December 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Facebook...
by Klaus Hochlehnert 27 May '12

27 May '12

Hi Guys, just saw that you already have a Facebook entry. Wouldn't it be possible to connect the news/twitter feeds somehow with Facebook? I'd really like to see all news on Facebook... Cheers, Klaus

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Not recognizing voice in mod_pocketsphinx
by Thangappan.M 30 Jan '12

30 Jan '12

Dear all, I have installed the mod_pocketsphinx module successfully and enabled with FreeSWITCH. Copied the relevant files for pizza application and placed into appropriate directory. Made a call to "74992" and able to hear the voice. When I said "TAKEOUT' it was not recognizing my words. In the freeswitch CLI it shown the following messages. EXECUTE sofia/internal/1000(a)192.168.1.222 javascript(ps_pizza.js) 2010-09-06 13:18:48.023673 [DEBUG] sofia_glue.c:2594 AUDIO RTP [sofia/internal/1000(a)192.168.1.222] 192.168.1.222 port 32590 -> 192.168.6.60 port 8000 codec: 8 ms: 20 2010-09-06 13:18:48.024705 [DEBUG] switch_rtp.c:1182 Starting timer [soft] 160 bytes per 20ms 2010-09-06 13:18:48.025710 [DEBUG] sofia_glue.c:2774 Set 2833 dtmf send payload to 101 2010-09-06 13:18:48.026799 [DEBUG] sofia_glue.c:2779 Set 2833 dtmf receive payload to 101 2010-09-06 13:18:48.026799 [DEBUG] mod_sofia.c:636 Local SDP sofia/internal/ 1000(a)192.168.1.222: v=0 o=FreeSWITCH 1283726738 1283726739 IN IP4 192.168.1.222 s=FreeSWITCH c=IN IP4 192.168.1.222 t=0 0 m=audio 32590 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2010-09-06 13:18:48.028235 [DEBUG] switch_core_session.c:641 Send signal sofia/internal/1000(a)192.168.1.222 [BREAK] 2010-09-06 13:18:48.028235 [DEBUG] sofia.c:4153 Channel sofia/internal/ 1000(a)192.168.1.222 entering state [completed][200] 2010-09-06 13:18:48.029256 [NOTICE] mod_spidermonkey.c:2066 Channel [sofia/internal/1000(a)192.168.1.222] has been answered 2010-09-06 13:18:48.030269 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms 2010-09-06 13:18:48.123555 [DEBUG] sofia.c:4153 Channel sofia/internal/ 1000(a)192.168.1.222 entering state [ready][200] 2010-09-06 13:18:48.171705 [DEBUG] switch_rtp.c:2066 Correct ip/port confirmed. 2010-09-06 13:18:50.212417 [DEBUG] switch_ivr_play_say.c:1444 done playing file EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(pocketsphinx pizza_order undefined) 2010-09-06 13:18:51.768030 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000(a)192.168.1.222 EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(resume) 2010-09-06 13:18:51.774074 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms 2010-09-06 13:18:54.492023 [DEBUG] switch_ivr_play_say.c:1444 done playing file EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(resume) EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(resume) 2010-09-06 13:19:01.534474 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms Please help me..... -- Regards, Thangappan.M

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FreeSWITCH cluster with IM/Presence
by Even André Fiskvik 22 Dec '11

