May 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Can't get rtcp to work...
by Allan Kristensen 23 May '16

23 May '16

Hello, I've recently moved from Asterisk to Freeswitch and I'm trying to setup a FS on a cloud solution. I have mixed results from previous projects with Asterisk on cloud servers, but this one is just going to proxy media, so I'll give it a shot ;-) I'm going to log the rtcp stats into the cdr, so I can be sure things are ok, but most of the rtcp related stats all contain "0" and mos score 4.5 (even if I randomly drop rtp packets for a couple of seconds to simulate packet loss... not good). The client uses an asterisk, which is sending rtcp reciever reports to FS, however FS is not listening on the rtcp port at all (rtp port +1) and server just sends back icmp unreach. (it's not muxed into the rtp) Also FS is not sending rtcp packets, from what I can see, so I suspect rtcp is somehow not enabled for the channel / session at all. I've added the "rtcp-audio-interval-msec" to the sip profile, so it should work, but it's not.. I've checked the source and even enabled the RTCP_DEBUG in the source to get a clue, but no help there. Asterisk is not actually adding anything about rtcp to the sdp, but is that really needed to get the rtcp working? I'm not using ICE or anything, just a simple setup for this. Is there any way to see if FS is disabling the rtcp feature and if so, why? Any ideas on what can be done to get things working? ;-) Thank you and have a excellent day! / Allan

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Websocket Lost 0 sleep: 1000msec
by Murugan Pandian 22 May '16

22 May '16

Hai, My verto client(Chrome browser) Call drop automatically after 1000msec,In chrome console it shows following error jquery.jsonrpcclient.js:325 Websocket Lost 0 sleep: 1000msec.

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Ubuntu + FS
by Gregor Nanger 20 May '16

20 May '16

I am trying to install FS on Ubuntu. When I set make, I get following error: make: *** [libfreeswitch_libyuv_la-row_common.lo] Error 1 Any advice what I am doing wrong? Best regards, Gregor

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How to preserve To: header between two legs of the call
by Victor Chukalovskiy 20 May '16

20 May '16

Good day, The scenario is that FS receives INVITE with: From = A, To = B, request line = C FS on an outgoing leg will change it so that To header inherits the value from the request line (essentially, destination number): From = A, To = C, request line = C What should I do in the dial-plan to prevent FS from putting destination number "C" in the To header, and instead retain the value "B" from the incoming leg? Thank you! -Victor

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freeswitch-codec-vpx-1.6.8 not in files directory
by Richard Screene 20 May '16

20 May '16

Hello, Is there any reason why freeswitch-codec-vpx-1.6.8-1.el7.centos.x86_64.rpm is not present in the http://files.freeswitch.org/yum_1.6/7/x86_64/ <http://files.freeswitch.org/yum_1.6/7/x86_64/> directory? (versions up to and including 1.6.6 are present) I understand that the underlying library has changed but will still need a FreeSWITCH wrapper don’t we? Or, am I missing something obvious? Thanks, Richard

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Re: [Freeswitch-users] Lua scripts
by Jude Mukundane 20 May '16

20 May '16

Hello Joselyne, try this. it works for me: *session:answer()session:setHangupHook("onHangup","rien")if (session:ready()) then* just replace your first two lines with the above and try. good luck! Jude Mukundane On Tue, May 17, 2016 at 4:58 PM, Nduwayezu, Joselyne <35633(a)heb.be> wrote: > 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 > _________________________________________________________________________ > 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 >

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database disk image is malformed
by Steven Schoch 20 May '16

20 May '16

I'm moving a FS installation from one Linux machine to another, and I'm getting this error: 2016-05-12 16:10:48.243672 [ERR] switch_core_sqldb.c:1197 SQL ERR: [select call_id,sip_user,sip_host,contact,status,rpid,expires,user_agent,server_user,server_host,profile_name,network_ip, network_port,0,sip_realm from sip_registrations where expires > 0 and expires <= 1463094648] database disk image is malformed The trouble is that it doesn't say which database disk image, since there are several (that start with "sofia_" that have that table. How do I find the broken one, or figure out what exactly the problem is? -- Steve

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API question
by Chris Mandra 20 May '16

