May 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

wss and nginx
by Philipp Zeitschel 18 May '16

18 May '16

Hi, i've googled for hours but i don't get it to work i have a nginx proxy at my public ip and it forwards the traffic through 10.1.71.5 to my freeswitch box at 10.1.71.10 i've set ext-rtp-ip and ext-sip-ip to my external IP adress and created the acl.conf.xml in autoload_configs with following content: <configuration name="acl.conf" description="Network Lists"> <network-lists> <list name="lan" default="deny"> <node type="allow" cidr="10.1.1.1/32" /> </list> </network-lists> </configuration> In fs_cli i get the message 2016-05-18 08:13:43.245201 [WARNING] switch_core_media.c:3458 NO candidate ACL defined, Defaulting to wan.auto So i think the acl is working. My nginx config looks like this: server { listen 0.0.0.0:443; server_name hostname; keepalive_timeout 70; ssl on; ssl_certificate /etc/ssl/my.ct; ssl_certificate_key /etc/ssl/my.key; ssl_dhparam /etc/ssl/dhparam.pem; ssl_protocols TLSv1 TLSv1.1 TLSv1.2; ssl_prefer_server_ciphers on; ssl_session_cache shared:SSL:10m; ssl_ciphers 'EECDH+AESGCM:EDH+AESGCM:AES256+EECDH:AES256+EDH'; location / { proxy_set_header Upgrade $http_upgrade; proxy_set_header Connection $connection_upgrade; proxy_set_header X-Real-IP $remote_addr; proxy_set_header Host $host; proxy_set_header X-Forwarded-For $proxy_add_x_forwarded_for; proxy_read_timeout 3600; proxy_http_version 1.1; proxy_pass https://10.1.71.10:7443; } } I can login and establish calls but i don't hear voice so i think the rtp packages are not ok Because of that i've enabled sip trace and get this: send 1668 bytes to wss/[10.1.71.5]:32890 at 08:13:43.266640: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/WSS keenaijeb5h6.invalid;branch=z9hG4bK779005;received=10.1.71.5;rport=32890 From: <sip:user@domain>;tag=n8m38teo73 To: <sip:conf@domain>;tag=gX23c0rey78vH Call-ID: toghv9ifikpaeib85mh8 CSeq: 6099 INVITE Contact: <sip:conf@138.201.x.x:5080;transport=udp> User-Agent: FreeSWITCH-mod_sofia/1.6.8-15-99de0ad~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 901 Remote-Party-ID: "conf" <sip:conf@domain>;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 1463528835 1463528836 IN IP4 10.1.71.10 s=FreeSWITCH c=IN IP4 10.1.71.10 t=0 0 a=msid-semantic: WMS 7y98fmZXWMapBpCUcRfJdKQGM9JU5t83 a=end-of-candidates m=audio 30388 UDP/TLS/RTP/SAVPF 111 101 106 a=rtpmap:111 opus/48000/2 a=fmtp:111 useinbandfec=1; minptime=10 a=rtpmap:101 telephone-event/8000 a=rtpmap:106 CN/8000 a=ptime:20 a=fingerprint:sha-256 5E:AE:56:B1:0B:39:38:7B:00:6F:8A:84:F6:0F:DC:86:8D:24:9A:7D:4D:59:F8:95:8F:CC:ED:A2:6F:90:84:2C a=setup:active a=rtcp-mux a=rtcp:30388 IN IP4 10.1.71.10 a=ice-ufrag:wLDAlj6YjGJfZH8H a=ice-pwd:F0vQH6N0H0LyFhs1SVr45oSe a=candidate:0546395930 1 udp 659136 10.1.71.10 30388 typ host generation 0 a=ssrc:2671785415 cname:R71XkQK7tLmU0PF3 a=ssrc:2671785415 msid:7y98fmZXWMapBpCUcRfJdKQGM9JU5t83 a0 a=ssrc:2671785415 mslabel:7y98fmZXWMapBpCUcRfJdKQGM9JU5t83 a=ssrc:2671785415 label:7y98fmZXWMapBpCUcRfJdKQGM9JU5t83a0 ------------------------------------------------------------------------ My freeswitch (running on debian jessie) version is FreeSWITCH Version 1.6.8-15-99de0ad~64bit (-15-99de0ad 64bit) What am i doing wrong? What else can i do to force freeswitch to use my external IP in the RTP Packages? Regards Philipp

