February 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

xmpp
by Peter P GMX 19 Oct '12

19 Oct '12

I saw that xmpp is supported in Fresswitch. See wiki: http://wiki.freeswitch.org/wiki/Mod_xmpp_event Has anybody already set this up? I have found no mod_xmpp neither in my mod directory nor in the source? There was also a question: " Q: Is it possible to send commands to fs via xmpp? Answer: Yes. " Anybody knows what can be done here and how to do this? Best regards Peter

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FreeSWITCH & Heartbeat / Pacemaker - Please Help setting up
by Avi Marcus 26 Sep '12

26 Sep '12

Hi - I'm setting up 2 parallel computers for a HA setup. (low volume so far, no need to load balance.) I already have heartbeat able to grab the public IP if the first box goes down, but since I'm basically completely unfamiar with pacemaker, I'd like some help. I'd imagine I'm not the only one doing this, there should be no reason for me to recreate the wheel. Info should be on: http://wiki.freeswitch.org/wiki/Freeswitch_HA Specifically: 1) It seems freeswitch takes 8 seconds to restart. So if it crashes (unlikely) calls should immediately get shunted over to the other box. How do I set that up? 2) How do I automatically trigger a sofia recover? I know via commanline it's "fs_cli -x "sofia recover" but where do I tell pacemaker to do that? And I suppose a reload_xml before (maybe after?) is a good idea, too. Also, can I put <param name="track-calls" value="true"/> in a global file, or does it need to be in the actual profile files? And anything else I may be overlooking. Thanks guys! -Avi Marcus

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mod_dingaling picking wrong IP address / no audio?
by Mark Campbell-Smith 14 Sep '12

14 Sep '12

Hi! I am trying to call from my corporate network (firewalled) using Gtalk to Freeswitch. I am not getting any audio. In the logs I see that mod_dingaling is using my internal corporate IP address which is not publically addressable. 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2932 1 candidates 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2960 candidates 146.xx.xx.xx:50320 2009-06-23 22:46:52.140382 [DEBUG] mod_dingaling.c:2993 Acceptable Candidate 146.xx.xx.xx:50320 Further on in the log, I can see GTalk sending a new candidate IP address to use: 2009-06-23 22:46:52.940486 [DEBUG] libdingaling.c:503 New Candidate 1 name=rtp type=local protocol=udp username=e+JTkVHT1xEkqXGD password=fAxU6Pr1oF9Zq48U address=192.168.1.102 port=50322 pref=1.00 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.940486 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:52.957753 [DEBUG] libdingaling.c:503 New Candidate 2 name=rtp type=stun protocol=udp username=RBqyF2XNMYLfJNoU password=DQMjon1fSVoJIRTp address=124.xxx.xxx.xxx port=50323 pref=0.90 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:52.957753 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] and 2009-06-23 22:46:53.788341 [DEBUG] libdingaling.c:503 New Candidate 3 name=rtp type=relay protocol=udp username=62L5zs2FHbcUdeCJ password=KxmNgkUmZsLfuX6S address=209.xx.xxx.xxx port=19295 pref=0.50 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2634 using Existing session for 4085152502 2009-06-23 22:46:53.788341 [DEBUG] mod_dingaling.c:2928 Already picked an IP [146.xx.xx.xx] Because of this, I never get audio. Any ideas how to fix this? Thanks!

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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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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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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

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