22 Dec '11

Hi. I currently have a project where I'm researching how to establish a clustered and scalable platform with: - registration - presence/blf between eyebeam/bria and Linksys SPA962 clients - IM between eyebeam/bria clients. - calling between the clients The first scenario I want to test is with two FreeSWITCH servers and shared ODBC db running in a "cluster". How the registrations and calls etc. are balanced between them is not decided yet, but I'm thinking that a UltraMonkey setup or DNS SRV records could to the trick here? The main challenge in this setup is sharing presence, and doing call-setup and IM between two FreeSWITCH servers (which can be scaled by adding n-nodes). I'm interested in user's that can confirm wether this solution is possible or not with FreeSWITCH and it's current state, and what it would take to make something like this work. If there's any part of FS that is lacking some implementation to make such a scenario work, we are interested in providing the funding to support those features. I've seen that the core db has added support for ODBC, but I'm not sure wether this is something that would be helpful in this scenario or not. This is only conceptual right now, so I'm interested in paying someone that could help me configure the two FS servers to support this scenario on an hourly basis to get a quick proof-of-concept up and running. The proof-of-concept has to be completed by March 19th, after which we will evaluate what technologies and setup we want to commit to. Some other variations I'm concidering is Opensips/Kamailio -> Asterisk, Opensips/Kamailio -> FreeSWITCH, Opensips/Kamailio -> Asterisk and Openfire + Asterisk-im plugin for IM/presence. I'm also open to other suggestions, but I'd like this topic to stay on how FreeSWITCH could facilitate such a platform. Best regards, Even André Fiskvik CTO - Oyatel AS

2 1

Grandstream GXP 2000
by Jonas Gauffin 18 Dec '11

18 Dec '11

Hello I have problems with TCP and GXP2000 phones. They work fine using UDP, but no calls arrive if I switch to TCP. I've upgraded to the latest GS firmware today to see it that helped, but it didn't. Have anyone else had problems with TCP? Regards, Jonas

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Conference and Speex Wideband audio quality problem
by Richard Alam 23 Nov '11

23 Nov '11

Hi Everyone, Hopefully, the answer to my problem isn't in the book as I've yet to order it. :) First off, I'm one ot the devs of BigBlueButton (http://code.google.com/p/bigbluebutton/) We currently use Asterisk for voice conference. Leif Jackson made BBB work with FreeSWITCH. Now, from BBB, a user can join a voice conference in FS. We have a java SIP client (based on Red5Phone) that calls FS. All is working well for Ulaw. I am now trying to make Speex WB work since Flash Player (where our client runs) supports Speex WB. Using Speex WB frees the java SIP client from transcoding between Nellymoser and Ulaw. The java SIP client just takes the speex audio data from RTMP and sends it to FS through RTP. However, the sound quality is bad. It is choopy and robotic. Let me describe what I have done and what I found from debugging so far. I modified vars.xml to add the speex codec. ================ <X-PRE-PROCESS cmd="set" data="global_codec_prefs=speex@16000h@20i,speex@8000h@20i,G7221@32000h,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=speex@16000h@20i,PCMU,PCMA,GSM"/> Setup the conference in dialplan/public.xml ================== <extension name="bbb_conferences"> <condition field="destination_number" expression="^(7\d{4})$"> <action application="answer"/> <action application="conference" data="$1@wideband"/> </condition> </extension> When I start up freeswitch, it looks like it's loading the speex codec fine. Full console log is here http://pastebin.com/rnHfgQYj =============== 2010-07-30 16:04:10.620375 [NOTICE] switch_loadable_module.c:251 Adding Application 'AvoidingDeadlock' 2010-07-30 16:04:10.620658 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_g723_1.so **/opt/freeswitch/mod/mod_g723_1.so: cannot open shared object file: No such file or directory** 2010-07-30 16:04:10.620726 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_g729.so **/opt/freeswitch/mod/mod_g729.so: cannot open shared object file: No such file or directory** 2010-07-30 16:04:10.621189 [CRIT] switch_loadable_module.c:926 Error Loading module /opt/freeswitch/mod/mod_amr.so **/opt/freeswitch/mod/mod_amr.so: cannot open shared object file: No such file or directory** 2010-07-30 16:04:10.621547 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_ilbc] 2010-07-30 16:04:10.621618 [NOTICE] switch_loadable_module.c:185 Adding Codec 'iLBC' (iLBC) 8000hz 30ms 2010-07-30 16:04:10.621647 [NOTICE] switch_loadable_module.c:185 Adding Codec 'iLBC' (iLBC) 8000hz 20ms 2010-07-30 16:04:10.622043 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_speex] 2010-07-30 16:04:10.622108 [NOTICE] switch_loadable_module.c:185 Adding Codec 'SPEEX' (Speex) 32000hz 20ms 2010-07-30 16:04:10.622135 [NOTICE] switch_loadable_module.c:185 Adding Codec 'SPEEX' (Speex) 16000hz 20ms 2010-07-30 16:04:10.622160 [NOTICE] switch_loadable_module.c:185 Adding Codec 'SPEEX' (Speex) 8000hz 20ms 2010-07-30 16:04:10.622513 [CONSOLE] switch_loadable_module.c:944 Successfully Loaded [mod_h26x] When I call into the conference, it looks like mod_conference does not use speex codec but L16@16000hz. ====== 2010-07-30 16:05:56.874138 [DEBUG] sofia.c:4317 Channel sofia/external/75115(a)192.168.0.166 entering state [completed][200] 2010-07-30 16:05:56.874138 [DEBUG] switch_core_session.c:647 Send signal sofia/external/75115(a)192.168.0.166 [BREAK] 2010-07-30 16:05:56.874138 [DEBUG] switch_channel.c:2494 (sofia/external/75115(a)192.168.0.166) Callstate Change RINGING -> ACTIVE 2010-07-30 16:05:56.874138 [NOTICE] mod_dptools.c:746 Channel [sofia/external/75115(a)192.168.0.166] has been answered EXECUTE sofia/external/75115(a)192.168.0.166 conference(75115@wideband) 2010-07-30 16:05:56.875481 [DEBUG] mod_conference.c:5237 Raw Codec Activation Success L16@16000hz 1 channel 20ms 2010-07-30 16:05:56.875481 [DEBUG] mod_conference.c:5282 Raw Codec Activation Success L16@16000hz 1 channel 20ms 2010-07-30 16:05:56.877940 [DEBUG] switch_core_codec.c:116 sofia/external/75115(a)192.168.0.166 Push codec L16:10 tport_wakeup_pri(0x95bfd98): events IN Is this the correct behavior? I've tried using xlite to connect to the conference, and the audio is good. Also, when connecting from our client, the audio file that gets played by FS to notify how many users are in the conference is good. It's my voice that's bad. I tried to use the echo application instead of the conference. When I call from the BBB client, the sound quality is good. I appreciate any tips on how I can fix the problem or proceed to figure out how to fix it. Thank you very much. Richard -- --- BigBlueButton http://www.bigbluebutton.org http://code.google.com/p/bigbluebutton