1 0

sounds/music directory reorganization
by Steven Schoch 17 May '16

17 May '16

While I'm troubleshooting my major FreeSWITCH problems, I found a couple of minor ones. This one is with the debian/jessie repository. The default autoload_configs/local_stream.conf.xml file specifies the location of the default music stream files. This default configuration file is found in /usr/share/freeswitch/conf/*/autoload_configs example (default) config files. The directory specified is $${sounds_dir}/music/8000, $${sounds_dir}/music/16000, etc. However, the music files now seem to be installed in /usr/share/freeswitch/sounds/music/default/8000, etc. In other words, there is an extra "default" directory that wasn't there in an earlier (repository) version. This results in error messages that say: 2016-05-13 08:25:47.280695 [CONSOLE] mod_local_stream.c:282 Can't open directory: /usr/share/freeswitch/sounds/music/8000 So what's the intent here? Should the (default) config files be changed, or should the music files be moved out of the "default" directory? -- Steve

3 3

Thursday FreeSWITCH Bug Hunt
by Ítalo Rossi 17 May '16

17 May '16

FreeSWITCHers, Join us Thursday 2PM CST for the Thursday FreeSWITCH Bug Hunt! Where? https://conference.freeswitch.org/vc/#/?autocall=888 What? FreeSWITCH Bug Hunt, Jira Reviews, and General FS Support! Help us help you, Join the Bug Hunt! Ítalo Rossi italo(a)freeswitch.org IRC chat.freenode.net #freeswitch #freeswitch-dev Bugs? https://freeswitch.org/jira Docs? https://freeswitch.org/jira Chat? https://hipchat.freeswitch.org/gUdAgy0m6

1 0

Re: [Freeswitch-users] How to receive incoming calls from Huawei dongle and mod_gsmopen?
by Jude Mukundane 17 May '16

