October 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_dingaling & inbound audio
by Locutis of Borg 14 Sep '12

14 Sep '12

Followed instructions on wiki. Enabled mod_dingaling to register over at GV. Set external rtp to right address. Outbound calling terminates, and outbound audio works, but seems that I can not get return audio. FS is on DMZ - same problem Moved to internal (behind NAT, port forwarding) - same problem Moved back to DMZ and killed iptables and router spi - same problem In and out calling on SIP to provider works fine in both cases. So my questions are: Can mod_dingaling be used for GV or not? Since it uses TLS, NAT shouldn't be an issue? and Does GV not like us FS users? Thank you for you help. "Resistance is Fruitful"

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caller_id_number
by Boris Kovalenko 16 Aug '12

16 Aug '12

Hello! When the call is incoming for my network I want CID is translated by my rules. So I translate caller_id_number and set effective_caller_id_number. But there is a little problem - when the call arrived to a analog phone (for example Panasonic KTX-TCD540) I see two number on display - original caller_id_number and effective_caller_id_number. Is there a way to not show caller_id_number in CID? Or may I modify caller_id_number instead of setting effective_caller_id_number? -- Regards, Boris

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ESL problem
by Thangappan.M 31 Jul '12

31 Jul '12

Dear all, In the previous post, I got the information that using event outbound socket we can implement the IVR and also see the example in libs/esl/perl/server2.pl. I have seen it and understood the flow of the script.But when I was running that script it tells the following error. Can't locate loadable object for module ESL in @INC (@INC contains: /etc/perl /usr/local/lib/perl/5.8.8 /usr/local/share/perl/5.8.8 /usr/lib/perl5 /usr/share/perl5 /usr/lib/perl/5.8 /usr/share/perl/5.8 /usr/local/lib/site_perl .) at ESL.pm line 11 Compilation failed in require at server2.pl line 1. Then I checked the 11th line in the ESL.pm that line is (bootstrap ESL;) But in perl directory there is no directory called ESL. What would be the issue?. Is ESL necessary is necessary for implementing IVR using event outbound socket? -- Regards, Thangappan.M

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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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audo sync issues with record_session to mp3
by Tom Parrott 13 Jan '12

13 Jan '12

Hi, We are running a cluster of 3 SIP servers all running Freeswitch. Each call is recorded using the record_session application to record to mp3 files using mod_shout. Longer calls, after about 10 minutes start to introduce sync issues between the A-leg and the B-leg. We are running record_session on the A-leg, and it seems to get ahead of the B-leg. For example the caller on the A-leg will be heard to answer a question whilst the person on the B-leg is asking it. I am aware that recording to .mp3 can use additional resources, but these are quad core servers with 4GB of ram and they are only doing around 10 concurrent calls each. Additionally I have seen others experiencing these issues with Wav files too: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-July/060107.html I am using the git head version from Tue 11 Oct 2011. I have also tried turning off enable_file_write_buffering on the A-leg however this makes no difference either. Does anyone have any suggestions? Thanks

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Overlap dialing problem
by Štefan Čudai 10 Jan '12

10 Jan '12

Hi, does FreeSwitch support for overlap dialing or did somebody try to use it? I am using Sangoma card A102 under Windows 7 with FreeTDM. I have configured my freetdm.conf.xml with this two lines: <param name="overlap" value="yes"/> <param name="min_digits" value="0"/> But when I'm dialing number (which has 14 digits) I hear disconnect tone after dialing first 5 digits. I think that reason is because FreeTDM module is waiting number with minimum digits equals to "0" (see my min_digits). I have also try set min_digits to 20 and then dial same number with 14 digits but result was same. So, I looked to log file and found that FreeTDM module got in thist case STATUS CONFIRM message but he was still waiting for next digits and his time expired. Can you help me please? Thanks a lot. Stefan

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