August 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] mod_verto and vc are not working
by Alexander Haugg 29 Aug '16

29 Aug '16

I was installing this setup: https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian… The problem is, the client 1008 calls the client 1010 and the client 1010 do answer. The signaling part is connected successfully. But client 1008 and 1010 show the black video field only and voice is don’t working too. In the wireshark trace on the freeswitch host I can see, the freeswitch received from the client 1008 and 1010 the rtp streams, but he send the streams not to the other side. I can’t see any outbound rtp streams. The only one error message in the CLI output is: 2016-08-26 11:37:59.642048 [ERR] switch_core_session.c:511 Could not locate channel type FreeSWITCH Version 1.6.9-16-d574870~64bit (-16-d574870 64bit) on Debian 8.5 The config is default with the adapted IPs What can I do to locate the Problem? Von: freeswitch-users-bounces(a)lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Alexander Haugg Gesendet: Donnerstag, 25. August 2016 14:39 An: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Betreff: Re: [Freeswitch-users] mod_verto and https://webrtc.freeswitch.org/verto/index.html#page-main Thanks a lot, that’s the solution. Von: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Marcel Haldemann Gesendet: Mittwoch, 24. August 2016 18:03 An: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Betreff: Re: [Freeswitch-users] mod_verto and https://webrtc.freeswitch.org/verto/index.html#page-main If u start to play with webrtc and Freeswitch I’d recommend to install FS + cert + verto communicator using the script here: https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian… note: change hostname and ip in the script. Then just go to “https://yourhostname.com/vc” and you should have a working verto server and client. Note: change the default password to get rid of the 10 sec delay or remove the delay in the dial plan. Von: freeswitch-users-bounces(a)lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] Im Auftrag von Alexander Haugg Gesendet: Mittwoch, 24. August 2016 15:24 An: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Betreff: [Freeswitch-users] mod_verto and https://webrtc.freeswitch.org/verto/index.html#page-main Hi, it is my first time with mod_verto. The certificate is generated like described here: https://webrtc.freeswitch.org/verto/index.html#page-main And the ca.crt is installed on the client pc. But the following error message is again: 2016-08-24 15:03:47.044880 [INFO] mod_verto.c:4025 192.168.241.2:43246 Client Connect. 2016-08-24 15:03:47.044880 [INFO] mod_verto.c:1978 192.168.241.2:43246 Starting client thread. 2016-08-24 15:03:47.084876 [DEBUG] mod_verto.c:1825 192.168.241.2:43246 WS SETUP FAILED [] 2016-08-24 15:03:47.084876 [INFO] mod_verto.c:2005 192.168.241.2:43246 Ending client thread. 2016-08-24 15:03:47.084876 [INFO] mod_verto.c:2013 192.168.241.2:43246 Thread ended What can I do? Is the certificate the problem? The Client PC and the Freeswitch PC are in the same Network Cluster. The Freeswitch PC is a Debian 8.5 system. Thanks a lot Alex

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explanation of values from <audio> section in xml_cdr
by Igor Karymov 29 Aug '16

29 Aug '16

Hi there! Can i find somewhere explanation about the meaning of values of <audio> section in xml_cdr? One of my clients having a one-way audio. CDR for this call looks pretty strange: http://pastebin.com/jqhCXKec I guess that freeswitch actually got packets from the client but drop all of them because of something. But not sure if my interpretation is correc

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How to find whether a user is talking from LUA script?
by Yehavi Bourvine 28 Aug '16

28 Aug '16

Hello, I've implemented a LUA script (executed in the background with "luarun") that camps on a busy user, and originates a call when it becomes free. After calling "originate" I need to know whether the call has actually been established; I am querying the FS core database (dialogs table) for this, but looking for a more standard way to do so. What I need is something like "show calls from-extsntion to-extension" which can be executed from LUA and have the result. Thanks! __Yehavi:

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Event Socket Outbound - Playback Sequence
by Luiz Fernando Softov 27 Aug '16

