September 2016 - FreeSWITCH-users - FreeSWITCH Mailing Lists

call is not routing from asterisk->freeswitch(billing platform)->provider
by devang nathwani 16 Sep '16

16 Sep '16

Hello, I am trying to call from vicidial to freeswitch(billing platform) to provider(gateway) so call flow would be like this My vicidial 60.80.70.250 my freeswitch 92.42.132.242 my provider(gateway) 72.2.242.42 I am dialing 18177779695 When i try using dialtype 'MANUAL' the call is routing correctly without any issue. But when i try using dialtype 'RATIO', i am facing issue when freeswitch is resending the provider(gateway)'s response to vicidial the asterisk of vicidial server responding with X-Asterisk-HangupCause: Unknown. X-Asterisk-HangupCauseCode: 0. I am attaching the sip log of both working(MANUAL) and and not working(RATIO) scenario here. Please note that the attached sip log is from freeswitch server(billing platform). Working sip log is here http://pastebin.com/ma4e3xzk Not working sip log is here http://pastebin.com/6PAnyDBc Please advice, what am I missing here?

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Best Practice - * Codes on some IP Phones
by William Colls 15 Sep '16

15 Sep '16

I am working with a Grandstream HT704 ATA device which has number special functions built in eg Do Not Disturb, Suppress Outgoing Caller ID Number, Enable/Disable Call waiting, - more than 20 in all. Unfortunately, one of the codes is *98 which pays back the extension number. In order for *98 to work for e-mail, I must disable all the call features. So my question- Is it generally considered best practice to disable the functionality on all phones, and provide it through the dial plan so that the codes will be consistent for all phones in the system, or let the phones provide the features, even if they may vary from phone to phone? Thanks for your time. William.

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Re: [Freeswitch-users] Broadcasting conference with mod_rtmp
by Charlie DeTar 15 Sep '16

15 Sep '16

On Wed, Sep 14, 2016, at 08:15 PM, Michael Jerris wrote: > we run rtmp streams all the time without crashing so something weird > is going on, i doubt it's the packaging. Can you get a backtrace of > the crash and report a bug to jira so we can investigate? Thanks, here is the bug report: https://freeswitch.org/jira/browse/FS-9520 .

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verto.js
by Mirko Brankovic 15 Sep '16

15 Sep '16

Hi, Just a general question... We were looking at jquery.verto.js and jquery.FSRTC.js, there are plenty of 'hacks' and 'fixes' like: *if (options.onAnswerSDP) sdp = sdp.replace(/(a=crypto:0 AES_CM_128_HMAC_SHA1_32)(.*?)(\r\n)/g, '');* *sdp = sdp.replace(/b=AS([^\r\n]+\r\n)/g, '');* or like *stereoHack()* Are these to fix problems on the FS core side, like removing this crypto from sdp or double carriage return. Is all that still necessary? I'm just generally worried about Verto.js <> FS dependency in case I want to use any of those components in projects. -- Regards, Mirko

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Server shutdown event process
by Josh Nankin 14 Sep '16

14 Sep '16

I see there are several events fired when Freeswitch terminates: - SERVER_DISCONNECTED - SESSION_CRASH - SHUTDOWN Under what circumstances is each of these fired? Is there an event that is guaranteed to always be fired before the server stops? Is there an event that is fired during an abnormal termination vs a normal/graceful shutdown?

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5 second delay when taking a phone off hold
by Dave Horton 14 Sep '16

14 Sep '16

I have noticed that when I put a phone on hold, and then later take it off hold that there is a 5 second delay when Freeswitch handles the re-INVITE off-hold (no such delay going on hold). My app is a fairly simple bridging application, where call is placed from A to B through Freeswitch; the B leg goes on hold and then later attempts to go off hold. I have tracked the 5 second delay to this line of code in switch_core_media.c: if (b_channel) { if (switch_channel_test_flag(session->channel, CF_HOLD)) { switch_ivr_unhold(b_session); switch_channel_clear_flag(session->channel, CF_SUSPEND); switch_channel_clear_flag(session->channel, CF_HOLD); } else { switch_channel_stop_broadcast(b_channel); switch_channel_wait_for_flag(b_channel, CF_BROADCAST, SWITCH_FALSE, 5000, NULL); <== 5 second delay happens here } } Couple of other data points: 1. When I go on hold, the INVITE on hold that Freeswitch receives on one leg is not propagated out the associated leg, for whatever reason 2. I am using late negotiation of the SDP between the A and B legs. 3. I am running version 1.68 Is this a bug and should I open a Jira? I can obviously provide traces but wanted to first know if this is a bug or some configuration issue Dave

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Madboss enter tone
by Shawn Wheeler 14 Sep '16

