May 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Possible mod_smpp
by Michael Collins 08 Nov '12

08 Nov '12

Hello all, If you are interested in giving monetary support to have a professional software firm create mod_smpp then please contact me off list and I will give you more details. Thanks, Michael

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Mod_limit questions
by ivdreg ivdreg 29 Oct '12

29 Oct '12

Hi All, I have some questions regarding mod_limit. 1. I'm using 16674 in production but tested and with 17001. I try mod_limit with odbc/mysql and sqlite and have same problem: mod_limit using limit_hash never tries to write data to SQL. It works (counts sessions) on log screen but no data in MySQL/Sqlite. It seems that data is cached but cache never flushed to DB. Is anyone has this problem. My limits are: <action application="limit_hash" data="op_${sip_h_X-SRC-Cust} ${destination_number} 5 !INCOMING_CALL_BARRED" /> <action application="limit_hash" data="in_${sip_h_X-SRC-Cust} ${caller_id_number} 10/300 !OUTGOING_CALL_BARRED" /> and they are applied on inbound call profile in same extension. 2. I need to implement rule maximum of 50 calls per day/24h from the same CLID (A NUMBER) after that calls to be limited. I'm not sure but rule in example "Limit calls to 5 calls every 10 minutes" is about concurrent call but not calls processed as count. Any idea ? Thanks for any help

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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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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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