17 May '16

Hello Gabriel, I guess at this point you have 1) Confirmed that the Modem does voice by testing it with some other PNX 2) Confirmed that you see activity in fs_cli when you make calls to the modem - this shows that the modem is well connected to the modem. On ubuntu this is easier to debug - usually it is making sure the modem lines listed under /dev/ttyUSB* correspond to the lines in gsmopen.conf.xml and have read access - even on windows, please make sure these match. it is possible that the modems are on different COM lines to the ones specified in gsmopen.conf.xml, in which case FS will not see your modem. Jude Mukundane On Mon, May 16, 2016 at 9:07 PM, Gabriel MdS <gabriel(a)cefet.net> wrote: > Hi Giovanni, > > Thanks for your response. > > As I reported in the original message, the module mod_gsmopen is installed > and running (auto-loaded). I was able to call others from my softphone > using my modem and mod_gsmopen without any problems. > I just can't get FreeSWITCH to automatically answer the call and forward > it to an IVR menu. It seems like it is not receiving the RING signal or I > am missing some other setting... > > My application is really simple. I have it running on asterix+chan_dongle > on a virtual machine, but just recently found FS which runs on windows, > which is better for me. > > If you have any further advice I'd appreciate it! > > > Thanks! > > -- Gabriel MdS > > 2016-05-16 16:48 GMT-03:00 Giovanni Maruzzelli <gmaruzz(a)gmail.com>: > >> Have you compiled and loaded the module? >> >> Do you have any trace of its presence in freeswitch console? >> >> Also, if you are not an expert I would definitely counseil you not to try >> uncharted things like gsmopen on won with fusionpbx. It's just too complex. >> Do it after you master both fusionpbx and FreeSWITCH. >> >> My two cents :) >> -giovanni >> >> sent from mobile >> cell: +39 347 266 56 18 >> Giovanni Maruzzelli >> OpenTelecom.IT >> Il 16/Mag/2016 21:10, "Gabriel MdS" <gabriel(a)cefet.net> ha scritto: >> >>> Hi, >>> >>> I'm a newbie trying to setup FreeSWITCH with FusionPBX and mod_gsmopen. >>> I'm using the 32bit version in Windows 7, compiled from current source >>> (version 1.7) in Visual Studio 2015 Community. >>> >>> My Huawei dongle is E1756, unlocked and with voice calls enabled. >>> >>> Currently I am able to call my IVR menu directly by dialing from my >>> softphone and I am also able to originate outbound calls after adding a >>> rule using FusionPbx Dialplan Manager. >>> >>> However if I try to call the dongle it keeps ringing, not being answered >>> and no log is displayed in console. My gsmopen.conf.xml file has the >>> following lines: >>> >>> <param name="context" value="public"/> >>> <param name="destination" value="incoming_gsm_call"/> >>> >>> And I've set a new dialplan in fusionpbx rule with: >>> >>> condition > context > public >>> condition > destination_number > incoming_gsm_call >>> action > transfer > 500 XML default (500 being the main IVR menu >>> extension) >>> >>> Am I missing something to get this working or maybe there is some >>> setting wrong with my dongle? >>> >>> >>> Thanks! >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://confluence.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://confluence.freeswitch.org >> http://www.cluecon.com >> >> 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 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://confluence.freeswitch.org > http://www.cluecon.com > > 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 >

2 1

Join us for ClueCon 2016!
by Kathleen King 17 May '16

17 May '16

<https://cluecon.com> [image: https://cluecon.com] <https://cluecon.com>

1 0

Lua scripts
by Nduwayezu, Joselyne 17 May '16

17 May '16

Hello i use this lua script tha i found on freeswitch wiki But when i run it i have the errors: syntax error near unexpected token ; seesion.answer(); and when i remove the ";" i have the error on the while loop, syntax error near the unexpected token ==, while (session:ready() == true. I there something i did not conigure in order the script to work? do we put ";" after an instructtion like session:answer() or not. Thank you or your reply. session:answer(); while (session:ready() == true) do session:setAutoHangup(false); session:set_tts_params("flite", "kal"); session:speak("Welcome. Welcome to the VoIp World!. this is a Blind Users Programing Community. powered by Freeswitch, the free / ultimate PBX. thank to toni!"); session:sleep(100); session:speak("please select an Action."); session:sleep(100); session:speak("to call the conference, press 1"); session:sleep(100); session:speak("to call Freeswitch IVR, press 2"); session:sleep(100); session:speak("to call Voice Mail, press 3"); session:sleep(100); session:speak("for Music on hold, press 4"); session:sleep(100); session:speak("to call me, press 0"); session:sleep(3000); digits = session:getDigits(1, "", 3000); if (digits == "1") then session:execute("transfer","9888"); end if (digits == "2") then session:execute("transfer","5000"); end if (digits == "3") then session:execute("transfer","4000"); end if (digits == "4") then session:execute("transfer","9999"); end if (digits == "0") then session:execute("transfer","voipaware(a)sip.voipuser.org"); end end NDUWAYEZU Joselyne -- Haute École de Bruxelles

2 1

How to receive incoming calls from Huawei dongle and mod_gsmopen?
by Gabriel MdS 16 May '16

16 May '16

Hi, I'm a newbie trying to setup FreeSWITCH with FusionPBX and mod_gsmopen. I'm using the 32bit version in Windows 7, compiled from current source (version 1.7) in Visual Studio 2015 Community. My Huawei dongle is E1756, unlocked and with voice calls enabled. Currently I am able to call my IVR menu directly by dialing from my softphone and I am also able to originate outbound calls after adding a rule using FusionPbx Dialplan Manager. However if I try to call the dongle it keeps ringing, not being answered and no log is displayed in console. My gsmopen.conf.xml file has the following lines: <param name="context" value="public"/> <param name="destination" value="incoming_gsm_call"/> And I've set a new dialplan in fusionpbx rule with: condition > context > public condition > destination_number > incoming_gsm_call action > transfer > 500 XML default (500 being the main IVR menu extension) Am I missing something to get this working or maybe there is some setting wrong with my dongle? Thanks!