14 Sep '16

I have what I think should be an easy question, I am just too ignorant to know where to look. I am using the Madboss to initiate group calls. When the group call is initiated from my handset, I hear a tone. I also hear a tone from my receiving SIP devices. I have been able to change and or mute the tone on the receiving SIP devices. I would like to be able to change the tone on my handset, where the call is initiated. I have looked in the conf\autoload_confs\confernece.conf.xml file. I see enter and exit tones for both wide and ultawide band. However on my system I only hear the entry tone. If you have a thought were I can look, please let me know. Shawn

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Install freeswitch on ubuntu 15.10
by Yuriy Gorlichenko 14 Sep '16

14 Sep '16

Hi. I trying to install freeswitch on ubuntu 15.10. I know that it is not TLS release and etc. Just need to install it there Im realdy to install it manually I found freeswitch on a github and cloned it, just need packages-dependenses for successfull installation it on this OS. Can anybody provide link for this or give me a list of packages?

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Play hold music while try to call to customer
by Dmitry Mordovin 14 Sep '16

14 Sep '16

Hello, I can't figure out how to configure hold music to play from begin extension till customer answer. The problems is: - Hold music play while bridge calling to customer only and silence while sleep 5 sec before second try - Hold music begin play .wav from start each time when bridge switch to next route. Any ideas how to configure hold music whole call session and, start playing after ANSWER (1 step) and stop play when customer answer(3 step) or bridge fail (4 step)? My config: <extension name="srv_sctv"> <condition field="destination_number" expression="^12345$">  <action application="answer"/> <action application="set" data="continue_on_fail=true"/> <action application="playback" data="/opt/freeswitch-share/fs_sounds/ring_welcome.wav"/>  <action application="set" data="ignore_early_media=true" /> <action application="set" data="instant_ringback=true" /> <action application="export" data="hold_music=/opt/freeswitch-share/moh/0/2_8.wav" /> <action application="set" data="transfer_ringback=file_string:///opt/freeswitch-share/moh/0/2_8.wav"/> <action application="set" data="ringback=/opt/freeswitch-share/moh/0/2_8.wav"/>  <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="[leg_timeout=20]sofia/gateway/gw1/79262410000@wg1|[leg_timeout=20]sofia/gateway/gw1/79262411111@gw1"/>  <action application="sleep" data="5000"/> <action application="bridge" data="[leg_timeout=20]sofia/gateway/gw1/79262410000@wg1|[leg_timeout=20]sofia/gateway/gw1/79262411111@gw1"/>  <action application="playback" data="/opt/freeswitch-share/fs_sounds/ru/connection_not_established.mp3"/> <action application="hangup"/> </condition> </extension> Thank All!

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Broadcasting conference with mod_rtmp
by Charlie DeTar 14 Sep '16