3 4

switch_rtp.c - Auto Changing Port
by Ross McKillop 27 Oct '11

27 Oct '11

Hi, I have a number of SIP endpoints behind NAT on a cellular network and am experiencing some odd behaviour with RTP traffic. No audio at all is present on the call for about the first 10s or so, and then the following is shown in the console; > 2010-09-06 01:04:53.101801 [INFO] switch_rtp.c:2049 Auto Changing port from 10.21.6.213:4000 to 217.171.129.67:22263 After which the call works perfectly. Is there any way I can force this process to happen quicker, within the first second of the call? or is it possible to force FreeSWITCH to use the external IP from the start? Any assistance you can offer would be much appreciated. Best Regards, Ross

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Hardware phones that do 48Khz Celt: Was Re: Is there any license G729?
by Craig Askings 29 Sep '11

29 Sep '11

Are there any hardware phones that support 48 Khz Celt and automated/mass deployment? Craig. 2009/6/30 Jason White <jason(a)jasonjgw.net>: > Brian West <brian(a)freeswitch.org> wrote: >> Everyone has this need for lower bandwidth calls... I tend to march the >> other way. 48kHz baby! (btw you can do 48kHz in the same bandwidth as a >> single ulaw call) > > 48khz Celt (celt@48000 in your codec preferences) sounds wonderful with > FreeSWITCH. To test, run two FreeSWITCH instances, both with mod_portaudio. > This also works well in 48khz conferences. > > I wouldn't use G.729 even if it weren't encumbered by patents - it's G.711, > G.722, G.722.1 and (my current favourite) Celt all the way. > > > _______________________________________________ > 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 > -- Craig Askings Network Engineer | Over the Wire Pty Ltd craig(a)overthewire.com.au | www.overthewire.com.au Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365

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Re: [Freeswitch-users] Radius/Authentication/Authorization
by Abid Saleem 06 Aug '11