3 7

Re- switch_event.c and switch_cpp.cpp [NOTICE] logging. Request for two events to be logged as INFO or DEBUG logtype instead?
by Andrew Keil 16 May '16

16 May '16

To FreeSWITCH Users, I have a simple request in order to improve the ESL events that I capture inside my application. Currently there are two events that are of log type NOTICE that I would really appreciate become log type DEBUG or INFO (if possible). (1) switch_event.c:2113 Event Binding deleted for ..\..\src\switch_cpp.cpp:CUSTOM a. From the C code: switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "Event Binding deleted for %s:%s\n", n->id, switch_event_name(n->event_id)); b. Can the SWITCH_LOG_NOTICE be changed to SWITCH_LOG_DEBUG or SWITCH_LOG_INFO? (2) switch_cpp.cpp:83 bound to CUSTOM .... a. From the C code: switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "bound to %s %s\n", event_name, switch_str_nil(subclass_name)); b. Can the SWITCH_LOG_NOTICE be changed to SWITCH_LOG_DEBUG or SWITCH_LOG_INFO? Please note the line numbers are based on version 1.6.5 of FreeSWITCH, so they may have changed slightly now that the production build is 1.6.8. Let me know if this change is possible or not. Kind Regards, Andrew Keil Visytel Pty Ltd [visytellogo] www.visytel.com<http://www.visytel.com/> Mobile: +61 (0) 429 076 431 Skype: visytel This e-mail message may contain confidential and commercially sensitive information. If you are not the intended recipient, any use, interference with, disclosure or copying of this material is unauthorised and prohibited. If you receive this e-mail message by mistake, please contact Visytel Pty Ltd on +61 429 076 431.

3 2

Re: [Freeswitch-users] external voicemail storage?
by Avi Marcus 16 May '16

16 May '16

After a *bit *more investigating (looking at FusionPBX's extensive lua script for VM) it *looks like *I'd be able to use mod_http_cache <https://freeswitch.org/confluence/display/FREESWITCH/mod_http_cache> as a bridge to Amazon S3 for *both *recording and playback. Am I missing anything? Would using it for all of that mean I'd need to use file format option, which would prevent me from using http_api? Is anyone using this in production? I'm hoping this example <https://freeswitch.org/confluence/display/FREESWITCH/mod_http_cache#mod_htt…>is a typo: > enable-file-formats=false > <action application="playback" data="*http_cache://https*:// > example.com/media/hello_world.wav"/> -Avi Marcus BestFone On Wed, May 11, 2016 at 8:58 AM, Avi Marcus <avi(a)avimarcus.net> wrote: > I checked confluence and I don't see any updates... > > Is it possible to store voicemail messages & greetings somewhere NOT on > the disk? > e.g. S3, in the same database at the voicemail prefs, etc? > > I'm rebuilding an old cluster and this is an issue I didn't find a good > solution to. > It's one of the only things I'd *need *to have a distributed filesystem > for and I haven't had much luck with those. > > Thanks, > -Avi Marcus > BestFone >

4 4

Sending concatenated SMS
by Nandy Dagondon 15 May '16

15 May '16

Hello folks, I can't find any discussion about concatenated SMS or PUD mode (to break the 160 message limit in text-mode sms) in mod_sms or gsmopen. Question, is it possible? If yes, any hint how to do it? Thanks. /Nandy

2 2