Re: [asterisk-users] Confbridge for 80 devices

On 10/20/2022 5:35 PM, Jerry Geis wrote: > > ;dsp_drop_silence=yes  ; This option drops what Asterisk detects as > silence from >                        ; entering into the bridge.  Enabling this > option will drastically >                        ; improve performance and help remove the > buildup

Re: [asterisk-users] Confbridge for 80 devices

On Thu, Oct 20, 2022 at 6:15 PM Eric Wieling wrote: > > https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces > > Thanks - so based on this wiki - seems like "The only functionality that requires internal timing is IAX2 trunking" - which I am not using . Just ConfBridge... And getting

Re: [asterisk-users] Confbridge for 80 devices

https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces On 10/20/22 17:35, Jerry Geis wrote: [modules] autoload = yes noload = res_timing_pthread noload = res_timing_timerfd SO I "dont" want to load res_timing_anything ??? I have preload on res_timing_dahdi, then res_timing_pthread and

Re: [asterisk-users] Confbridge for 80 devices

This is on the bare metal machine Recalculating Codec Translation (number of sample seconds: 1) Translation times between formats (in microseconds) for one second of data Source Format (Rows) Destination Format (Columns) ulaw alaw gsm g726 g726aal2

Re: [asterisk-users] Confbridge for 80 devices

[modules] autoload = yes noload = res_timing_pthread noload = res_timing_timerfd SO I "dont" want to load res_timing_anything ??? I have preload on res_timing_dahdi, then res_timing_pthread and not res_timing_timerfd at all. confbridge.conf is below [general] ; The general section of this

Re: [asterisk-users] Confbridge for 80 devices

Hi, Dahdi timing is for dahdi hardware. See here: https://wiki.asterisk.org/wiki/display/AST/Timing+Interfaces You could check your asterisk modules using "module show" on asterisk cli. Sounds like you might be doing transcoding,

Re: [asterisk-users] Confbridge for 80 devices

On 10/20/2022 5:17 PM, Jerry Geis wrote: > What is the trick to get "preload => res_timing_dahdi" working ? [modules] autoload = yes noload = res_timing_pthread noload = res_timing_timerfd However, it's unlikely to be a timing problem. Can you share your ConfBridge configuration? Kind

Re: [asterisk-users] Confbridge for 80 devices

What is the trick to get "preload => res_timing_dahdi" working ? I have tried to add to both a CentOS 7 (metal box) and Ubuntu 20.04 (VMware guest) system restart asterisk and neither print anything about res_timing_dahdi in the /var/log/asterisk/messages file. Both are having issues with around

Re: [asterisk-users] Confbridge for 80 devices

On Thu, Oct 20, 2022 at 1:53 PM Jerry Geis wrote: > Bringing 80 devices into a conf bridge with CPU: AMD Phenom(tm) II X6 > 1045T Processor at 2.7G and audio is reported as staticy or not the best > audio quality. > > Network is r8169 :02:00.0 eth0: RTL8168e/8111 > Link is 1G. > > Asterisk

[asterisk-users] Confbridge for 80 devices

Bringing 80 devices into a conf bridge with CPU: AMD Phenom(tm) II X6 1045T Processor at 2.7G and audio is reported as staticy or not the best audio quality. Network is r8169 :02:00.0 eth0: RTL8168e/8111 Link is 1G. Asterisk 18.14.0 I would think this should be able to handle 80 calls (one

Re: [asterisk-users] ConfBridge user joining not getting video

On Thu, Jan 13, 2022 at 10:45 AM Jerry Geis wrote: > >> Hi Josh > > >chan_sip did not add a video stream. What is the actual configuration for > > it? What is the actual call file used for it? > > sip.conf has videosupport in the general section. > > I did find that where I am "joining" the

Re: [asterisk-users] ConfBridge user joining not getting video

On Thursday 13 January 2022 at 15:45:02, Jerry Geis wrote: > > Hi Josh > > > >chan_sip did not add a video stream. What is the actual configuration for > > > > it? What is the actual call file used for it? > > sip.conf has videosupport in the general section. > > I did find that where I am

Re: [asterisk-users] ConfBridge user joining not getting video

> > > Hi Josh >chan_sip did not add a video stream. What is the actual configuration for > it? What is the actual call file used for it? sip.conf has videosupport in the general section. I did find that where I am "joining" the person in the conference I did not have the Codecs: set. I added

Re: [asterisk-users] ConfBridge user joining not getting video

On Thu, Jan 13, 2022 at 10:01 AM Jerry Geis wrote: > > > On Wed, Jan 12, 2022 at 5:09 PM Jerry Geis wrote: > >> I am running 18.8.0 - videosupport is enabled. I get video calls no >> problem. >> >> However when I make a call file to a soft phone and include: >> Codecs: ulaw,h264 >> in the call

Re: [asterisk-users] ConfBridge user joining not getting video

On Wed, Jan 12, 2022 at 5:09 PM Jerry Geis wrote: > I am running 18.8.0 - videosupport is enabled. I get video calls no > problem. > > However when I make a call file to a soft phone and include: > Codecs: ulaw,h264 > in the call file... > > sip show channels - shows: > 4013c15f1f4cdff

