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sip.js has no audio
by Hossein Gholizadeh 18 Jun '18

18 Jun '18

hi all ... i'm using sip.js for my IVR project ... it could connect to freeswitch correctly but there is no audio between call legs ... first of all i have to say i really cannot use Verto . this is my function of when call button pressed * phoneCallButtonPressed : function(sessionid) {* * var s = ctxSip.Sessions[sessionid],* * target = $("#numDisplay").val();* * if (!s) {* * $("#numDisplay").val("");* * ctxSip.sipCall(target);* * } else if (s.accept && !s.startTime) {* * s.accept({* * media : {* * stream : ctxSip.Stream,* * constraints : { audio : true, video : false },* * render : {* * remote : { audio: $('#audioRemote').get()[0] }* * },* * RTCConstraints : { "optional": [{ 'DtlsSrtpKeyAgreement': 'true'} ]}* * }* * });* * }* * },* ----------------------------------------------------------------------------------------- also i have to say that i tested it between my phone and x-lite ... X-lite could receive the voice but my phone could'nt ... i was using Zoiper in my phone.

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Recording skipping silence on outbound call recording
by Andrew Keil 18 Jun '18

18 Jun '18

To FreeSWITCH Users, I am attempting to make a recording of an outbound call using the originate API call which starts a Lua script which then simply records to disk exactly what it hears. I just noticed an interesting issue (or feature) where the recording skips any silence. What I am aiming for is for no silence to be skipped inside the recording, if someone can provide some settings for me to try I would be more than happy to experiment further. The settings for the Lua session:recordFile are length of recording: 60 seconds; silence threshold: 300; silence seconds: 60 seconds {essentially termination on silence is disabled since the silence seconds = recording seconds} I also tried with silence threshold = 0 and silence is still skipped inside the recording. I also tried the setting inside the appropriate profile: <param name="suppress-cng" value="true"/> and this made no difference, silence is still skipped inside the recording. My setup is one single FreeSWITCH 1.6.20 (production version) box with a simple internal gateway setup to land on the same FreeSWITCH IVR (just for testing). API command: originate {origination_uuid=121e20a6-1245-439a-8527-53efab009334-out-1,origination_caller_id_number=FLOODOUT,call_timeout=15,ignore_early_media=true,return_ring_ready=true,suppress_cng=true}sofia/gateway/visytel-pc-ivr/01111111111 FLOODOUT The SDP inside my log is similar to this for the outbound and inbound (Note: The outbound call is performing the recording): 121e20a6-1245-439a-8527-53efab009334-out-1 Local SDP: 121e20a6-1245-439a-8527-53efab009334-out-1 v=0 121e20a6-1245-439a-8527-53efab009334-out-1 o=FreeSWITCH 1529274701 1529274702 IN IP4 192.168.15.15 121e20a6-1245-439a-8527-53efab009334-out-1 s=FreeSWITCH 121e20a6-1245-439a-8527-53efab009334-out-1 c=IN IP4 192.168.15.15 121e20a6-1245-439a-8527-53efab009334-out-1 t=0 0 121e20a6-1245-439a-8527-53efab009334-out-1 m=audio 25994 RTP/AVP 9 8 0 101 121e20a6-1245-439a-8527-53efab009334-out-1 a=rtpmap:9 G722/8000 121e20a6-1245-439a-8527-53efab009334-out-1 a=rtpmap:8 PCMA/8000 121e20a6-1245-439a-8527-53efab009334-out-1 a=rtpmap:0 PCMU/8000 121e20a6-1245-439a-8527-53efab009334-out-1 a=rtpmap:101 telephone-event/8000 121e20a6-1245-439a-8527-53efab009334-out-1 a=fmtp:101 0-16 121e20a6-1245-439a-8527-53efab009334-out-1 a=silenceSupp:off - - - - 121e20a6-1245-439a-8527-53efab009334-out-1 a=ptime:20 121e20a6-1245-439a-8527-53efab009334-out-1 a=sendrecv 6da97c7c-1052-4f78-8dde-83901147a0c1 2018-06-18 15:44:55.596296 [DEBUG] sofia.c:7094 Remote SDP: 6da97c7c-1052-4f78-8dde-83901147a0c1 v=0 6da97c7c-1052-4f78-8dde-83901147a0c1 o=FreeSWITCH 1529274701 1529274702 IN IP4 192.168.15.15 6da97c7c-1052-4f78-8dde-83901147a0c1 s=FreeSWITCH 6da97c7c-1052-4f78-8dde-83901147a0c1 c=IN IP4 192.168.15.15 6da97c7c-1052-4f78-8dde-83901147a0c1 t=0 0 6da97c7c-1052-4f78-8dde-83901147a0c1 m=audio 25994 RTP/AVP 9 8 0 101 6da97c7c-1052-4f78-8dde-83901147a0c1 a=rtpmap:9 G722/8000 6da97c7c-1052-4f78-8dde-83901147a0c1 a=rtpmap:8 PCMA/8000 6da97c7c-1052-4f78-8dde-83901147a0c1 a=rtpmap:0 PCMU/8000 6da97c7c-1052-4f78-8dde-83901147a0c1 a=rtpmap:101 telephone-event/8000 6da97c7c-1052-4f78-8dde-83901147a0c1 a=fmtp:101 0-16 6da97c7c-1052-4f78-8dde-83901147a0c1 a=silenceSupp:off - - - - 6da97c7c-1052-4f78-8dde-83901147a0c1 a=ptime:20 6da97c7c-1052-4f78-8dde-83901147a0c1 Kind Regards, Andrew Keil