06 Aug '11

Dear Tihomir, Any update and help on the below please. Abid From: abid_freeswitch(a)live.com To: tculjaga(a)gmail.com CC: neal(a)wanlink.com Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization Date: Wed, 31 Mar 2010 19:35:32 +0500 Dear Tihomir, Thank you very much for the configuration example but in which files to place these configurations. Please bear with me because I am new to FreeSwitch and if you could provide complete steps. Also when I compile FS and load the module mod_rad_auth in conf/autoload_configs/modules.conf.xml, I get an error while starting FS as follows. 2010-03-31 19:32:29.399466 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_rad_auth.so**/usr/local/freeswitch/mod/mod_rad_auth.so: undefined symbol: rc_conf_str** Please advise. Thanks. Regards-----------Abid Saleem --Forwarded Message Attachment-- From: tculjaga(a)gmail.com CC: neal(a)wanlink.com To: freeswitch-users(a)lists.freeswitch.org Date: Tue, 30 Mar 2010 21:42:03 +0200 Subject: Re: [Freeswitch-users] Radius/Authentication/Authorization hello, here is an example <configuration name="rad_auth.conf" description="radius authentification module"> <settings>   </settings> <client> <param name="authserver" value="10.1.1.10:1812:gateway"/> <param name="dictionary" value="/usr/local/etc/radiusclient/dictionary.all"/> <param name="seqfile" value="/var/run/radius.seq"/> <param name="mapfile" value="/usr/local/etc/radiusclient/port-id-map"/> <param name="default_realm" value=""/> <param name="radius_timeout" value="3"/> <param name="radius_retries" value="2"/> <param name="radius_deadtime" value="0"/> <param name="bindaddr" value="*"/> </client> <vsas>   <param name="h323-conf-id" id="24" value="CALLID" pec="9" expr="1" direction="in"/> <param name="h323-prompt-id" id="104" value="SERVICENUM" pec="9" expr="1" direction="in"/> <param name="Cisco-AVPair" id="1" value="TRANSACTIONID" pec="9" expr="1" direction="in"/> <param name="Calling-Station-Id" id="31" value="CALLINGNUMBER" pec="0" expr="1" direction="in"/> <param name="NAS-Port-Type" id="61" value="0" pec="0" expr="0" direction="in"/> <param name="NAS-Port-Id" id="87" value="ISDN 3/0:D:14" pec="0" expr="0" direction="in"/> <param name="Login-User" id="1" value="1" pec="0" expr="0" direction="in"/>  <param name="BILING_MODEL" id="109" value="billing_model" pec="9" expr="0" direction="out"/> <param name="CREDIT_AMOUNT" id="101" value="credit_amount" pec="9" expr="0" direction="out"/> <param name="CURRENCY" id="110" value="currency" pec="9" expr="0" direction="out"/> <param name="PREFFERED_LANG" id="107" value="preffered_lang" pec="9" expr="0" direction="out"/> <param name="CREDIT_TIME" id="102" value="credit_time" pec="9" expr="0" direction="out"/> <param name="H323-IVR-IN:DIRATION" id="1" value="h323_ivr_duration" pec="9" expr="0" direction="out"/> <param name="RADIUS_RETURN_CODE" id="103" value="return_code" pec="9" expr="0" direction="out"/>  </vsas> </configuration> in the dialplan you need to trigger auth as: <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> there are two behaviours: 1. authorize the call according to username&pass and dialed number - if authorized, the radius server returns credit time towards the dialed number 2. authorize the call according to username&pass - if authorized, the radius server returns the current account balance <extension name="unitest_rad-ANI-auth"> <condition field="destination_number" expression="^601$"> <action application="log" data="INFO PRIJE RAD_AUTH "/> <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/> <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/> <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>  <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/> <action inline="true" application="set" data="USERNAME=38516060333"/>  <action inline="true" application="set" data="PASSWD=003282"/>  <action application="sleep" data="2000"/> <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> <action application="log" data="INFO AUTH_RESULT=${AUTH_RESULT}"/> <action application="log" data="INFO billing_model=${billing_model}"/> <action application="log" data="INFO credit_amount=${credit_amount}"/> <action application="log" data="INFO currency=${currency}"/> <action application="log" data="INFO preffered_lang=${preffered_lang}"/> <action application="log" data="INFO credit_time=${credit_time}"/> <action application="log" data="INFO h323_ivr_duration=${h323_ivr_duration}"/> <action application="log" data="INFO return_code=${return_code}"/>  </condition> </extension> <extension name="unitest_rad-ANI-balance"> <condition field="destination_number" expression="^602$"> <action application="log" data="INFO PRIJE RAD_AUTH "/> <action inline="true" application="set" data="CALLID=h323-conf-id=${uuid}"/> <action inline="true" application="set" data="SERVICENUM=h323-prompt-id=${destination_number}"/> <action inline="true" application="set" data="TRANSACTIONID=h323-ivr-out=transactionID:1234"/>  <action inline="true" application="set" data="CALLINGNUMBER=38516060333"/> <action inline="true" application="set" data="USERNAME=38516060333"/>  <action inline="true" application="set" data="PASSWD=003282"/> <action inline="true" application="set" data="DIALED_NUMBER=16094191500"/> <action application="sleep" data="2000"/> <action application="auth_function" data="in ${DIALED_NUMBER}, in ${USERNAME}, in ${PASSWD}, out AUTH_RESULT"/> <action application="log" data="INFO AUTH_RESULT=${AUTH_RESULT}"/> <action application="log" data="INFO billing_model=${billing_model}"/> <action application="log" data="INFO credit_amount=${credit_amount}"/> <action application="log" data="INFO currency=${currency}"/> <action application="log" data="INFO preffered_lang=${preffered_lang}"/> <action application="log" data="INFO credit_time=${credit_time}"/> <action application="log" data="INFO h323_ivr_duration=${h323_ivr_duration}"/> <action application="log" data="INFO return_code=${return_code}"/>  </condition> </extension> will update the wiki by the end of the week. you have enough information for now. Tihomir. On Tue, Mar 30, 2010 at 1:52 PM, Abid Saleem <abid_freeswitch(a)live.com> wrote: Hi Neal and other Contributors to FS, I recieved an answer on the list that mod_rad_auth is ready. I upgraded FS to download and install it by "make current" and it is successfully built and installed. Could you please outline the detailed steps to do all this configuration. If it is still not ready, is there any other method that somebody has already implemented like perl scripts etc. Any help in this regard is badly required. THanks for your cooperation. Regards----------Abid SaleemSr. Product Manager Hotmail: Powerful Free email with security by Microsoft. Get it now. _________________________________________________________________ Hotmail: Powerful Free email with security by Microsoft. https://signup.live.com/signup.aspx?id=60969

