Re: [asterisk-users] Experiencing what I think are issues with the confbridge 'video_mode = follow_talker' and also the talk detection
On Wed, Mar 13, 2019 at 1:24 PM Dan Cropp wrote: > Running a test using asterisk 16.1.1 and two PCs with Firefox browsers. > I’m running the cmp2k demo. > > > > I place calls into the same asterisk and using AMI answer the calls and > then add them into the same confbridge. > > Video mode is configured to follow_talker. However, the Remote Video > displayed to both browsers is always the video of the opposite call. It’s > not following whoever talked last. I have the talk_detection_events > enabled and looking at the event it seems asterisk isn’t always detecting > the talking correctly. > If Asterisk is not always detecting the talking then it might be that you need to adjust the dsp_talking_threshold and/or dsp_silence_threshold settings. I'm not sure why the video is not switching unless it's the same thing as the other problem you ran into and the browsers aren't handling the streams swapping. > > > Next question…. > > > > If asterisk is reporting channel PJSIP/webrtc_client1-000e is talking, > shouldn’t the confbridge follow_talker setting change this confbridge > VideoSource to be the Uniqueid of this channel? > > > > Event: ConfbridgeTalking^M > > Privilege: call,all^M > > Conference: Bridge2^M > > BridgeUniqueid: ea6e7cb6-cc86-413e-885c-b0a25be887ee^M > > BridgeType: base^M > > BridgeTechnology: softmix^M > > BridgeCreator: ConfBridge^M > > BridgeName: Bridge2^M > > BridgeNumChannels: 3^M > > BridgeVideoSourceMode: talker^M > > BridgeVideoSource: 1552492132.33^M > > Channel: PJSIP/webrtc_client1-000e^M > > ChannelState: 6^M > > ChannelStateDesc: Up^M > > CallerIDNum: webrtc_client1^M > > CallerIDName: ^M > > ConnectedLineNum: ^M > > ConnectedLineName: ^M > > Language: en^M > > AccountCode: 19^M > > Context: ABC^M > > Exten: 5^M > > Priority: 14^M > > Uniqueid: 1552492117.32^M > > Linkedid: 1552492117.32^M > > TalkingStatus: on^M > > Admin: No^M > While the follow_talker setting initiates the "switching" process I'd expect most of the time the Id's to be different here. When talking is detected, and when this event is dispatched can differ from when the video source actually changes. When the video source changes you should see a BridgeVideoSourceUpdate event. I'd suspect that usually it'd follow the ConfbridgeTalking event in most cases. > Last question…. > > > > I see times where it seems the talk detection seems to become stuck for a > channel. I see this event happening and the confbridge videosource becomes > this channel’s uniqueid. I have not talked into this PC/browser’s mic in > hours. Literally went to lunch and it seems stuck in the TalkingStatus: on > state. > > I have hundreds of ConfbridgeTalking events for the other channel (on and > off) over the next several hours, but that channel’s TalkingStatus seems > stuck. > I am unsure what could cause this unless there is some kind of ambient noise that is being detected and your thresholds need to be adjusted. Otherwise it sounds like a bug to me. If this is happening consistently and your settings seem good then I'd suggest opening a ticket on the issue tracker [1]. Please attach (as *.txt files) a full asterisk debug log [2] (debug and verbose set to at least 5, and sip/pjsip debugging), along with the AMI log, and relevant configurations. [03/13 10:49:00.595] DEBUG[49360] manager.c: Examining AMI event: > > Event: ConfbridgeTalking^M > > Privilege: call,all^M > > Conference: OpBridge2^M > > BridgeUniqueid: ea6e7cb6-cc86-413e-885c-b0a25be887ee^M > > BridgeType: base^M > > BridgeTechnology: softmix^M > > BridgeCreator: ConfBridge^M > > BridgeName: OpBridge2^M > > BridgeNumChannels: 2^M > > BridgeVideoSourceMode: talker^M > > BridgeVideoSource: 1552492132.33^M > > Channel: PJSIP/webrtc_client1-000f^M > > ChannelState: 6^M > > ChannelStateDesc: Up^M > > CallerIDNum: webrtc_client1^M > > CallerIDName: ^M > > ConnectedLineNum: ^M > > ConnectedLineName: ^M > > Language: en^M > > AccountCode: 19^M > > Context: