- FreeSWITCH-users - FreeSWITCH Mailing Lists

Re-Invite not passed to other leg
by Jason Grant 20 Jul '21

20 Jul '21

I was hoping someone could help clarify on whether re-invites are supposed to pass between both legs of a call when a call is placed on hold? As far as I can see, the re-invite when a caller is placed on hold is passed to FreeSWITCH but it is not passed on to the other leg. The issue I’m having is the SDP body changes of a=sendonly or a=sendrecieve never reach the other leg. Is this intended behaviour? Is there a way to change this? Kind regards Jason Grant

4 3

Re: [Freeswitch-users] RE-Invite in conference
by Giovanni Maruzzelli 20 Jul '21

20 Jul '21

Brian and Kaiduan, Thanks a lot! I had not thought that setting session-timeout would translate directly into session-expires! Also, I was misled by not receiving the reinvite in a non conference call. Enjoy Summer! -giovanni On Tue, Jul 20, 2021 at 3:57 PM kaiduan xie via FreeSWITCH-users < freeswitch-users(a)lists.freeswitch.org> wrote: > > > > ---------- Forwarded message ---------- > From: kaiduan xie <kaiduanx(a)yahoo.ca> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 20 Jul 2021 13:13:38 +0000 (UTC) > Subject: Re: [Freeswitch-users] RE-Invite in conference > Giovanni, > > I think the reINVITE is sent by session timer. > > Per rfc4028 <https://datatracker.ietf.org/doc/html/rfc4028#section-10> > https://datatracker.ietf.org/doc/html/rfc4028#section-10 > > rfc4028 > > Session Timers in the Session Initiation Protocol (SIP) (RFC ) > <https://datatracker.ietf.org/doc/html/rfc4028#section-10> > > > > It is RECOMMENDED that this refresh be sent once half the session interval has elapsed. > > > 900 seconds = 1/2 * 1800 > > > /Kaiduan > > > > > On Tuesday, July 20, 2021, 08:52:31 a.m. EDT, Giovanni Maruzzelli < > gmaruzz(a)gmail.com> wrote: > > > Hello FreeSWITCHers! > > I've found that in a conference (mod_conference) a re-invite is sent by FS > at around 900 sec after call confirmed (ACK). > > This has nothing to do with session-timeout (eg, session-expires) that are > both set to 1800, and sent by FS and clients accordingly. > > I can see this behavior only in conference (but maybe is elsewhere?). > > I've looked into both mod_sofia and mod_conference sources, but I've not > found anything related. > > I've tried to debug the call with all kind of debugging/tracing (sofia > lib, mod sofia, internals, etc etc) but I've not found any hint of what can > trigger that reinvite. > > Is it configurable? > > Have a nice continuation, > > keep on with the superb work, > > -giovanni > > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > > 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 > > > > ---------- Forwarded message ---------- > From: kaiduan xie via FreeSWITCH-users < > freeswitch-users(a)lists.freeswitch.org> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Tue, 20 Jul 2021 06:57:03 -0700 (PDT) > Subject: Re: [Freeswitch-users] RE-Invite in conference > _________________________________________________________________________ > > 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 -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

1 0

RE-Invite in conference
by Giovanni Maruzzelli 20 Jul '21

20 Jul '21

Hello FreeSWITCHers! I've found that in a conference (mod_conference) a re-invite is sent by FS at around 900 sec after call confirmed (ACK). This has nothing to do with session-timeout (eg, session-expires) that are both set to 1800, and sent by FS and clients accordingly. I can see this behavior only in conference (but maybe is elsewhere?). I've looked into both mod_sofia and mod_conference sources, but I've not found anything related. I've tried to debug the call with all kind of debugging/tracing (sofia lib, mod sofia, internals, etc etc) but I've not found any hint of what can trigger that reinvite. Is it configurable? Have a nice continuation, keep on with the superb work, -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

3 2

Issue with ReInvite after a=inactive
by Pirmin Walthert 15 Jul '21

15 Jul '21

Hi there I have the following issue: A calls B over FreeSWITCH B sends 180 Ringing without SDP (forwarded by FreeSWITCH to A correctly) B answers the call with 200 OK but with a=inactive (forwarded by FreeSWITCH to A correctly) B sends a ReInvite about 100ms after the 200 OK, but this time with a=sendrecv => while FreeSWITCH is answering with 200 OK and negotiating the codec, it is not sending any information about this event to A. Means: for A media is still in inactive state Any idea what the reason for this behavior could be or how to influence it? I am using version 1.10.6 on debian 10.8 (1.10.6-release-18-1ff9d0a60e~64bit) Best regards, Pirmin

