June 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How to break a playback with loops
by Daniel Liang 24 Jun '12

24 Jun '12

Hi, I am trying to playing back a wav file a few times, and on receiving a dtmf event, it stops playing the file. The command I sent is: sendmsg call-command: execute execute-app-name: playback execute-app-arg: <the wav file> loops: 4 On receiving a dtmf event, I send a break command: sendmsg call-command: execute execute-app-name: break However, that only breaks one loop of the playback. And I have to press a dtmf for 4 times to stop the file. I tried sending 4 break commands all at once. But it didn't work. And even worse, my program stops receiving dtmfs anymore. My guess is that the first break stops the first loop, but before the second loop starts, the following 3 breaks stop something else that I don't know what they are. Any help is appreciated. Thanks. Daniel Liang

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Re: [Freeswitch-users] Questions on Building an application for FreeSWITCH
by Kristian Kielhofner 09 Jun '12

09 Jun '12

Glad to hear. I still say to build the app and logic completely separately and connect it to FreeSWITCH over the socket using ESL. -- Kristian Kielhofner http://blog.krisk.org ----- Original Message ----- From: freeswitch-users-bounces(a)lists.freeswitch.org <freeswitch-users-bounces(a)lists.freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org <freeswitch-users(a)lists.freeswitch.org> Sent: Wed May 12 18:28:13 2010 Subject: Re: [Freeswitch-users] Questions on Building an application forFreeSWITCHThis is legit. We are not into harassing anyone. There are too many people doing that already. We currently have this working on a Windows/Dialogic platform, but would like to get it over to FreeSWITCH for a number of reasons. It is always nice to avoid spending alot of time developing using a method that you later find has serious issues or limitations and then having to change horses and start over. I just thought I would tap on someone else's wisdom in regards to a good way to talk to FS that works well. I did notice that the wiki talks like LUA is the preferred way to go. Thanks Michael Collins wrote: > > > On Wed, May 12, 2010 at 2:50 PM, Kristian Kielhofner > <kris(a)kriskinc.com <mailto:kris@kriskinc.com>> wrote: > > Are you going to be calling my cell phone to ask if I owe $10,000 > or more to > the IRS? > > Seriously, the world does not need another robodialer to harass > people... > > Agreed, however there are legitimate use cases for this, like a > retailer calling to say an order is in, or reminder calls sent out by > a doctor's office. But yes, if it's dialing for dollars then I'm with > you: Por favor! No mas! > -MC > > > > With that being said, write your app in whatever you want using > ESL and > connect to the socket to originate calls and listen for events. > > > -- > Kristian Kielhofner > http://blog.krisk.org > > ----- Original Message ----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > <mailto:freeswitch-users-bounces@lists.freeswitch.org> > <freeswitch-users-bounces(a)lists.freeswitch.org > <mailto:freeswitch-users-bounces@lists.freeswitch.org>> > To: freeswitch-users(a)lists.freeswitch.org > <mailto:freeswitch-users@lists.freeswitch.org> > <freeswitch-users(a)lists.freeswitch.org > <mailto:freeswitch-users@lists.freeswitch.org>> > Sent: Wed May 12 15:40:58 2010 > Subject: [Freeswitch-users] Questions on Building an application > forFreeSWITCHI am looking for a boot in the right direction on the > following project. > > I am looking to build an "application" with FS where it will > frequently > check a database and then make an outbound call when a record with > certain conditions is found in the DB. It needs to determine if the > call was then answered by a machine/vm, busy, or answered by a person, > play a message and get a response back to be recorded in the db. > It may > find many calls that need to be made at the same time. It will be > running on a PRI. > > I am confused as to what approach to use to accomplish this. > > Lua Script, Perl Script, Javascript, Phython, PHP, Ruby....etc. > Liverpie, ESL....etc. > Or a totally different approach. > > I realize that there is not only one "correct" way to build this, > but I > am looking for advise from those who have done this type of thing > before. I value the opinion of those who have gone down this road and > would be willing to share their thoughts on a recommended path to take > to accomplish this. > > Thanks > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > <mailto:FreeSWITCH-users@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 > <mailto:FreeSWITCH-users@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 > _______________________________________________ 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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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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skypopen scalability tests
by Giovanni Maruzzelli 24 Mar '12

24 Mar '12

On Sat, Apr 30, 2011 at 6:52 PM, Anton VG <anton.vazir(a)gmail.com> wrote: >> Any other ideas on how to proceed for scalability tests? >> > > Hm, since 2.0.0.72 still allows simultaneous calls, you do not have to > register lots of usernames, just 2 of them for 2 PC's > than make 100 calls from 1 machine to another I was going that way, couple years ago, but the Skype client was scaling kind of badly for me. No more than a bunch of simultaneus calls. But at that time I was testing using a standard ALSA sound driver, and Skype client for ALSA. Maybe using Skype for OSS and skypopen.ko it will scale better. Good hint, at least I will check into it in the future. Other ideas? -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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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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How to troubleshoot "503 Maximum Calls In Progress"?
by Chris Fowler 05 Jan '12

05 Jan '12

Hi, On our Production FreeSWITCH box: FreeSWITCH Version 1.0.head (git-4c435ec 2011-03-14 11-54-08 -0500) UP 0 years, 14 days, 6 hours, 28 minutes, 44 seconds, 474 milliseconds, 751 microseconds 12519 session(s) since startup 2 session(s) 0/30 1000 session(s) max min idle cpu 0.00/100.00 At 14:30:37 a reloadxml command was issued. At 14:30:43 the box started rejecting all calls with 503 error. At 14:38 it started working again. Looking at the code I see three conditions can trigger this behavior: if (sess_count >= sess_max || !sofia_test_pflag(profile, PFLAG_RUNNING) || !switch_core_ready()) { nua_respond(nh, 503, "Maximum Calls In Progress", SIPTAG_RETRY_AFTER_STR("300"), TAG_END()); switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "No more sessions allowed at this time.\n"); goto done; } Logs show it wasn't lack of available sessions. How can sofia_test_pflag(profile, PFLAG_RUNNING) or switch_core_ready() fail and busy the system? Thoughts? Thx, Chris.

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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

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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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