14 Sep '16

Hi, I'm trying to broadcast a WebRTC video conference (via mod_conference and mod_verto) over an rtmp stream, with the ultimate goal of transcoding the rtmp stream for broadcast as an HTML5 video stream (likely using something like https://github.com/arut/nginx-rtmp-module for HSL and MPEG-DASH broadcasts). I expect I have the same sort of requirements of freeswitch that one would need for broadcasting rtmp to youtube, which the docs mention glancingly without detail [0]. :) The video conferencing is working great -- but I'm struggling to figure out how to access the RTMP stream for it. My rtmp.conf.xml is basically the default (but with `auth-calls` set to `false`). My dialplan is as follows: <include> <extension name="plenary"> <condition field="destination_number" expression="^plenary-.*$" break="never"> <action application="answer"/> <action application="info"/> <action application="conference" data="${destination_number}@plenary-conference-profile"/> </condition> </extension> </include> I'm able to connect to freeswitch using RTMP clients (I've tried rtmpdump, mplayer, and vlc). In fs_cli, I see debug info (example copied below [1]) that seems to indicate successful connections. However, the RTMP client never receives any data from the server and disconnects after around 30 seconds. I'm trying to connect via URLs like: rtmp://<domain>:<port>/plenary-conferencename rtmp://<domain>:<port>/plenary-conferencename@plenary-conference-profile rtmp://<domain>:<port>/default/plenary-conferencename and so on, and all exhibit the same behavior. It doesn't seem to matter what path I put -- freeswitch accepts the connection, sends no data, then disconnects after a while. This leads me to suspect I might just be getting the path name wrong. My questions, if anyone has experience with this: What is the correct path for the rtmp URL to connect to a video conference -- is it just the $destination_number, or something else? Are there any obvious steps I'm missing to set up rtmp feeds, or any suggestions for how to do smarter debugging of this? Does anyone have an example project that successfully broadcasts RTMP from a video conference to e.g. youtube? Is there a better or more convenient approach to broadcasting a video stream from a conference that you prefer to rtmp? The full freeswitch config I'm using is here: https://gitlab.com/cdetar/plenary-server/tree/master/ansible/roles/plenary/… , if that's helpful. (Those are templates for an ansible config, so they contain variables that get interpolated in using {{var}} syntax). best, Charlie [0] Glancing reference to youtube+rtmp: https://freeswitch.org/confluence/display/FREESWITCH/FreeSWITCH+1.6+Video#F… [1] fs_cli debug info while attempting to connect via RTMP: 2016-09-13 23:37:40.551744 [NOTICE] mod_rtmp.c:903 New RTMP session [1513874f-9084-470c-a2de-7e5f92bb2296] 2016-09-13 23:37:40.551744 [INFO] rtmp_tcp.c:234 Rtmp connection from 98.127.250.255:44131 2016-09-13 23:37:40.551744 [DEBUG] rtmp.c:882 Sent handshake response 2016-09-13 23:37:40.711748 [DEBUG] rtmp.c:907 Done with handshake 2016-09-13 23:37:40.711748 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=204 2016-09-13 23:37:40.711748 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for connect 2016-09-13 23:37:40.711748 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1 stream_id=0x0] len=4 2016-09-13 23:37:40.711748 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x5 stream_id=0x0] len=4 2016-09-13 23:37:40.711748 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x6 stream_id=0x0] len=5 2016-09-13 23:37:40.711748 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4 stream_id=0x0] len=6 2016-09-13 23:37:40.711748 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14 stream_id=0x0] len=201 2016-09-13 23:37:40.711748 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14 stream_id=0x0] len=61 2016-09-13 23:37:40.711748 [NOTICE] rtmp_sig.c:122 Sent connect reply 2016-09-13 23:37:40.851748 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x5 ts=0 stream_id=0x0] len=4 2016-09-13 23:37:40.851748 [DEBUG] rtmp.c:1184 Set window size: from 2097152 to 2097152 bytes 2016-09-13 23:37:40.851748 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x4 ts=0 stream_id=0x0] len=10 2016-09-13 23:37:40.851748 [DEBUG] rtmp.c:91 Control (3): 00 00 00 00 00 00 01 2c 2016-09-13 23:37:40.851748 [INFO] rtmp.c:118 stream=0 Client buffer set to 300ms 2016-09-13 23:37:40.851748 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=25 2016-09-13 23:37:40.851748 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for createStream 2016-09-13 23:37:40.851748 [DEBUG] rtmp.c:656 [amfnumber=3 type=0x14 stream_id=0x0] len=29 2016-09-13 23:37:40.851748 [INFO] rtmp_sig.c:137 Replied to createStream (0) 2016-09-13 23:37:40.991748 [DEBUG] rtmp.c:1081 [chunk_stream=8 type=0x14 ts=0 stream_id=0x1] len=82 2016-09-13 23:37:40.991748 [DEBUG] rtmp.c:192 [amfnumber=8] Got INVOKE for play 2016-09-13 23:37:40.991748 [DEBUG] rtmp_sig.c:198 Got play for plenary-test-pattern-alpha@plenary-conference-profile on stream 1 2016-09-13 23:37:40.991748 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x1 stream_id=0x0] len=4 2016-09-13 23:37:40.991748 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4 stream_id=0x0] len=6 2016-09-13 23:37:40.991748 [DEBUG] rtmp.c:656 [amfnumber=2 type=0x4 stream_id=0x0] len=10 2016-09-13 23:37:40.991748 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14 stream_id=0x1] len=135 2016-09-13 23:37:40.991748 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x14 stream_id=0x1] len=135 2016-09-13 23:37:40.991748 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x12 stream_id=0x1] len=44 2016-09-13 23:37:40.991748 [DEBUG] rtmp.c:656 [amfnumber=5 type=0x12 stream_id=0x1] len=24 2016-09-13 23:37:40.991748 [DEBUG] rtmp.c:1081 [chunk_stream=2 type=0x4 ts=0 stream_id=0x0] len=10 2016-09-13 23:37:40.991748 [DEBUG] rtmp.c:91 Control (3): 00 00 00 01 02 25 51 00 2016-09-13 23:37:40.991748 [INFO] rtmp.c:118 stream=1 Client buffer set to 36000000ms 2016-09-13 23:37:43.951733 [DEBUG] rtmp.c:1081 [chunk_stream=3 type=0x14 ts=0 stream_id=0x0] len=34 2016-09-13 23:37:43.951733 [DEBUG] rtmp.c:192 [amfnumber=3] Got INVOKE for deleteStream 2016-09-13 23:37:43.951733 [NOTICE] rtmp.c:918 Disconnected from flash client 2016-09-13 23:37:43.951733 [DEBUG] rtmp_tcp.c:243 Closing socket 2016-09-13 23:37:44.691705 [NOTICE] mod_rtmp.c:1025 RTMP session ended [1513874f-9084-470c-a2de-7e5f92bb2296]

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