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api_hangup_hook and hupall
by Dome Charoenyost 04 Aug '11

04 Aug '11

I use api_hangup_hook for do something when call hangup. but when i use hupall command from CLI api_hangup_hook do nothing. Is it bug ? Dome C.

3 3

mod_pocketsphinx
by roy 19 Jun '11

19 Jun '11

Hi, Trying to get mod_pocketsphinx working from the current git tree, FreeSWITCH version: 1.0.head (git-2629a57 2010-06-16 22-42-15 +0200) however, I got this error during startup: <snip> 2010-06-17 11:18:08.032645 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_pocketsphinx.so **/usr/local/freeswitch/mod/mod_pocketsphinx.so: undefined symbol: ngram_model_get_counts** </snip> Lib paths were already defined in /etc/ld.so.conf as: include /usr/local/freeswitch/lib /usr/local/freeswitch/mod # ls -al mod/mod_pocketsphinx.* -rwxr-xr-x 1 root staff 1467 Jun 17 10:49 mod/mod_pocketsphinx.la -rwxr-xr-x 1 root staff 905769 Jun 17 10:49 mod/mod_pocketsphinx.so Was wondering if someone was able to get this working. Manually downloading the current pocketsphinx and sphinxbase and renaming them into 0.4.99 and rebuilding didn't help either. Would appreciate if anyone can point me into the right direction. TIA. Roy

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