for the RSS applications under freeswitch. I followed the instructions on http://wiki.freeswitch.org/wiki/Mod_rss and built one extension to test out the RSS in default.xml <extension name="rss"> <condition field="destination_number" expression="^752$"> <action application="set" data="tts_engine=flite"/> <action application="set" data="tts_voice=kal"/> <action application="javascript" data="news.js"/> <action application="rss"/> </condition> </extension> But I still got the errors as follows: 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] mod_spidermonkey.c:2495 js_fetchurl_file() Error! 2008-07-28 15:21:09 [ERR] switch_core_speech.c:60 switch_core_speech_open() inva lid speech module [cepstral]! 2008-07-28 15:21:09 [ERR] mod_rss.c:266 rss_function() Invalid TTS module! Could anybody direct me to the right track? I have the flite TTS engine and want to use that, no cepstral on the freeswitch. Thanks, Chris

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Contributors notes in source files
by Michael Jerris 28 Jul '08

28 Jul '08

If anyone is a contributor to a source file but their name is not in the list of contributors for that file, please contact me off list so that we can get that corrected. Mike

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Weekly Docs Meeting - Need Items to Discuss
by Michael Collins 28 Jul '08

28 Jul '08

FYI, Here's the link to the weekly documentation conf call page: http://wiki.freeswitch.org/wiki/Wiki_meet_2008_07_30 We need items to discuss. What's going on out there that needs more or better documentation? What are the hot items? Thanks, MC

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Extension calling.
by Jair Santos 28 Jul '08

28 Jul '08

Hello, I have two users 1000 and 2001. The first is in the same network as FS. The second is in another network. I am trying to : 1000->NAT->internet->NAT->2001 Both users can call PSTN. I can call 2001->1000. When I try to cal 1000->2001 it reachs the configured gateway and I get invalid number. Below are the console outputs. Does anybody know what could be wrong ? thanks Jair Santos 1- Calling 2001->1000 2008-07-28 09:59:29 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1000(a)192.168.1.117 [15f1d32c-1f5d-4b8d-a795-825ecd980b95] 2008-07-28 09:59:29 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Cliconnect->2001@default 2008-07-28 09:59:29 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/external/2001 [01c73138-db8f-48a5-8c03-21b15806196e] 2008-07-28 09:59:30 [NOTICE] sofia.c:2491 sofia_handle_sip_i_state() Channel [sofia/external/2001] has been answered 2008-07-28 09:59:30 [NOTICE] switch_ivr_originate.c:1298 switch_ivr_originate() Channel [sofia/internal/1000(a)192.168.1.117] has been answered 2008-07-28 09:59:34 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/internal/1000(a)192.168.1.117 [CS_EXECUTE] [NORMAL_CLEARING] 2008-07-28 09:59:34 [NOTICE] switch_ivr_bridge.c:382 audio_bridge_on_exchange_media() Hangup sofia/external/2001 [CS_RESET] [NORMAL_CLEARING] 2008-07-28 09:59:34 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 8 (sofia/external/2001) Ended 2008-07-28 09:59:34 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/external/2001 [CS_HANGUP] 2008-07-28 09:59:34 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 7 (sofia/internal/1000(a)192.168.1.117) Ended 2008-07-28 09:59:34 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1000(a)192.168.1.117 [CS_HANGUP] 2-Calling 1000->2001 2008-07-28 10:27:20 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing FreeSwitch->1000@default 2008-07-28 10:27:20 [INFO] switch_ivr_async.c:1481 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 1 execute_extension::dx XML features 2008-07-28 10:27:20 [INFO] switch_ivr_async.c:1481 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 2 record_session::/usr/local/freeswitch/recordings/2001.2008-07-28-10-27-20.wa v 2008-07-28 10:27:20 [INFO] switch_ivr_async.c:1481 switch_ivr_bind_dtmf_meta_session() Bound B-Leg: 3 execute_extension::cf XML features 2008-07-28 10:27:20 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1000(a)192.168.1.105;user=phone [5de50898-165d-408b-ba90-ae12c44a6b89] 2008-07-28 10:27:20 [NOTICE] sofia.c:2167 sofia_handle_sip_i_state() Ring-Ready sofia/internal/1000(a)192.168.1.105;user=phone! 2008-07-28 10:27:20 [NOTICE] mod_sofia.c:1058 sofia_receive_message() Ring-Ready sofia/doublenat/2001@24.67.78.200:5090! 2008-07-28 10:27:20 [NOTICE] switch_ivr_originate.c:1148 switch_ivr_originate() Ring Ready sofia/doublenat/2001@24.67.78.200:5090! 2008-07-28 10:27:23 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/doublenat/2001@24.67.78.200:5090 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2008-07-28 10:27:23 [NOTICE] switch_ivr_originate.c:1277 switch_ivr_originate() Hangup sofia/internal/1000(a)192.168.1.105;user=phone [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2008-07-28 10:27:23 [ERR] switch_ivr_originate.c:926 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2008-07-28 10:27:23 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: ORIGINATOR_CANCEL 2008-07-28 10:27:23 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 9 (sofia/doublenat/2001@24.67.78.200:5090) Ended 2008-07-28 10:27:23 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 10 (sofia/internal/1000(a)192.168.1.105;user=phone) Ended 2008-07-28 10:27:23 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1000(a)192.168.1.105;user=phone [CS_HANGUP] 2008-07-28 10:27:23 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/doublenat/2001@24.67.78.200:5090 [CS_HANGUP] API CALL [sofia(status)] output: Name Type Data State ============================================================================ ===================== internal profile sip:mod_sofia@192.168.1.117:5060 RUNNING (0) nat profile sip:mod_sofia@24.67.78.200:5070 RUNNING (0) default alias internal ALIASED internal2 profile sip:mod_sofia@24.67.78.200:5061 RUNNING (0) external profile sip:mod_sofia@24.67.78.200:5080 RUNNING (0) inphonex gateway sip:3462101@sip.varphonex.com REGED 192.168.1.117 alias internal ALIASED outbound alias external ALIASED doublenat profile sip:mod_sofia@24.67.78.200:5090 RUNNING (0) ============================================================================ =====================

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grammars
by Cesar Bermudez 28 Jul '08

28 Jul '08

can be in spanish?

