August 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Recommended ULIMIT settings
by Nuno Reis 13 Aug '13

13 Aug '13

Hi guys. I'm currently using the recommended ulimit settings as defined here: http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations Although there's at least one parameter that doesn't seem to be right when i do a status at fs_cli: UP 0 years, 0 days, 10 hours, 25 minutes, 14 seconds, 617 milliseconds, 626 microseconds FreeSWITCH is ready 55390 session(s) since startup 0 session(s) 0/100 5000 session(s) max min idle cpu 0.00/98.00 *Current Stack Size/Max 240K/240K* The stack size should be 8192 as defined, but FS shows otherwise. I'm using centOS x86_64 and a 64 bit version of FS. Is this normal? Thanks. -- *Nuno Miguel Reis* | *Unified Communication** Systems* M. +351 913907481 | nreis(a)wavecom.pt WAVECOM-Soluções Rádio, S.A. Cacia Park | Rua do Progresso, Lote 15 3800-639 AVEIRO | Portugal T. +351 309 700 225 | F. +351 234 919 191 *GPS<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…>| www.wavecom.pt** <http://www.wavecom.pt/>* [image: Description: Description: WavecomSignature]<http://www.wavecom.pt/pt/wavecom/premios.php> [image: Publicity] <http://www.wavecom.pt/pt/mail_eventos.php>

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Re: [Freeswitch-users] Real-time billing application for the FreeSWITCH (mod_lua, mod_perl or ESL)
by Sami Montour 13 Aug '13

13 Aug '13

Thanks Mike. I appreciate it. From: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Michael Jerris Sent: Monday, August 12, 2013 10:25 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Real-time billing application for the FreeSWITCH (mod_lua, mod_perl or ESL) If your happy with C, go for it. You can take a look at mod_xml_cdr and mod_nibblebill for some hints on hooks for billing. I might use the state hooks for reporting instead and the data you get from that one as you are less time constrained to handle the data in that state. Mike On Aug 9, 2013, at 2:29 PM, Sami Montour <smontour(a)verizon.net> wrote: Hello Everyone, I am a long time user of OpenSIPS and have recently been tasked with developing a call control/real-time billing application for the FreeSWITCH. I would like to develop the application in either C/C++, Perl or lua (totally new to lua) in this order. I have used ESL to develop some basic functionality of the application using Perl. For instance, when the call comes in, I pass over control from FreeSWITCH to the call control application using outbound TCP socket. Then the call control application listens for CHANNEL_ANSWER event to start billing process and CHANNEL_HANGUP event to stop billing. The application does some database work as well to retrieve and check user status and balance and all that stuff. My main concern is with load testing because of the TCP connection between the FreeSWITCH and call control application and also TCP connection to database to retrieve subscriber info. The basic functionality seems to work fine but haven't tested it with high call volume, which I am about to do. The last couple of days I have been experimenting with mod_lua but can't find a method under "Session" to get information related to CHANNEL_ANSWER and CHANNEL_HANGUP events. The "session:geVariable" does not have variables related to session answer time or hangup time, something similar to "$e->getHeader("Caller-Channel-Answered-Time")" in ESL. I must be missing something very simple here. Again, my experience with mod_lua is only a couple of days old. I am new Freeswitch and thought I would seek the advice of all the experts and people with more experience with Freeswitch. For this type of application, am I better off with mod_lua, or mod_perl, or shall I stick with ESL using Perl? How about ESL with C/C++? Will I gain some performance gain there? Any feedback would be appreciated. Thanks. Sam

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freeswitch trunk cisco call manager
by Nick Zaitsev 13 Aug '13

13 Aug '13

Good day to you, Could you help me to configure the trunk between freeswitch and cisco call manager at the freeswitch side,please. I can not find any information abot that in the wiki page. Thank you for your time, Best regards. Nikolay Zaitsev. -- Nick Zaitsev

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mod_nibblebill.c:558 Failed to log to database
by Veniamin (Benjamin) Beskrovny 13 Aug '13

13 Aug '13

Hi everyone. Can anyone tell me how can I troubleshoot that kind of error: mod_nibblebill.c:558 Failed to log to database PS: using MySQL via ODBC. In nibblebill conf I have: odbc-dsn="odbc://billing" ISQL test: isql billing - works good. Show tables, queries, e.t.c. Any ideas?

