- FreeSWITCH-users - FreeSWITCH Mailing Lists

Conference multi-canvas question
by Ryan Delgrosso 05 May '21

05 May '21

All, In mod_conference in a room using multiple canvases, I can see using the vid-canvas api how to place a user into different canvases but is there a way to place a participant into multiple canvases? Ex: Canvas 1: active speaker Canvas 2: 1up+5 I would like one user to be shown in both canvases and let the other participants opt into their view, choosing from only the speaker, or the speaker + audience. The current vid-canvas api seems to make their an either-or prospect. Suggestions? Thanks much! -Ryan

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Horizontally scaling mod_callcenter
by Nathan Stratton 02 May '21

02 May '21

We are currently using FreeSWITCH mod_callcenter for live video agents. It's working well, but recently we have been asked to greatly scale the number of queues and agents to the point where we will need many FreeSWITCH boxes to support them. Based on the documentation and posts I have read mdo_callcenter does NOT support multiple instances with a shared database. So what is the best practice to scale? ><> nathan stratton

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Re: [Freeswitch-users] Lua API call delays the packet sending out
by Seven Du 01 May '21

01 May '21

I have a patch that could do async curl using the curl multi api, but not finished yet. On Mon, Apr 19, 2021 at 7:07 PM VERMA Vineet (BCS/SPS) via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: "VERMA Vineet (BCS/SPS)" <vineet.verma(a)bics.com> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 19 Apr 2021 11:00:44 +0000 > Subject: Lua API call delays the packet sending out > > Dears, > > > > I am facing the issue that when REST API triggers the freeswitch I am > sending the reply with > > > > Following code: > > > > * stream:write("HTTP/1.1 200 OK\r\n")* > > * if (accept == "application/json" or accept == "*/*") then* > > * stream:write("Content-Type: application/json\r\n")* > > * else* > > * stream:write("Content-Type: text/plain\r\n")* > > * end* > > * stream:write("\r\n")* > > > > but it seems before packet it out , there is as deplay of 2-3 sec . > > > > can you please suggest how to solve this? > > I already tried > > stream:shutdown(); > > and > > stream:close() > > > > but did not helped. > > > > > > Thanks & Regards > > *Vineet Verma* > This e-mail cannot be used for other purposes than BICS business use. > > > > ---------- Forwarded message ---------- > From: "VERMA Vineet (BCS/SPS) via FreeSWITCH-users" < > freeswitch-users(a)lists.freeswitch.org> > To: "freeswitch-users(a)lists.freeswitch.org" < > freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Mon, 19 Apr 2021 04:07:36 -0700 (PDT) > Subject: [Freeswitch-users] Lua API call delays the packet sending out > _________________________________________________________________________ > > The FreeSWITCH project is sponsored by SignalWire https://signalwire.com > Enhance your FreeSWITCH install with disruptive priced SMS and PSTN > services. > Build your next product on our scalable cloud platform. > > Join our online community to chat in real time > https://signalwire.community > > Professional FreeSWITCH Services > sales(a)freeswitch.com > https://freeswitch.com > > Official FreeSWITCH Sites > https://freeswitch.com/oss > https://freeswitch.org/confluence > https://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 > https://freeswitch.com -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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Forking with one leg using a redirect server.
by David VILLAUME 30 Apr '21

30 Apr '21

Hello, We are having troubles with following scenario : Freeswitch receives an incoming call and forks to one user (B1) and a sip redirect-server (B2). If the contact inside the redirect (B2) is local, the leg B1 is is cancelled when the 302 reaches the server. d521d2f2-9c69-11eb-aec3-b7e8effe21d8 2021-04-13 17:06:27.569223 [NOTICE] sofia.c:6506 Hangup sofia/external/+33123456789 [CS_CONSUME_MEDIA] [REDIRECTION_TO_NEW_DESTINATION] d521c17c-9c69-11eb-aebe-b7e8effe21d8 2021-04-13 17:06:27.589224 [NOTICE] switch_ivr_originate.c:3601 Hangup sofia/internal-centrex/internaluser@172.16.33.40:5062<mailto:sofia/internal-centrex/internaluser@172.16.33.40:5062> [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] This occurs even if manual mode. Does anyone have idea about how to avoid this cancel ? Regards, David

