- FreeSWITCH-users - FreeSWITCH Mailing Lists

Need help with ESL using Perl
by Mo Rajib 06 Jul '14

06 Jul '14

Hi, I am trying to use ESL with Perl but I am having difficulties receiving events from the FS. I tested a simple Perl script to connect to FS (inbound connection) and execute an API command (show status). I can connect successfully but not receiving the response of the API command. I have listed the simple Perl script and the error I receive. However I can telnet and execute the same API command successfully using telnet command (> telnet 127.0.0.1 8021). I am running FS 1.4.6 on Debian release 7.5 32-bit. Any suggestion on how to troubleshoot this is appreciated. Here is the Perl script I am using to test ESL inbound connection. # perl test.pl $VAR1 = undef; Can't call method "serialize" on an undefined value at test1.pl line 29. ---------------------------------------------------------------------------- ------------ use Data::Dumper; require ESL; my $host = "localhost"; my $port = "8021"; my $pass = "ClueCon"; # Establish an event socket connection to FreeSWITCH my $con = new ESL::ESLconnection($host, $port, $pass); if (! $con) { die "Unable to establish connection to $host:$port\n"; } my $cmd = "sofia status"; my $args = ""; my $e = $con->api($cmd, $args); print Dumper $e; print $e->serialize("plain"); # disconnect from FreeSWITCH $con->disconnect(); --- This email is free from viruses and malware because avast! Antivirus protection is active. http://www.avast.com

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The best way to IVR talks the Address based on CallerID Number
by Normando Hall 06 Jul '14

06 Jul '14

Hello. I am working on an IVR based on FS. This is what I want to do: I have a huge database with a few fields: 'phone number' - 'address street' - 'address number' - 'address building' - 'address floor' - etc. When I receive a call I will query the CallerID into de 'phone number' field. If found it, the phrase macro says something like "Hello, your address is xxxx, xxx" If the caller confirm pressing 1, then we will use this query results to activate other systems in the server, such as storing or trigger an event. I was recorder each street name and numers with an annoucer (broadcaster) into a separate wav files. I was reading about *mod_cidlookup*, but I don't know if this is the better way to achieve this. Another way I see are related to*mod_db*, *mod_dptools*, *mod_easyroute*, etc. Thanks for your advices. Normando Hall

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Mod_snmp problem.
by Volkan Oransoy 06 Jul '14

06 Jul '14

Hi all, I have updated my FS installation to 1.2.24. Everthing compiled and installed fine. But snmp module gives error while I try to load. 2014-07-06 20:01:10.480236 [CRIT] switch_loadable_module.c:1391 Error Loading module /usr/local/freeswitch/mod/mod_snmp.so **/usr/lib/libnetsnmpagent.so.15: undefined symbol: netsnmp_register_null_context** This is an Ubuntu 12.04LTS box. Does anyone have an idea about the reason? Thanks. Volkan Oransoy

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mod managed. Memory leak FS 1.4.6
by Andrew 06 Jul '14

06 Jul '14

Hi, I was using 1.4.6 and was having a hard time with running out of memory. I hit 50 gb then the switch stopped working correctly. I was ready to give up until I tried 1.2. Not sure what happen but The new mod managed is missing some of the functions (I will put a jar in soon) and memory leak. I have not spent much time on hunting down the issue but it's not the .net c# code as I ran a memory profile to find the issue and it said it was not managed code causing the issue. When the last swig was done I think it caused more issues than good. Not knowing much about swig. If I was to open a jar what info would be needed to show the leak? Program never crashes. Andrew

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ESL Synchronous Playback
by Yisroel M. Olewski 06 Jul '14

06 Jul '14

Hi everyone I'm using eslconnection.execute to playback a music file, but the code just sends the command and continues running This is not good for me, since I need to block code execution until the file finishes Eslconnection.api is supposed to be sync but doesn't have a uuid parameter Actually execute itself is also supposed to be sync How can this be done? I've seen a few questions about it, but no answers Thanks and All the best! Sincerely, Yisroel M. Olewski CEO Machshevet.com +972 (52) 764-9971 Websites . CRM . ERP . Telephony . Inventory . Content Management

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TLS, SRTP configuration on the FreeSWITCH and Groundwire ?
by Herold Park 05 Jul '14

05 Jul '14

How can i configure the FreeSWITH and Softphone, which is Groundwire? Certificate and key files made as following wiki on the FreeSWITCH server. It's created under conf/ssl directory. And a account create, it work fine with udp mode. But it doesn't register tls mode on the Outbound Proxy. What's my mistakes ? please help me ALL. -- 세상은 아름답다.