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Invalid application oreka_record
by Bob McCarthy 13 Aug '13

13 Aug '13

I am trying to test/use mod_oreka, I have configured via http://wiki.freeswitch.org/wiki/Mod_oreka but in the dialplan I get 2013-08-12 16:35:09.231636 [ERR] switch_core_session.c:2557 Invalid Application oreka_record Is there something more I need to do to compile/inlude the oreka module ? mod_oreka.so is present in the mod directory Bob

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nibblebill query error
by Veniamin (Benjamin) Beskrovny 13 Aug '13

13 Aug '13

After updating nibblebill conf and freeswitch I've mentioned the following error: [ERR] mod_nibblebill.c:403 Error running this query: [SELECT cash AS nibble_balance FROM clients WHERE accountcode='1' Running isql from the command line with the same sql gives perfect answer: [root@sip log]# isql billing +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> SELECT cash AS nibble_balance FROM clients WHERE accountcode=1 +-------------------------+ | nibble_balance | +-------------------------+ | -3126.29999017714 | +-------------------------+ SQLRowCount returns 1 1 rows fetched SQL> What could be the problem? Any ideas?

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Httapi playback: input-timeout
by Denis Gasparin 13 Aug '13

13 Aug '13

Hi. I setup an ivr using httapi and I've got a strange behaviour with input-timeout and digit-timeout attributes. The ivr xml is: <?xml version="1.0"?> <document type="text/freeswitch-httapi"> <work> <playback file="ivr/ivr-please_enter_the_number_where_we_can_reach_you.wav" name="FluentIvrDigitInput" terminators="#" input-timeout="10000" digit-timeout="5000"> <bind action="--my_bind_action_url--">~^\d+#</bind> </playback> <continue action="--my_continue_action--"/> </work> </document> With no input, t he timeout triggers after 10s but I was expecting 5s according to digit-timeout. The relevant log of the call is: 2013-08-13 11:46:10.315627 [DEBUG] mod_httapi.c:1217 Process Tag: [playback] 2013-08-13 11:46:10.315627 [INFO] switch_ivr_async.c:212 Digit parser HTTAPI: Setting realm to 'default' 2013-08-13 11:46:10.315627 [DEBUG] switch_ivr_async.c:318 Digit parser HTTAPI: binding ^\d+#/default/0 callback: 0x7fa1e8486bd0 data: 0x880478 2013-08-13 11:46:10.315627 [INFO] switch_ivr_async.c:212 Digit parser HTTAPI: Setting realm to 'default' 2013-08-13 11:46:10.315627 [INFO] switch_ivr_async.c:201 Digit parser HTTAPI: Setting terminators for realm 'default' to '#' 2013-08-13 11:46:10.315627 [DEBUG] switch_ivr_play_say.c:1315 Codec Activated L16@8000hz 1 channels 20ms 2013-08-13 11:46:12.275614 [DEBUG] switch_ivr_play_say.c:1703 done playing file /u/freeswitch/sounds/en/us/callie/ivr/ivr-please_enter_the_number_where_we_can_reach_you.wav 2013-08-13 11:46:22.275617 [DEBUG] mod_httapi.c:1217 Process Tag: [continue] With a single digit input, the timeout triggers after 5s: i was expecting 10s (according to input-timeout). The log follows: 2013-08-13 11:48:34.335616 [DEBUG] mod_httapi.c:1217 Process Tag: [playback] 2013-08-13 11:48:34.335616 [INFO] switch_ivr_async.c:212 Digit parser HTTAPI: Setting realm to 'default' 2013-08-13 11:48:34.335616 [DEBUG] switch_ivr_async.c:318 Digit parser HTTAPI: binding ^\d+#/default/0 callback: 0x7fa1e8486bd0 data: 0x7fa1ec0b7718 2013-08-13 11:48:34.335616 [INFO] switch_ivr_async.c:212 Digit parser HTTAPI: Setting realm to 'default' 2013-08-13 11:48:34.335616 [INFO] switch_ivr_async.c:201 Digit parser HTTAPI: Setting terminators for realm 'default' to '#' 2013-08-13 11:48:34.335616 [DEBUG] switch_ivr_play_say.c:1315 Codec Activated L16@8000hz 1 channels 20ms 2013-08-13 11:48:36.295615 [DEBUG] switch_ivr_play_say.c:1703 done playing file /u/freeswitch/sounds/en/us/callie/ivr/ivr-please_enter_the_number_where_we_can_reach_you.wav 2013-08-13 11:48:37.375645 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 1:880 2013-08-13 11:48:37.375645 [DEBUG] switch_channel.c:471 RECV DTMF 1:880 2013-08-13 11:48:42.395644 [DEBUG] mod_httapi.c:1217 Process Tag: [continue] With a multi-digit input, the timeout triggers after 10s from the end of playback (according to input-timeout). I was expecting a 10s timeout after each digit. 2013-08-13 11:51:44.715652 [DEBUG] mod_httapi.c:1217 Process Tag: [playback] 2013-08-13 11:51:44.715652 [INFO] switch_ivr_async.c:212 Digit parser HTTAPI: Setting realm to 'default' 2013-08-13 11:51:44.715652 [DEBUG] switch_ivr_async.c:318 Digit parser HTTAPI: binding ^\d+#/default/0 callback: 0x7fa1e8486bd0 data: 0x7fa1dc3c0468 2013-08-13 11:51:44.715652 [INFO] switch_ivr_async.c:212 Digit parser HTTAPI: Setting realm to 'default' 2013-08-13 11:51:44.715652 [INFO] switch_ivr_async.c:201 Digit parser HTTAPI: Setting terminators for realm 'default' to '#' 2013-08-13 11:51:44.715652 [DEBUG] switch_ivr_play_say.c:1315 Codec Activated L16@8000hz 1 channels 20ms 2013-08-13 11:51:46.675626 [DEBUG] switch_ivr_play_say.c:1703 done playing file /u/freeswitch/sounds/en/us/callie/ivr/ivr-please_enter_the_number_where_we_can_reach_you.wav 2013-08-13 11:51:47.615624 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 1:800 2013-08-13 11:51:47.615624 [DEBUG] switch_channel.c:471 RECV DTMF 1:800 2013-08-13 11:51:48.255615 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 2:800 2013-08-13 11:51:48.255615 [DEBUG] switch_channel.c:471 RECV DTMF 2:800 2013-08-13 11:51:48.975626 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 3:800 2013-08-13 11:51:48.975626 [DEBUG] switch_channel.c:471 RECV DTMF 3:800 2013-08-13 11:51:49.735623 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 4:800 2013-08-13 11:51:49.735623 [DEBUG] switch_channel.c:471 RECV DTMF 4:800 2013-08-13 11:51:50.355645 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 5:800 2013-08-13 11:51:50.355645 [DEBUG] switch_channel.c:471 RECV DTMF 5:800 2013-08-13 11:51:51.035623 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 6:800 2013-08-13 11:51:51.035623 [DEBUG] switch_channel.c:471 RECV DTMF 6:800 2013-08-13 11:51:51.755623 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 7:800 2013-08-13 11:51:51.755623 [DEBUG] switch_channel.c:471 RECV DTMF 7:800 2013-08-13 11:51:52.455625 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 8:800 2013-08-13 11:51:52.455625 [DEBUG] switch_channel.c:471 RECV DTMF 8:800 2013-08-13 11:51:53.235631 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 9:880 2013-08-13 11:51:53.235631 [DEBUG] switch_channel.c:471 RECV DTMF 9:880 2013-08-13 11:51:53.995624 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 0:880 2013-08-13 11:51:53.995624 [DEBUG] switch_channel.c:471 RECV DTMF 0:880 2013-08-13 11:51:54.695615 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 7:800 2013-08-13 11:51:54.695615 [DEBUG] switch_channel.c:471 RECV DTMF 7:800 2013-08-13 11:51:55.575615 [DEBUG] switch_rtp.c:3917 RTP RECV DTMF 8:800 2013-08-13 11:51:55.575615 [DEBUG] switch_channel.c:471 RECV DTMF 8:800 2013-08-13 11:51:56.675645 [DEBUG] mod_httapi.c:1217 Process Tag: [continue] According to Freeswitch wiki, t hese are the definitions of input-timeout and digit-timeout attributes: digit-timeout : Pause to wait for digits after file plays (when input bindings are present) input-timeout : Pause to wait for more digits in a multi-digit input However the input-timeout acts as the maximum time the user has to enter the total input digits after the end of playback while the digit-timeout seems to be "ignored" when input-timeout is present. What am I missing? I'm using the latest stable version of freeswitch (1.2.12). Thank you for your help. Denis Gasparin