27 Aug '16

Hi all, I'm using FS with Event Socket Outbound. Eg: The user calls to XXX and the calls goes to socket. <extension name="VoiprExt"> <condition> <action application="socket" data="127.0.0.1:3012 async full"/> <action application="hangup"/> </condition> </extension> I receive a call, and play some files (sometimes http:// or shout://). Using command action playback. sendmsg <uuid> call-command: execute execute-app-name: playback execute-app-arg: /sounds/system/chr_ 1 .mp3 The problem is, I'm sending commands to playback files 1, 2, 3 And the system are playing 3, 2, 1 I already tried to use event-lock. S ometimes, if user press any DTMF , I need to stop playback I f I use event-lock : true. I can't do anything until the files are done. The same, using command break (break only show up when files are done). Is there another way, or a correct way to playback multiple sequential files, in order , using application="socket"? This is a BUG? I don't if I made myself clear here. In resume, I can't wait until the current file is done, to send another command to play the next file, since I have one list (options), with several list of files to play (files). And, using asynchronous events (kqueue), the list(s) can change in the middle of process. It's like * Option 1: ** file: wellcome.mp3 ** file: file_ abc 1 .mp3 * Option 2 : ** file: protocol.mp3 ** file: digit _ 8 .mp3 ** file: digit_ 7 .mp3 ** file: digit_ 1 .mp3 ** file: digit_ 2 .mp3 ** file: digit_ 3 .mp3 * Option 3 : ** file: sales .mp3 And, after this, the files to be played are wellcome.mp3 file_ abc 1 .mp3 protocol.mp3 digit _ 8 .mp3 digit_ 7 .mp3 digit_ 1 .mp3 digit_ 2 .mp3 digit_ 3 .mp3 sales .mp3 But, the files are not played in sequence. [DEBUG] switch_channel.c:3770 (sofia/profile_1/6001(a)192.168.142.12) Callstate Change RINGING -> ACTIVE [DEBUG] sofia.c:6858 Channel sofia/profile_1/6001(a)192.168.142.12 entering state [completed][200] [DEBUG] switch_ivr.c:623 sofia/profile_1/6001(a)192.168.142.12 Command Execute playback(/sounds/ivr/ivr_intro.mp3) EXECUTE sofia/profile_1/6001(a)192.168.142.12 playback(/sounds/ivr/ivr_intro.mp3) [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms [DEBUG] sofia.c:6858 Channel sofia/profile_1/6001(a)192.168.142.12 entering state [ready][200] [DEBUG] switch_rtp.c:6706 Correct audio ip/port confirmed. [DEBUG] switch_rtp.c:1845 rtcp_stats_init: audio ssrc[1125545576] base_seq[28137] [DEBUG] switch_ivr_play_say.c:1910 done playing file /sounds/ivr/ivr_intro.mp3 [DEBUG] switch_ivr.c:623 sofia/profile_1/6001(a)192.168.142.12 Command Execute playback(/sounds/ivr/ivr_menu.mp3) EXECUTE sofia/profile_1/6001(a)192.168.142.12 playback(/sounds/ivr/ivr_menu.mp3) [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms [DEBUG] switch_ivr.c:623 sofia/profile_1/6001(a)192.168.142.12 Command Execute playback(/sounds/system/transfer_request.mp3) EXECUTE sofia/profile_1/6001(a)192.168.142.12 playback(/sounds/system/transfer_request.mp3) [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms [DEBUG] switch_ivr.c:623 sofia/profile_1/6001(a)192.168.142.12 Command Execute playback(/sounds/system/chr_1.mp3) EXECUTE sofia/profile_1/6001(a)192.168.142.12 playback(/sounds/system/chr_1.mp3) [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms [DEBUG] switch_ivr.c:623 sofia/profile_1/6001(a)192.168.142.12 Command Execute playback(/sounds/system/chr_2.mp3) EXECUTE sofia/profile_1/6001(a)192.168.142.12 playback(/sounds/system/chr_2.mp3) [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms [DEBUG] switch_ivr.c:623 sofia/profile_1/6001(a)192.168.142.12 Command Execute playback(/sounds/system/chr_3.mp3) EXECUTE sofia/profile_1/6001(a)192.168.142.12 playback(/sounds/system/chr_3.mp3) [DEBUG] switch_ivr_play_say.c:1467 Codec Activated L16@8000hz 1 channels 20ms [DEBUG] switch_ivr_play_say.c:1910 done playing file /sounds/system/chr_3.mp3 [DEBUG] switch_ivr_play_say.c:1910 done playing file /sounds/system/chr_2.mp3 [DEBUG] switch_ivr_play_say.c:1910 done playing file /sounds/system/chr_1.mp3

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Detecting jitter
by Oleg Stolyar 27 Aug '16

27 Aug '16

Hi guys, is there a way in FS to detect and report the amount of jitter on a leg?