ABC^M > > Exten: ^M > > Priority: 14^M > > Uniqueid: 1552492132.33^M > > Linkedid: 1552492132.33^M > > TalkingStatus: on^M > > Admin: No^M > > > > > > The templates I’m using are > > > > Action: SetVar > > ActionID: C173 > > Channel: PJSIP/webrtc_client1-000f > > Variable: CONFBRIDGE(bridge,template) > > Value: 2 > > > > Action: SetVar > > ActionID: C174 > > Channel: PJSIP/webrtc_client1-000f > > Variable: CONFBRIDGE(user,template) > > Value: 4 > > > > Action: SetVar > > ActionID: C176 > > Channel: PJSIP/webrtc_client1-000e > > Variable: CONFBRIDGE(bridge,template) > > Value: 2 > > > > Action: SetVar > > ActionID: C177 > > Channel: PJSIP/webrtc_client1-000e > > Variable: CONFBRIDGE(user,template) > > Value: 4 > > > > [2] > > type = bridge > > language = en > > internal_sample_rate = 0 > > mixing_interval = 20 > > record_file_append = no > > max_members = 10 > > video_mode = follow_talker > > > > [4] > > type = user > > admin = no > > marked = no > > startmuted = no > >
Re: [asterisk-users] pattern matching "+"
Le 15/03/2019 à 15:18, sean darcy a écrit :
From my provider I get extensions of:
+1<10digit number>
1<10 digit number>
<10 digit number>
seemingly randomly.
What I'd like to do is
exten=_!1234567890,1,Answer()
which would match anything ending in 1234567890.
But that doesn't work since ! can only be used at the end of a pattern.
I tried
[+\ ][1\ ]1234567890
which didn't work, probably because "\ " means space, not nothing.
Any suggestions?
exten = _+.,1,Goto(${EXTEN:-10})
exten = _X.,1,Goto(${EXTEN:-10})
...
Daniel
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Re: [asterisk-users] Does anyone know if there is a problem with the Chrome browser and asterisk cmp2k video
Thank you Joshua -Original Message- From: asterisk-users On Behalf Of Joshua C. Colp Sent: Friday, March 15, 2019 9:32 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Does anyone know if there is a problem with the Chrome browser and asterisk cmp2k video On Fri, Mar 15, 2019, at 11:29 AM, Dan Cropp wrote: > > Thank you Kevin. > > > Any idea if Google developers may decide to reset the decoder, per > Joshua’s experience? It's hard to really know the way things will go in the end, it's always evolving. > Or perhaps asterisk developers would consider eventually add the > rewrite support in asterisk? It's certainly something that could be done, I just don't know of anyone working on it right now or if anyone plans to. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Does anyone know if there is a problem with the Chrome browser and asterisk cmp2k video
Thank you Kevin.
Any idea if Google developers may decide to reset the decoder, per Joshua’s
experience?
Or perhaps asterisk developers would consider eventually add the rewrite
support in asterisk?
Always fun dealing with newer technology as it goes through several revisions.
Have a great day!
Dan
From: asterisk-users On Behalf Of
Kevin Harwell
Sent: Thursday, March 14, 2019 4:50 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Does anyone know if there is a problem with the
Chrome browser and asterisk cmp2k video
On Wed, Mar 13, 2019 at 10:15 AM Dan Cropp
mailto:d...@amtelco.com>> wrote:
Using asterisk 16.1.1.
I’m setting up a test using the cmp2k (Cyber Mega Phone 2K Ultimate Dynamic
Edition).
I have noticed Chrome 72 had some issues with video streams. I just upgraded
to Chrome 73 and see they still have some issues. If I have 2 calls in a
confbridge with video set to none. I then set the video source to a Chrome
browser and the Remote Video shown to both calls from Firefox and Chrome do not
update. However, if I set the video source to the Firefox browser, my Remote
Video is accurate in both Firefox and Chrome.
I confirmed that asterisk is indicating the video source changed by looking at
the AMI BridgeVideoSourceUpdate event.
When I use the Firefox (65.0.2) browser I can set either call to be the video
source and the Remote Video updates accordingly.
Is this caused by Chrome’s video sent to asterisk being some format which
asterisk can’t use in the confbridge?