2 6

Multiple Registrations inconsistency
by Stefan 13 Jul '21

13 Jul '21

Hello, I've a problem with multiple registrations on freeswitch 1.10.6. My internal profile contains the following: <param name="multiple-registrations" value="contact"/> And in the dialplan I use the bridge like this: <action application="bridge" data="${sofia_contact(${dialed_extension})}"/> It works as expected, all registered endpoints ring when i call an extension which is registered multiple times. The problem starts when one of the registered endpoints unregisteres. Then freeswitch derigisteres all, and none is reachable anymore. To be clear: a client with extension Number 2000 is registered 2 times, one Registration is through Linphone, the other one through a polycom phone. All is fine: When I call the extension 2000 both, Linphone and polycom ring. But now the user chooses to close the Linphone, and Linphone deregisters. The effect of this on freeswitch is, that the user is completely deregistered, not just the Linphone Registration, but also the one from polycom. As a result, I can't reach him anymore. Even more astonishing: When the user now turns on his Linphone again, it will register, and on calling the extension 2000 again both, Linphone and polycom will ring. I tried to set: <param name="inbound-use-callid-as-uuid" value="true"/> in my internal profile. But it did not change the described behaviour. So my question is: how can I make freeswitch deregister multiple registrations of the same user individualy? __ s.

4 4

Leg A and Leg B flow
by Andres Lavariega 08 Jul '21

08 Jul '21

Hello good Morning. I am looking for a little help and I am grateful for it. I have this scenario, on an established call, where I have leg A, and leg B. (source, destination) By pressing a DTMF on the side of leg A, send leg B to another trunk for an IVR process, and when terminating the call on that IVR. Leg A and Leg B, come back together in the initial flow. How can I send to another trunk, and put leg A on hold? Currently I have thought that I can do something like that, but I don't know how to put the B leg on hold, and then put the channels back together. <action application = "bind_meta_app" data = "5 a a execute_extension :: 1234 $ {destination_number} XML other-validation" />

1 0

Marking an active speaker in video stream
by Mirjana Jovanovic 08 Jul '21

08 Jul '21

Hi, Is it possible to mark an active speaker in the video stream, without affecting his position in the canvas? I am trying to achieve a similar effect that can be observed in Microsoft Teams. Active speaker is marked with a border around his video box. For example, there are 4 participants in the meeting. All of them are using video, and all are visible in the canvas. When the active speaker changes, I would like to avoid the shuffle effect, and just mark the active speaker with a border. I have tried to find available options in confluence, and haven’t found an option for drawing a different-color border for the active speaker. It is clear that a custom layout can be defined with an image with floor=true attribute. This would mean that an active speaker is always shown in this position, and that is the effect I would like to avoid. Thanks

1 0

Collect voice and dtmf input with lumenvox ASR/TTS
by shivali chalke 08 Jul '21

08 Jul '21

Hello, I am trying to collect voice and dtmf input at same time with lumenvox ASR. So far all the techniques I tried either work with speech or dtmf input. It doesn't work for both simultaneously. Any help regarding this would be appreciated. Regards, Shivali

1 0

SIP packets delay issue with rtp_notimer_during_bridge parameter
by Melek Oktay 07 Jul '21

07 Jul '21

Hello, In order to solve the delay issue for long calls, rtp_notimer_during_bridge parameter has been set to true. But this time, the SIP traffic delay problem occurs for all calls in my environment ! Has anybody faced the same issue ? Any help would be appreciated .

1 1

Call drop with 406 after being answered
by Igor Potjevlesh 07 Jul '21

07 Jul '21

Hello! I'm facing a problem with a call scenario with an in-dialog INVITE just after the call has been picked up. Freeswitch receive on his leg B this in-dialog INVITE. Not forwarded to leg A. In a SIP trace, I can see 406 Not Acceptable on leg B whereas there is no difference in the Media except "Bandwidth Information" which is "AS:80" versus "AS:82" in the RE-INVITE. I turn on the logs on debug, but nothing specific appears. The first thing: 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [NOTICE] sofia.c:1012 Hangup sofia/internal/ABC [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] switch_ivr_bridge.c:787 BRIDGE THREAD DONE [sofia/internal/ABC] 3f6d8dd1-5415-4a58-a948-551d841cf5dd 2021-06-11 10:34:12.873542 [DEBUG] switch_core_state_machine.c:653 (sofia/internal/ ABC) State EXCHANGE_MEDIA going to sleep [.] 5f1866f7-f6c6-43c9-83bb-07cbb51cc3a5 2021-06-11 10:34:12.893545 [DEBUG] switch_core_session.c:2815 sofia/lega/+FROM(a)W.X.Y.Z. skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) Is there any reason for this? SDP is not duplicated, but there is no specific reason for replying with 406. Regards, Igor. -- L'absence de virus dans ce courrier électronique a été vérifiée par le logiciel antivirus Avast. https://www.avast.com/antivirus

3 7