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freeswitch and mediaproxy-ng
by jay binks 13 Aug '13

13 Aug '13

So.. when I stumbled across this : https://github.com/sipwise/mediaproxy-ng I was quite interested in the kernel based RTP proxying and the efficiency gained by reducing interrupts. has anyone ever investigated the integration of this with Freeswitch ? Im not even sure if FS's RTP stack is modular, but I thought id ask the questions. -- Sincerely Jay

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problem with menuexecapp
by covici＠ccs.covici.com 13 Aug '13

13 Aug '13

Hi. I am having a problem with the freeswitch ivr. If someone types a number of digits equal to the maximum length, and then types a #, the # is saved and passed to wherever the transfer or whatever the menuexecapp wants to go. Now I have an app which responds to # by hanging up, so this makes it difficult. Is there a way that either the app could flush the dtmf queue when it comes in, or could (and this would be much better) the # be ignored when the maximum length is reached? I thought this was the way it was supposed to work, anyway. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Securing FS
by Kaleem 13 Aug '13

13 Aug '13

Hi all, Just a quick question I need to secure my FS, I can think of 2 ways. Please advise of best option and also how to setup 2nd for learning 1: have FS behind IPTables FW and use access list on which ip can talk to FS and only allow providers. 2: setup another FS as SBC and have it that only internal FS IP can generate calls (no external users, if Ever get any, probably get them to VPN in. Thanks in advance Regards, K Sent from my iPhone

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