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Freeswitch crashed daily
by Achintha 27 Oct '17

27 Oct '17

hi all Our freeswitch (Version 1.6.18-35-6e79667~64bit) handled around 7000 calls per day. Its coredb configured with postgresql (bdr) database (freeswitch DB) and following modules. mod_xml_curl : Dynamic Dialplans (from rest Service) mod_json_cdr : for CDR (from rest Service) and one module developed by our self (mod_ards) freeswitch crashed daily. here i have upload a core dump trace on pastbin https://pastebin.freeswitch.org/view/bf2b479d -- Best Regards.. Achintha Udukumbura

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call forward failing with enterprise originates
by sangeeth＠xoyal.com 26 Oct '17

26 Oct '17

Hi We have an extension(1000) registered at an IP phone which have a call forwarding to an external number. and the other extension(1002) on other ip phone. when i make an enterprise originate to two destinations. the 1000 gives me infinite invite requests instead of forwarding. and stop ringing. i refered to the Jira https://freeswitch.org/jira/browse/FS-6814 but looking if it had been resolved yet . BR, sangeeth kumar.

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mod_conference video_no_video_avatar_png flag with video-mode passthrough
by Richard Screene 26 Oct '17

26 Oct '17

Hello, In mod_conference, is it possible to use the video_no_video_avatar_png flag with video-mode passthrough? I can’t get it working but that could indicate that either (a) its not possible or (b) I’ve done something wrong. What I’m trying to achieve is: in a conference with both audio/video and audio only members the video stream follows the active speaker. However, if one of the audio only participants is speaking then a placeholder image is transmitted to the audio/video members instead. Obviously, I’d prefer to reduce the amount of transcoding/canvas manipulation on the server hence the use of the passthrough video-mode. If there any better ways of achieving this then please let me know! Many thanks for reading this, Richard

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Share Screen stop work after ~20 seconds (FS 1.6.18/1.6.19/master)
by José Lopes 26 Oct '17

26 Oct '17

Hello, I have a Freeswitch WebRTC SIP connected with a Freeswitch MCU. When i use Video Conference on WebRTC, when I share the Screen, the screen is shared, but after 20 seconds the screen freeze and it stops to be presented. The video call of Screen Share is still active. I have this situation after upgrade to FreeSwitch 1.6.18. On Freeswitch 1.6.17 this situation does not happen. I tried also on Freeswitch 1.6.19 and master and this situation still happens. There is any configuration that I need to use on the new versions related to this scenario, so this will not happen? Additional Notes: * On Freeswitch MCU, I made a change that when it is video from Screen Share, the conference member is with flags : mute and deaf. After 20 seconds the video freeze , but after some seconds the screen share start working again. * I tried a Freeswitch WebRTC with MCU, and this situation doesn't happen. Thanks in advance Best regards, *José Lopes*

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is it possible to use mod_callcenter queues without answering the line
by Khalil Khamlichi 26 Oct '17

26 Oct '17

Hi, is it possible to use mod_callcenter queues without answering the line, that is the customer is held in the queue and hearing ringing (without the call being answered). In fact this is a mandatory requirement when calls have a cost and we don't want to charge customer for waiting time but only for talk time. Thanks in advance. Khalil

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[ERR] mod_lua.cpp:103 attempt to call a nil value in recordFile function
by Akshaya j 26 Oct '17

26 Oct '17

Hi team, Spec: OS - Debian FS server Version -1.6.15 I use Zoiper clients (in Ubuntu machine) directly connected to FS server. I'm supposed to write voicemail scripts where i get the call and bridge to Avaya gateway if there is no response i load a lua script. The lua script contains........... if session:ready() then session:speak("Record your message now, to end recording press pound"); session:setInputCallback('onInput',''); test = session:recordFile(recording_filename, 60, 30, 5) session:consoleLog("info", "session:recordFile() = " .. test ) And my Input CallBack function has got..... function onInput(session, input_type, data) session:consoleLog("info", "inside" ) if input_type == "dtmf" then return "break" end end Now my problem is I dont see any log which is inside the callback function.I can record the voice and getting saved but there is an error.... [DEBUG] switch_rtp.c:7255 RTP RECV DTMF #:1280 [INFO] switch_channel.c:515 RECV DTMF #:1280 [ERR] mod_lua.cpp:103 attempt to call a nil value stack traceback: [C]: in function 'recordFile' /usr/local/freeswitch/scripts/test.lua:19: in main chunk I can witness that the session is receiving the DTMF signal but then the function is not getting executed. I tried commenting the func call of callback and I can witness the DTMF is not sensed and hence the record termination happens only after silence_threshold. I badly need to process the keypress DTMF for more options without console err. NOTE: instead of recordFile() I tried speak() even by then i get the same error "attempt to call a nil value" in speak function. Please save me FS.!