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Calling web services, JMS or EJB from freeswitch
by Erol Akarsu 28 Jul '08

28 Jul '08

I am interested in calling web services,JMS or EJB from inside freeswitch, using lua or javascript. Is this functionality existing? If not, how hard to add this function? If it is not so difficult, then I can volunteer. Best Regards Erol Akarsu

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SIP Registrations ..mod_curl
by vidhya sagar dixit 28 Jul '08

28 Jul '08

Hi, First of all sorry to post again as i didn't see my mail in list .. I have installed freeswitch successfully and able to make SIP 2 SIP calls. I want to make registration using mod_curl. So i made following changes .. in xml_curl.conf.xml <configuration name="xml_curl.conf" description="cURL XML Gateway"> <binding name="directory fetcher"> <param name="gateway-url" value="http://localhost/directory.php" bindings="directory"/> </binding> </configuration> in sofia.conf.xml <param name="accept-blind-reg" value="true"/> in directory.php<?phpheader('Content-Type: text/xml');$response = <<< XML<?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="example_curl_directory"> <domain name="192.168.1.10"> <user id="vids"> <params> <param name="password" value="123456"/> </params> </user> </domain> </section></document>XML;die($response);?>No when i am trying to register with user vids pass :123456 i am getting the following message: 2008-07-28 20:02:38 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't find user [vids(a)192.168.1.10] You must define a domain called '192.168.1.10' in your directory and add a user with the id="vids" attribute and you must configure your device to use the proper domain in it's authentication credentials. my conf/directory/default.xml file contains : <include>  <domain name="$${192.168.1.10}"> <params> <param name="dial-string" value=" {presence_id=${dialed_user}@${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_domain}/${dialed_user}@${dialed_domain})}"/<%7Bpresence_id=$%7Bdialed_user%7D@$%7Bdialed_domain%7D,transfer_fallback_extension=$%7Bdialed_user%7D%7D$%7Bsofia_contact($%7Bdialed_domain%7D/$%7Bdialed_user%7D@$%7Bdialed_domain%7D)%7D%22/> > </params> <variables> <variable name="record_stereo" value="true"/> </variables> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> </domain> </include> Can anyone please help me .. where i am doing wrong .. -- Thanks and Regards Vidhya Sagar Dixit

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SIP Registration through mod_curl..
by vidhya sagar dixit 28 Jul '08

28 Jul '08

Hi, I have installed freeswitch successfully and able to make SIP 2 SIP calls. I want to make registration using mod_curl. So i made following changes .. *in xml_curl.conf.xml* <configuration name="xml_curl.conf" description="cURL XML Gateway"> <binding name="directory fetcher"> <param name="gateway-url" value="http://localhost/directory.php" bindings="directory"/> </binding> </configuration> in sofia.conf.xml *<param name="accept-blind-reg" value="true"/> in directory.php *<?php header('Content-Type: text/xml'); $response = <<< XML <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory" description="example_curl_directory"> <domain name="192.168.1.10"> <user id="vids"> <params> <param name="password" value="123456"/> </params> </user> </domain> </section> </document> XML; die($response); ?> * No when i am trying to register with user vids pass :123456 i am getting the following message: * 2008-07-28 20:02:38 [WARNING] sofia_reg.c:1247 sofia_reg_parse_auth() can't find user [vids(a)192.168.1.10] You must define a domain called '192.168.1.10' in your directory and add a user with the id="vids" attribute and you must configure your device to use the proper domain in it's authentication credentials. my conf/directory/default.xml file contains : <include>  <domain name="$${192.168.1.10}"> <params> <param name="dial-string" value="{presence_id=${dialed_user}@ ${dialed_domain},transfer_fallback_extension=${dialed_user}}${sofia_contact(${dialed_domain}/${dialed_user}@ ${dialed_domain})}"/> </params> <variables> <variable name="record_stereo" value="true"/> </variables> <X-PRE-PROCESS cmd="include" data="default/*.xml"/> </domain> </include> Can anyone please help me .. where i am doing wrong .. -- Thanks and Regards Vidhya Sagar Dixit

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Windows compiled binary doesnt generate certificate for tls
by xbipin 28 Jul '08

28 Jul '08

i have been looking for the freeswitch windows binary from long and i found that at freeswitch so i downloaded it etc and works perfect except i wanna test it by installing it on my server but voip is blocked in my country so i have to use tls for signalling but when i enable that in internal.xml then i cant find a way to generate a certificate for it so the ssl folder remains empty. can any1 tell me is there a way to copy a certificate file to that folder as i have 2 certificate generated from voxilla for tls and srtp but both just have the mini cert portion and key file, so how do i put this into a file and use it from freeswitch as the gen_cert command doesnt work to generate certificate files as i think the script is missing for windows or something like that, no idea as im new to this. basically some1 help me just get the device to register using tls, the srtp portion ill look into later on. i did a trace and found out that the tls packets from linksys spa3102 reach the server but nothing happens as certs missing in freeswitch -- View this message in context: http://www.nabble.com/Windows-compiled-binary-doesnt-generate-certificate-f… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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