February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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get cepstral volume lower
by Madovsky 05 Dec '11

05 Dec '11

I'm trying the SSML example from mod_cepstral wiki <action application="speak" data="cepstral|david|<prosody volume=13>Thank you for holding</prosody>"/> but the voice doesn't say the sentence but says something like "slash blablabla...". Any idea why I can't embed SSML tag in the cepstral sentence ? Thanks

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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

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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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playing files through a remote URL
by Holger Esser 06 Oct '11

06 Oct '11

Hello, I am trying to write a simple ivr script that answers the call, plays a greeting and collects a few digits. Is it possible to fetch the audio through a remote URL (i.e. http://messageServer/en.us/greeting.wav<http://messageserver/en.us/greeting.wav> )? Or does it have to reside on the local file system or a mapped drive? Kind regards, Holger

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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] play_and_get_digits question
by Madovsky 17 Sep '11

17 Sep '11

sorry I think I really didn't undertand the play_and_get_digits concept. is anyone can help ? I just want to use this app to check a pin code, hangup if wrong and continue dialplan if ok Thanks ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 1:01 PM Subject: Re: play_and_get_digits question or maybe this channel var read_result is the result of play_and_get_digits ? ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 12:43 PM Subject: Re: play_and_get_digits question Ok I think I misunderstood the concept of var where digits should be put in. it's with this var I need to ckeck after this application. self forum thank you ;) ----- Original Message ----- From: Madovsky To: freeswitch-users(a)lists.freeswitch.org Sent: Sunday, February 06, 2011 12:29 PM Subject: play_and_get_digits question I have this lines in my dialplan <action application="play_and_get_digits" data="4 4 3 5000 # phrase:wait_foo phrase:bad_foo foo ${foo}"/> <action application="transfer" data="otherext XML features"/> if I don't press any DTMF, after 4 loops the dialplan continues, but I thought that it was automatically hangup in case of no digits bad digits... Did I forget something ? Thanks Franck

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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.

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Complete wholesale app in freeswitch
by David Villasmil 21 Jul '11

21 Jul '11

Hello Guys, I'm finishing a "complete" wholesale application created on freeswitch and I was wondering whether it would be a good idea to put it up on the wiki. I just don't know how. Features include all the following parameters configurable via web interface: - Profile creation based on server IP where traffic is received. You can have multiple IPs, system will create multiple profiles/diaplans so it can differentiate. - i.e. offer to the same customer a "gold" routing on IP1 and cheap routing on IP2 - Customer add/modify/delete - IP source - Rates for client routes based on areacode - Prepaid or postpaid. - When cutomer balance is 0, no more calls are allowed. - limit max channels - Media by-pass - When by-passed, customer and provider will exchange RTPs directly. Else, server will be in the middle. - Provider add/modify/delete - costs for provider routes based on areacode - limit max channels - Routing based on areacode, gives great granularity. - Routes can be assigned multiple gateways/providers which can in turn be distributed based on weigth. Includes overflow to next configured GW. - Basic financial report generation (totals) by customer/provider - Basic traffic ASR/ACD report (totals) by cutomer/provider - Basic user administration. (No access level, only total access) - CDR export to csv file. I also have a prepaid card app... no web interface on that one though... Thanks all David

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