The way I understand it is that video stream has changed, so the browser needs
some way to know that. Otherwise the decoder thinks it's invalid data and drops
it. In these cases either Asterisk needs to issue a renegotiation (currently
not supported), or the codec needs to contain video stream information in their
payload. I spoke with Joshua Colp about this some as he's had some dealings
with this and he had the following to say:
"Some codecs (such as VP8/V9) embed information about the video stream within
their payload. Asterisk does not currently rewrite this information, so when a
stream change occurs in Asterisk using the selective source functionality this
can cause the receiving side (the browser) to drop the payload as it sees it as
not being part of the existing stream. Different browsers can behave
differently, such as resetting the video decoder to handle the new stream.
Rewriting this information in the video payload is not currently supported."
So you are probably seeing it work or not in Chrome vs Firefox due to browser,
and codec support of such occurrences.
--
Kevin Harwell
Digium - A Sangoma Company | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: https://digium.com & https://asterisk.org
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Re: [asterisk-users] Does anyone know if there is a problem with the Chrome browser and asterisk cmp2k video
On Fri, Mar 15, 2019, at 11:29 AM, Dan Cropp wrote: > > Thank you Kevin. > > > Any idea if Google developers may decide to reset the decoder, per > Joshua’s experience? It's hard to really know the way things will go in the end, it's always evolving. > Or perhaps asterisk developers would consider eventually add the > rewrite support in asterisk? It's certainly something that could be done, I just don't know of anyone working on it right now or if anyone plans to. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pattern matching "+"
On Friday 15 March 2019 at 15:18:04, sean darcy wrote:
> From my provider I get extensions of:
>
> +1<10digit number>
> 1<10 digit number>
> <10 digit number>
>
> seemingly randomly.
>
> What I'd like to do is
>
> exten=_!1234567890,1,Answer()
> Any suggestions?
exten => _+1XX,Goto(${EXTEN:2})
exten => _1XX,Goto(${EXTEN:1})
exten => _XX,Rest of your dialplan
Antony.
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Re: [asterisk-users] pattern matching "+"
On Fri, Mar 15, 2019 at 9:19 AM sean darcy wrote: > From my provider I get extensions of: > > +1<10digit number> > 1<10 digit number> > <10 digit number> > > seemingly randomly. > > What I'd like to do is > > exten=_!1234567890,1,Answer() > > which would match anything ending in 1234567890. > > But that doesn't work since ! can only be used at the end of a pattern. > > I tried > > [+\ ][1\ ]1234567890 > > which didn't work, probably because "\ " means space, not nothing. > > Any suggestions? > You must have multiple patterns to match the various starting sequences you receive. One that begins with + One that begins with 1 One that is for a 10 digit number Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] pattern matching "+"
From my provider I get extensions of: +1<10digit number> 1<10 digit number> <10 digit number> seemingly randomly. What I'd like to do is exten=_!1234567890,1,Answer() which would match anything ending in 1234567890. But that doesn't work since ! can only be used at the end of a pattern. I tried [+\ ][1\ ]1234567890 which didn't work, probably because "\ " means space, not nothing. Any suggestions? sean -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR Loop
Your IVR should only play audio prompts and only attempt to dial once a selection has been made, Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] IVR Loop
What do you mean by 8 digit? You want the user to have to enter an 8 digit
number to get to a specific operator?
On Fri, Mar 15, 2019 at 9:39 AM Gokan Atmaca wrote:
> Hello
>
> IVR call is coming. I want 8 (digit) in the loop. How can I do that ?
>
> Current confi:
> [ivr1]
> exten=>_,1,answer()
> exten=>_,n,background(/var/lib/asterisk/ivr/ob)
> exten=>_,n,WaitExten(10)
> exten=>_,n,Dial(${OPERATOR})
> exten=>i,1,Dial(${OPERATOR})
> exten=>t,1,Dial(${OPERATOR})
> exten=>1,1,dial(SIP/6001)
> exten=>2,1,dial(SIP/6002)
> exten=>3,1,dial(SIP/6003)
>
>
> Thanks.
>
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> https://community.asterisk.org/
>
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[asterisk-users] IVR Loop
Hello
IVR call is coming. I want 8 (digit) in the loop. How can I do that ?
Current confi:
[ivr1]
exten=>_,1,answer()
exten=>_,n,background(/var/lib/asterisk/ivr/ob)
exten=>_,n,WaitExten(10)
exten=>_,n,Dial(${OPERATOR})
exten=>i,1,Dial(${OPERATOR})
exten=>t,1,Dial(${OPERATOR})
exten=>1,1,dial(SIP/6001)
exten=>2,1,dial(SIP/6002)
exten=>3,1,dial(SIP/6003)
Thanks.
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