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Re: [Freeswitch-users] session.cause and session.causecode
by Francesco Facco de Lagarda 17 Jun '18

17 Jun '18

This is what the log says: 2018-06-17 11:18:41.186422 [ALERT] stuff.js:74 *********************** DIALING 2018-06-17 11:18:41.186422 [NOTICE] switch_channel.c:1104 New Channel sofia/external/065551234 [411ac79d-e8e0-4f77-b51f-7da24a71ea32] 2018-06-17 11:18:45.606424 [NOTICE] sofia.c:8273 Hangup sofia/external/065551234 [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER] 2018-06-17 11:18:45.626430 [NOTICE] switch_core_session.c:1683 Session 915 (sofia/external/065551234) Ended 2018-06-17 11:18:45.626430 [NOTICE] switch_core_session.c:1687 Close Channel sofia/external/065551234 [CS_DESTROY] 2018-06-17 11:18:45.626430 [ALERT] stuff.js:74 *********************** WAITING FOR SESSION READY .. 2018-06-17 11:18:45.626430 [ALERT] stuff.js:74 *********************** DISCONNECT CAUSE :false:undefined So this is what I want: 2018-06-17 11:18:45.606424 [NOTICE] sofia.c:8273 Hangup sofia/external/065551234 [CS_CONSUME_MEDIA] [UNALLOCATED_NUMBER] So why do I get: False: undefined from console_log("notice", "************************* disconnect cause:" + sessOut.cause + ":" + sessOut.causeCode); From: Francesco Facco de Lagarda <francesco(a)delagarda.com> Sent: domenica 17 giugno 2018 11:12 To: 'FreeSWITCH Users Help' <freeswitch-users(a)lists.freeswitch.org> Subject: session.cause and session.causecode 'Morning all I can't get session.cause and session.causecode to work. All I get is false for the code and blank for the cause! Im running FS 1.6.20 This is a simplified version of my JS script if (session.ready()) { session.answer(); var dialedNum = "3332094333"; console_log("notice", "*********************** DIALING"); var sessOut = new Session("sofia/gateway/realtoneFXO/" + dialedNum + "@192.168.0.216:5060"); console_log("notice", "*********************** WAITING FOR SESSION READY .."); if (sessOut.ready()) { console_log("notice", "*********************** SESSION READY"); bridge(session, sessOut); } else { console_log("notice", "************************* disconnect cause:" + sessOut.cause + ":" + sessOut.causeCode); } sessOut.hangup(); session.hangup(); console_log("notice", "*********************** CALL ENDED"); }

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wiki and org sites
by David Villasmil 14 Jun '18

14 Jun '18

Hello Guys, I can't connect to wiki.freeswitch.org, is the web down? I just get redirected to freeswitch's (new?) commercial web page (freeswitch.com) thanks! Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337 ᐧ

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number occupied
by Francesco Facco de Lagarda 14 Jun '18

14 Jun '18

I'm scripting in javascript How can I tell if the numer I called was occupied / busy? session.answer(); var sessOut = new Session(trunk + dialedNum); if (session.ready()) { /// all ok } // WHAT IF dialed num was busy or occupied ??