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Re: [Freeswitch-users] Detecting jitter
by Stanislav Sinyagin 27 Aug '16

27 Aug '16

Not as far as I know. But you can use tshark for that. At best, you capture the packet dump to a file and process later with tshark in nice priority. On 26 Aug 2016 17:57, "Oleg Stolyar" <olegstolyar(a)gmail.com> wrote: Hi guys, is there a way in FS to detect and report the amount of jitter on a leg? _________________________________________________________________________ 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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presence issue: NOTIFY not sent to presence watcher when presentity registers
by Dave Horton 26 Aug '16

26 Aug '16

I would like some quick feedback on whether I should open a JIRA on this — seems like a bug, but if it is a known issue please let me know. Scenario: User 1 is subscribing to presence events for User 2. Timeline: 1. User 1 registers successfully (user 2 is offline/unregistered at this time). 2. User 1 sends a SUBSCRIBE to FS for User 2. 3. FS sends a 202 Accepted to the subscribe 4. FS sends an initial NOTIFY (application/pidf+xml) indicating that User 2 unregistered (<status><basic>closed</basic></status>) 5. User 2 registers successfully At this point I expect to see FS send a NOTIFY to User 1 with a status update for User 2, showing that s/he is online, but this does not happen. Later, User 2 places an outbound call, and at that point FS _does_ send a NOTIFY with presence update (at both the start and end of the call, with correct information). But it should have sent a NOTIFY earlier when User 2 registers. Is this a known issue? Should I open a JIRA with traces etc? Environment notes: FS version: FreeSWITCH Version 1.6.8+git~20160505T153832Z~99de0ad502~64bit (git 99de0ad 2016-05-05 15:38:32Z 64bit) this is a single-node FS test — no clustering or anything like that

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correct method to play an audio file in a call originated from Freeswitch to a mobile number
by Deepika Yadav 26 Aug '16

26 Aug '16

Hi, I have a use case using ESL, where I need to originate a call from from Freeswitch to a mobile number and play a .wav after the recipient has answered the call. I am using following way to achieve this : Step 1 . Issue esl command = ""originate " + gateway + "+91" + phone_number + " &park"" Step 2. command = ""uuid_transfer " + uuid + " 4441"" where dialplan 4441 has : <extension name="ivr_demo"> <condition field="destination_number" expression="^4441$"> <action application="playback" data="/usr/local/freeswitch/recordings/file.wav"/> <action application="sleep" data="1000"/> </condition> </extension> This works fine but sometimes, after answering of the call, listener does not hear anything, file does not play. If I do not use the park command and directly transfer to dialplan then the file starts playing immediately even if the user has not answered. How can I add a condition of answering in dialplan (<action application="answer"/> does not work). Is any other alternative to achieve this correctly. Regards, Deepika

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explanation for values of <audio> section in xml_cdr
by Igor Karymov 26 Aug '16

26 Aug '16

Hi there! Can i find somewhere explanation about the meaning of values of <audio> section in xml_cdr? One of my clients having a one-way audio. CDR for this call looks pretty strange: http://pastebin.com/jqhCXKec I guess that freeswitch actually got packets from the client but drop all of them because of something. But not sure if my interpretation is correc

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ESL + BYE SIP reason code
by Gonzalo Gasca Meza 26 Aug '16

26 Aug '16

Hi guys, I'm getting the following BYE reason code from my Service Provider: recv 740 bytes from udp/[177.71.206.193]:5060 at 02:57:50.740042: ------------------------------------------------------------------------ BYE sip:gw+twilio@52.1.1.1:5090;transport=udp;gw=twilio SIP/2.0 CSeq: 1 BYE From: <sip:+551121174769@turn2data-telephonist.pstn.br1.twilio.com >;tag=76430350_6772d868_ebbbd5d1-d9b1-49db-981e-ec4b3e238564 To: "" <sip:+55114XXXYYYY@172.31.3.166>;tag=mjZyjF9Ute8DD Call-ID: 7d8a1944-e512-1234-4eaf-02d310a86cb3 Max-Forwards: 68 Via: SIP/2.0/UDP 177.71.206.193:5060;branch=z9hG4bKfaa.6c72ded1.0 Via: SIP/2.0/UDP 172.19.147.73:5060 ;rport=5060;received=177.71.206.215;branch=z9hG4bKebbbd5d1-d9b1-49db-981e-ec4b3e238564_6772d868_205642673073712 * Reason: MVTSLocal;cause=10;text="BYE received",SIP;cause=406;text="Service Unavailable"* User-Agent: Twilio Gateway X-Twilio-CallSid: CA5411d5015d3df04541c6e14e9fb41719 Content-Length: 0 When placing call via ESL I get reported NORMAL_CLEARING vs 406 I don't see 406 Listed in Wiki: https://freeswitch.org/confluence/display/FREESWITCH/Hangup+Cause+Code+Table According to RFC seems to be valid https://tools.ietf.org/html/rfc3326 Examples are: Reason: SIP ;cause=200 ;text="Call completed elsewhere" Reason: Q.850 ;cause=16 ;text="Terminated" Reason: SIP ;cause=600 ;text="Busy Everywhere" Reason: SIP ;cause=580 ;text="Precondition Failure" Can we add support to handle these type of reasons to a common one ? Like Unknown? I will talk to my SP to see if they can change it to 503. Any suggestions?

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