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Re: [Freeswitch-users] disable MOH
by Juan Nino 27 Apr '21

27 Apr '21

I am trying to get freeswitch to ignore re invites and don't play any MOH as >>> the MOH already comes from the PBX, even when disabling it I still see in >>> the logs that is trying to play MOH >>> 2019-02-18 17:24:49.427916 [DEBUG] switch_ivr.c:623 >>> sofia/rtp-dirty/%2B12345@x.x.x.x:5060 Command Execute >>> playback(local_stream://moh) >>> >>> EXECUTE sofia/rtp-dirty/%2B12345@x.x.x.x:5060 >>> playback(local_stream://moh) >>> >>> any idea what has to be done to get freeswitch to continue passing RTP >>> without trying to impose its own? >>> >>> thanks >>> >> >> >

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WSS error and disconnection
by Kakiman Expert 27 Apr '21

27 Apr '21

Hello, I am working on freeSWITCH 1.10.5 with webRTC configured. Im connecting through WSS on port 7443 with SIPJS client and Chrome. The issue seems to be random but when it happens, the websocket is closed. Resulting in websocket disconnected (1006). It is happening very often and today, I had a lot of messages such as "registrar_tport_error() tport error 0 " and then all the users were unregistered. I had to restart FS to get back the possibility to register and make calls. Did you already met this issue ? how could it be solved ? thanks franck

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Multiple DIDs per extension
by John Tuxies 27 Apr '21

27 Apr '21

I got an isolated network and i got Freeswitch in it. I took user 1000 and registered it in an Asterisk server. The same happened for accounts 1001-1003 to different servers. Here is an icon of what i am trying to accomplish [image: ITSP_Scenario.jpg] The network is not connected to Internet or any other Voice service. I would like Freeswitch to act as an ITSP for the Asterisk servers and route in/out numbers. For example Asterisk's_1 server user which has the DID +493087710024 to be able and dial the number +49308771360 and reach another user on Asterisk Server 4. All the communication will go through the Server (Freeswitch). May i have some help on how to configure this thing please?

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(no subject)
by Kakiman Expert 26 Apr '21

26 Apr '21

Hello, I am working on freeSWITCH 1.10.5 with webRTC configured. Im connecting through WSS on port 7443 with SIPJS client and Chrome. The issue seems to be random but when it happens, the websocket is closed. Resulting in websocket disconnected (1006). It is happening very often and today, I had a lot of messages such as "registrar_tport_error() tport error 0 " and then all the users were unregistered. I had to restart FS to get back the possibility to register and make calls. Did you already met this issue ? how could it be solved ? thanks franck

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Github FreeSwitch bug report should have more labels
by mario_fs 26 Apr '21

26 Apr '21

Back in the Jira days, we could specify the type of bug, for instance build error. Now, cannot find a way to categorize a bug. I think this is useful because for instance, a build error may get priority because it does not allow you to continue testing, at least that’s my case. In Jira, I could specify a build error. I found no way to add a label to the type of bug, and assume that is reserved to the developers. I know it can be done because if you look at the bottom of 1145 <https://github.com/signalwire/freeswitch/issues/1145> and click the reference to 75728, you will see that issue has a label of “build failure” (homebrew labels), which I think can then be filtered so only those errors can be found easily. Just a thought, Mario G

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disable music on hold when in a conference?
by David Villasmil 26 Apr '21

26 Apr '21

Hello guys, I have this case where 3 users are in a conference, and if one of the puts the call on hold, the others will hear his usual music on hold. Is there a way of disabling MOH while in a conference? thanks, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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