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outbound route question
by alex 26 Oct '17

26 Oct '17

I create two sip extension 1000 and 1001,i have to calls 1000 to 1001. This is my outbount route: <include> <context "name=default"> <extension name="outgoing"> <condition field="destination_number" expression="^(100)$"> <action application="bridge" data="sofia/freetdm/wp1/a/$1"/> </condition> </extension> </context> </include> But it didn't take the route. Why is that? what should i do? This are CLI log: 2017-10-13 05:02:36.296708 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1000(a)192.168.1.70 [b12d303a-1f58-4731-b568-78e2b256c28b] 2017-10-13 05:02:36.316705 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1234 in context public 2017-10-13 05:02:36.316705 [NOTICE] switch_ivr.c:2201 Transfer sofia/internal/1000(a)192.168.1.70 to XML[1001@default] 2017-10-13 05:02:36.316705 [INFO] mod_dialplan_xml.c:637 Processing 1000 <1000>->1001 in context default 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 Open /usr/local/freeswitch/conf/vars.xml and change the default_password. 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 Once changed type 'reloadxml' at the console. 2017-10-13 05:02:36.316705 [CRIT] mod_dptools.c:1782 WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING WARNING 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *1 execute_extension::dx XML features 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1000.2017-10-13-05-02-46.wav 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *3 execute_extension::cf XML features 2017-10-13 05:02:46.336708 [INFO] switch_ivr_async.c:4273 Bound B-Leg: *4 execute_extension::att_xfer XML features 2017-10-13 05:02:46.336708 [NOTICE] switch_channel.c:1104 New Channel sofia/internal/1001@192.168.1.69:5060 [4e2e818f-98f5-461b-ad4c-ad34eb933191] 2017-10-13 05:02:46.336708 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2017-10-13 05:02:46.396708 [NOTICE] sofia.c:7391 Ring-Ready sofia/internal/1001@192.168.1.69:5060! 2017-10-13 05:02:46.396708 [INFO] switch_ivr_originate.c:1220 Sending early media 2017-10-13 05:02:46.416708 [NOTICE] sofia_media.c:92 Pre-Answer sofia/internal/1000(a)192.168.1.70! 2017-10-13 05:02:50.016708 [NOTICE] sofia.c:8474 Hangup sofia/internal/1000(a)192.168.1.70 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2017-10-13 05:02:50.016708 [NOTICE] switch_ivr_originate.c:3629 Hangup sofia/internal/1001@192.168.1.69:5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2017-10-13 05:02:50.036718 [NOTICE] switch_ivr_originate.c:2868 Cannot create outgoing channel of type [user] cause: [ORIGINATOR_CANCEL] 2017-10-13 05:02:50.036718 [INFO] mod_dptools.c:3508 Originate Failed. Cause: ORIGINATOR_CANCEL 2017-10-13 05:02:50.036718 [NOTICE] switch_core_session.c:1731 Session 2 (sofia/internal/1001@192.168.1.69:5060) Ended 2017-10-13 05:02:50.036718 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1001@192.168.1.69:5060 [CS_DESTROY] 2017-10-13 05:02:50.056713 [NOTICE] switch_core_session.c:1731 Session 1 (sofia/internal/1000(a)192.168.1.70) Ended 2017-10-13 05:02:50.056713 [NOTICE] switch_core_session.c:1735 Close Channel sofia/internal/1000(a)192.168.1.70 [CS_DESTROY] Thanks for answer.

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ciphers not available for verto wss (neither wss on sip profile)
by GM phy 26 Oct '17

26 Oct '17

Hello, im having this issue and by now i found no way to figure it out. enabling tls on internal, if a exec openssl s_client -connect 10.10.10.9:5061 -tls1_2 Connection is established and server returns my certificate and this information -- New, TLSv1/SSLv3, Cipher is ECDHE-ECDSA-AES256-GCM-SHA384 Server public key is 256 bit Secure Renegotiation IS supported Compression: NONE Expansion: NONE No ALPN negotiated SSL-Session: Protocol : TLSv1.2 Cipher : ECDHE-ECDSA-AES256-GCM-SHA384 ... -- When i run the same comand for port 7443 (the wss binding on internal profile) or port 8082 (verto wss) i get (same on both cases) -- New, TLSv1/SSLv3, Cipher is ECDH-RSA-AES256-GCM-SHA384 Server public key is 256 bit Secure Renegotiation IS supported Compression: NONE Expansion: NONE No ALPN negotiated SSL-Session: Protocol : TLSv1.2 Cipher : ECDH-RSA-AES256-GCM-SHA384 ... -- This causes that all of the webrtc phones we're using (some with sipjs, some with verto) are unable to establish the wss socket in order to register and make calls. On the browser console i get Protocol mismatch, and on a pcap, server rejects the ssl handshake with code 40. I've been working with the code on mod_verto.c and ws.c with no luck. basically what im looking for is for verto and/or wss on internal profile, to support Cipher is ECDHE-ECDSA-AES256-GCM-SHA384 (or ECDHE-ECDSA) in order to webrtc can connect to fs. any help will be really appreciated. thanks in advance

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Control register behavior via a lua script when REGISTER?
by Charles Yu 26 Oct '17

26 Oct '17

Hi, Is there a way to control register behavior via a lua script when receiving sip REGISTER ? Such like: I can reject this REGISTER if it does not match some condition. Trigger a sofia::register_failure event? Any ideas?

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Adding authorization to SIP messaging
by Marko Mrvelj 25 Oct '17

25 Oct '17

Hi all, We are experimenting with chatplan, in order to get basic chat functionality. Messages do work as expected, but there is a problem that message can be submitted to server without any authorization. Could somebody tell me if it is possible to restrict messages to only authorized users? How can this be done? Thanks in advance, Marko

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