20 May '16

Hi guys - Happy Friday - I have a question: My main question is how can I generate a custom reply event for an api call with custom body and headers for an API method defined in a custom module? - is the only mechanism using this stream object? or can you directly create the reply event. and if so, how does the caller receive the reply event and know it corresponds to the API call made? I want to provide a response to an API command defined in a module via: #define SWITCH_STANDARD_API(name) static switch_status_t name (_In_opt_z_ const char *cmd, _In_opt_ switch_core_session_t *session, _In_ switch_stream_handle_t *stream) and my understanding (which may be incorrect) is that the stream object can be used to provide a response. When the API is called, an event is automatically generated which uses the stream object. Is there a way to directly modify the event that is generated in response to an API call? How can I modify the headers and body of the event generated from an API call directly? it looks like the stream data is put in the body of the response as per: switch_event_add_body(event, "%s", reply); the relevant code is in api_exec static void *SWITCH_THREAD_FUNC api_exec(switch_thread_t *thread, void *obj) { struct api_command_struct *acs = (struct api_command_struct *) obj; switch_stream_handle_t stream = { 0 }; char *reply, *freply = NULL; switch_status_t status; switch_mutex_lock(globals.listener_mutex); prefs.threads++; switch_mutex_unlock(globals.listener_mutex); if (!acs) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Internal error.\n"); goto cleanup; } if (!acs->listener || !switch_test_flag(acs->listener, LFLAG_RUNNING) || !acs->listener->rwlock || switch_thread_rwlock_tryrdlock(acs->listener->rwlock) != SWITCH_STATUS_SUCCESS) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "Error! cannot get read lock.\n"); acs->ack = -1; goto done; } acs->ack = 1; SWITCH_STANDARD_STREAM(stream); if (acs->console_execute) { if ((status = switch_console_execute(acs->api_cmd, 0, &stream)) != SWITCH_STATUS_SUCCESS) { stream.write_function(&stream, "-ERR %s Command not found!\n", acs->api_cmd); } } else { status = switch_api_execute(acs->api_cmd, acs->arg, NULL, &stream); } if (status == SWITCH_STATUS_SUCCESS) { reply = stream.data; } else { freply = switch_mprintf("-ERR %s Command not found!\n", acs->api_cmd); reply = freply; } if (!reply) { reply = "Command returned no output!"; } if (acs->bg) { switch_event_t *event; if (switch_event_create(&event, SWITCH_EVENT_BACKGROUND_JOB) == SWITCH_STATUS_SUCCESS) { switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "Job-UUID", acs->uuid_str); switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "Job-Command", acs->api_cmd); if (acs->arg) { switch_event_add_header_string(event, SWITCH_STACK_BOTTOM, "Job-Command-Arg", acs->arg); } switch_event_add_body(event, "%s", reply); switch_event_fire(&event); } } else { switch_size_t rlen, blen; char buf[1024] = ""; if (!(rlen = strlen(reply))) { reply = "-ERR no reply\n"; rlen = strlen(reply); } switch_snprintf(buf, sizeof(buf), "Content-Type: api/response\nContent-Length: %" SWITCH_SSIZE_T_FMT "\n\n", rlen); blen = strlen(buf); switch_socket_send(acs->listener->sock, buf, &blen); switch_socket_send(acs->listener->sock, reply, &rlen); } switch_safe_free(stream.data); switch_safe_free(freply); if (acs->listener->rwlock) { switch_thread_rwlock_unlock(acs->listener->rwlock); } Thanks, as ever, for your help, chris

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ASR\ACD stats in freeswitch
by k.presler＠megafit.su 20 May '16

20 May '16

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G.729 module uses two channels (licenses) for only one call in
by Fernando Hernandez 20 May '16

20 May '16

Hi All, Briefly I introduce our system: IPPBX device <-> FreeSwitch <-> bigbluebutton platform we have two big issues related to FreeSwitch G.729 commercial module. 1. When we receive a call in (from outside IPPBX) to join a meeting we can see in FreeSwitch logs two encoding/decoding outputs, resulting two channels in use for only one call in. At the beginning we thought it was related to an IVR message to ask the user for the meetingID and password, but we "harcoded" that info in FreeSwitch files to directly forward the user into the meeting and we had the same problem. For call out (from BBB to outside phone number) only one channel is used. 2. We have four G.729 channels, For instance, if we run 1 call out (1 G.729 channel in use) and 2 calls in (another 4 channels), the second call in will establish the connection but a weird "KAKAKA" sound will appear. We can see this error message in the log: - [ERR] switch_core_io.c:697 Codec RAW Signed Linear (16 bit) decoder error! [1] We now it happens because we only have four channels and we are trying to use the fifth one, but if we do the same with calls out, 5 calls out, the fifth one will just be closed as soon as the user pick up the phone and no weird sound will happen. Thanks and best regards. -- *Fernando Hernandez Esguevillas* *Cloudpe Corp.*

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