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Channel Hold Info
by mayamatakeshi 14 Jun '18

14 Jun '18

Hello, I have a requirement to generate reports of for how long calls were left on hold. My original plan was to monitor CHANNEL_HOLD/CHANNEL_UNHOLD events and update a channel variable preserving hold history. Then, I happened to find: https://freeswitch.org/confluence/display/FREESWITCH/CDR#CDR-Howlongwasthec… and realized there are already channel variables for this kind of purpose like hold_events, hold_accum_seconds and even hold-record. However, testing we recent commit: FreeSWITCH Version 1.9.0+git~20180516T213111Z~cfd99a03d4~64bit (git cfd99a0 2018-05-16 21:31:11Z 64bit) I verified all this data from XML CDR is related to the side that put the call on hold: Leg1) Side that put the other on hold: <hold_stamp>2018-06-14%2008%3A23%3A18</hold_stamp> <hold_events>%7B%7B1528932198592029,1528932199869251%7D%7D</hold_events> <last_hold_epoch>1528932198</last_hold_epoch> <last_hold_uepoch>1528932198592027</last_hold_uepoch> <hold_accum_seconds>1</hold_accum_seconds> <hold_accum_usec>1277224</hold_accum_usec> <hold_accum_ms>1277</hold_accum_ms> <hold-record> <hold on="1528932198592029" off="1528932199869251" bridged-to="1adf353f-a4e0-4e16-8953-fbd8beccca99"></hold> </hold-record> <last_hold_time>1528932198592027</last_hold_time> <hold_accum_time>1277224</hold_accum_time> Leg2) Side put on hold: <last_hold_epoch>0</last_hold_epoch> <last_hold_uepoch>0</last_hold_uepoch> <hold_accum_seconds>0</hold_accum_seconds> <hold_accum_usec>0</hold_accum_usec> <hold_accum_ms>0</hold_accum_ms> <last_hold_time>0</last_hold_time> <hold_accum_time>0</hold_accum_time> I am curious about this. Wouldn't it be better to have this kind of hold info at the side that was put on hold? Was this missed or done on purpose? I mean, we can with extra processing of hold-record of all Leg1s to resolve hold history for Leg2 however having this info at the Leg2 itself would be much simpler (and so I am considering in using the original CHANNEL_HOLD/CHANNEL_UNHOLD approach).

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Port numbers
by Rick Jarvis 14 Jun '18

14 Jun '18

Ok kind of FreeSWITCH 101 here I know, but something I’ve never understood completely: A provider wants to send us calls as number@our-ip-address - fine, but it’s hitting us on port 5060 (whereas external is set up as 5080). So can’t use external, have to use internal profile, right? So I can create a ACL to allow their IPs onto Internal, ‘inbound’ ACL I think? But how do I handle call routing for inbound calls from provider’s IPs, vs my registered users dialling out? If anyone can simplify this for me that would be great :) Thanks __ R

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When 1.6.21 ?
by Mario 13 Jun '18

13 Jun '18

Can anyone answer when 1.6.21 will be available? Just wondering since I haven’t been able to test FS production for macOS wiki since PCRE was updated (FS-11061). Also wanted to start testing on macOS Mojave to see how that goes. Thanks, Mario G

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Initiate SIP session with MRCPv2 server
by Rob Capo 13 Jun '18

13 Jun '18

Hello, I have an MRCPv2 client that I'd like to use in an ASR application which is connected to FreeSWITCH via eventsocket (application written in Go). I'd like to use this client for MRCP signaling because it has the fine-grained control that I need, and the API is easier to use than the unimrcp eventsocket API. However, I'd like to have FreeSWITCH take care of answering the incoming call from the user. I don't care quite as much about RTP and SIP signaling to/from the ASR, but would like some control over it if possible (e.g. being able to see the RTP data for recording segments of audio that are important to me). There are a few ways I can think of solving this, some hackier than others. Currently, what I'm doing is probably the hackiest: 1. answer the call with FS 2. Initiate MRCPv2 session with ASR using my client 3. get the call's local media port over ES 4. pcap that port, and proxy the media to the ASR manually Perhaps a better way would be: 1. answer the call with FS 2. initiate MRCPv2 session with ASR using my client 3. tell FS to proxy audio to the ASR's RTP port from SDP (maybe passing through a proxy UDP socket that I open) But I'm not sure if this is possible. Another way that might be more approachable: 1. answer the call with FS 2. bridge in the ASR using FS 3. get the SDP / MRCPv2 port from FS 4. initiate TCP connection to MRCPv2 port and use my client The disadvantage would be that I no longer have visibility / control over the SIP / RTP paths, but I could always use pcap as before if the need arises (the situations where I need this control are most likely going to be debugging / non-production purposes). Does anyone have better recommendations or details on how I could achieve the second and / or third methods? Thanks, Rob

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Access to release 1.8
by Kevin Olbrich 13 Jun '18

13 Jun '18

Hi! In Jan 2017 release 1.8 was described as "With the release date of 1.8 fast approaching" or "The result of all of this you ask? FreeSWITCH 1.8, and after that just think we'll be at FreeSWITCH 2.0!". As of now, there is only 1.6 available for public access. Freeswitch 1.6 is working fine but is 1.8 still in the works? Source: http://lists.freeswitch.org/pipermail/freeswitch-users/2017-January/124534.… Kind regards, Kevin

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