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SIP ACCOUNT SETTINGS FOR CISCO ROUTERS WITH FXS PORTS
by Matthew Halos 05 Jul '14

05 Jul '14

Hello to All! I am newbie to FreeSWITCH. Anybody knows how to set an SIP account on a Cisco router with FXS Ports? I am planning to use POTS telephone lines by the use of the FXS Ports that communicates to the FreeSWITCH server, in order to contact the registered users in the FreeSWITCH server. Thanks! MATTHEW HALOS

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handling signalling externally, RTP via freeswitch
by Russell Treleaven 04 Jul '14

04 Jul '14

here is a quick and dirty solution that I have not tried or even thought about in great detail. A leg is a sipp script that bridges a call through FS to your sip B leg. the sipp script sdp actually points to your custom endpoint. Does that make sense? Sincerely, Russell Treleaven rtreleaven(a)bunnykick.ca

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mod_spandsp: Inband DTMF not processed
by Nagalenoj H. 04 Jul '14

04 Jul '14

Hi, I am facing an issue with using Inband DTMFs. Inband DTMFs are getting detected in switch_rtp.c but then it is not processed further by mod_spandsp_dsp.c. This issue is inconsistent, for few calls it is working and not working for few other calls. For a working call, I'm getting the logs as below, 2014-06-19 21:24:40.334513 [DEBUG] switch_ivr_play_say.c:1682 done playing file file_string:///usr/local/freeswitch/sounds/music/switch/2539_SB_welcome_switch.wav!/usr/local/freeswitch/sounds/music/switch/1sec_silence.wav!/usr/local/freeswitch/sounds/music/switch/SB_landing_IVR.wav 2014-06-19 21:24:44.114522 [DEBUG] switch_rtp.c:3793 RTP RECV DTMF 3:1280 2014-06-19 21:24:44.694512 [DEBUG] mod_spandsp_dsp.c:372 DTMF BEGIN DETECTED: [3] 2014-06-19 21:24:44.854513 [DEBUG] mod_spandsp_dsp.c:384 DTMF END DETECTED: [3], duration = 165 ms 2014-06-19 21:24:45.434531 [DEBUG] switch_rtp.c:3793 RTP RECV DTMF 9:1280 2014-06-19 21:24:45.994513 [DEBUG] mod_spandsp_dsp.c:372 DTMF BEGIN DETECTED: [9] 2014-06-19 21:24:46.154514 [DEBUG] mod_spandsp_dsp.c:384 DTMF END DETECTED: [9], duration = 165 ms 2014-06-19 21:24:47.434530 [DEBUG] switch_rtp.c:3793 RTP RECV DTMF 2:1280 2014-06-19 21:24:48.034523 [DEBUG] mod_spandsp_dsp.c:372 DTMF BEGIN DETECTED: [2] 2014-06-19 21:24:48.034523 [CRIT] sb_main.js:390 digits pressed after collect input: 392 On the failure calls, I'm getting only the RTP RECV DTMF log and not the DTMF BEGIN and DTMF END DETECTED log messages and because of this, dialplan application is not receiving any DTMFs. I've attached the pcap file and full fs log for failure call. What could be the cause of this inconsistent behavior? Kindly help me to rectify this. -- Regards, Nagalenoj H.

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Transcoding
by Akib Sayyed 04 Jul '14

04 Jul '14

Dear List I am thinking of transcoding Speex to GSM or Speex to G729 which will be eating less resource -- Akib Sayyed Matrix-Shell akibsayyed(a)gmail.com akibsayyed(a)matrixshell.com Mob:- +91-966-514-2243

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