Re: [asterisk-users] ConfBridge user joining not getting video

On Wed, Jan 12, 2022 at 6:09 PM Jerry Geis wrote: > I am running 18.8.0 - videosupport is enabled. I get video calls no > problem. > > However when I make a call file to a soft phone and include: > Codecs: ulaw,h264 > in the call file... > > sip show channels - shows: > 4013c15f1f4cdff

[asterisk-users] ConfBridge user joining not getting video

I am running 18.8.0 - videosupport is enabled. I get video calls no problem. However when I make a call file to a soft phone and include: Codecs: ulaw,h264 in the call file... sip show channels - shows: 4013c15f1f4cdff (ulaw|h264) No Tx: ACK so clearly the caller has h264. Then

[asterisk-users] ConfBridge recording "Failed to get 160 samples from read factory" and "Read factory ... and write factory ... both fail to provide 160 samples"

We are running Asterisk 16.17.0 and discovered what we think is an issue. We have a single call in a ConfBridge. Tell the ConfBridge to start recording. We see non-stop audiohook.c 160 samples failures. As soon as we stop recording (AMI ConfBridgeStopRecord) these failures stop. [10/04

Re: [asterisk-users] asterisk-users Confbridge

Sorry about the trouble. Unsubscribed that user from the mailing lists. Matthew Fredrickson On Fri, Aug 7, 2020 at 9:20 PM Elizabeth wrote: > > I'm online on this site! > So contact me in my profile: > here > -- > _ > --

Re: [asterisk-users] asterisk-users Confbridge

Im online on this site! So contact me in my profile: galleries.daswanitailors.com here -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at:

Re: [asterisk-users] Confbridge

terisk-users [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of John T. Bittner Sent: Saturday, August 8, 2020 12:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion‏ Subject: [asterisk-users] Confbridge To all: No matter what I try, I cannot get the system to wait for the a

[asterisk-users] Confbridge

To all: No matter what I try, I cannot get the system to wait for the admin to join. It just dumps users into the bridge directly. I do not have a pin for users, does that matter? What am I missing? Another issue the absolute timeout is not working ? ... have recordings that last for over 24

Re: [asterisk-users] ConfBridge and sound prompts

On Tue, Oct 22, 2019, at 6:01 PM, Dan Cropp wrote: > > Just to add additional information, it seems this approach works with > the CONFBRIDGE user variables just not the bridge variables… > > > Action: SetVar^M > > ActionID: C81^M > > Channel: PJSIP/1003-0003^M > > Variable:

Re: [asterisk-users] ConfBridge and sound prompts

is working. Did the naming for the CONFBRIDGE bridge variables changed? Dan From: asterisk-users On Behalf Of Dan Cropp Sent: Tuesday, October 22, 2019 3:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ConfBridge and sound prompts We have a product that uses Asterisk via AMI. I am

[asterisk-users] ConfBridge and sound prompts

We have a product that uses Asterisk via AMI. I am relatively certain we used to be able to play prompts by actions like the following to make asterisk play the confbridge-join prompt when a new user joins the confbridge. However, that doesn't seem to work now. Action: SetVar ActionID: C58

Re: [asterisk-users] ConfBridge audio issues

On Behalf Of Antony Stone Sent: Monday, August 5, 2019 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] ConfBridge audio issues On Monday 05 August 2019 at 19:54:50, Dan Cropp wrote: > Channel B is the first to speak. Everything seems perfec

Re: [asterisk-users] ConfBridge audio issues

On Monday 05 August 2019 at 19:54:50, Dan Cropp wrote: > Channel B is the first to speak. Everything seems perfectly fine. Channel A > hears it well and ConfBridge recording sounds good. Then, channel B replies. Um, did Channel A say something in the meantime? Or did you mean "Channel A

[asterisk-users] ConfBridge audio issues

We have a system where two calls are in a ConfBridge with recording. This is Asterisk 16.3.0 Channel A seems to work perfectly. Wireshark is showing the RTP to/from working fine and having no jitter/lag issues. This call hears everything from channel B. Channel B we have more issues

[asterisk-users] ConfBridge audio issues (chan_sip)

We have a customer using ConfBridges. Party A is connected, audio is fine. We originate a call to party B through an Avaya switch. It forwards the call to IVR. The two channels are added to the same ConfBridge. Using a wireshark capture, I can listen to the audio for both channels. Initially,

[asterisk-users] ConfBridge: Identifying troublemakers

Hi list, imagine a ConfBridge conference with 10 participants. Now, one of them suddenly starts to yell and scream. Is there any built-in functionality (maybe not in ConfBridge, but in Asterisk itself) to identify the "loudest" caller? Or maybe already built-in functionality to

Re: [asterisk-users] Confbridge GUI?

> >> If you can provide details, even vague ones, about how you did it, I > >> can update the WMM package. > > > > See http://asterisk.gnat.com/meetme.tgz > > > > That's a gzipped tar of our working directory plus the relevant parts of > > extensions.conf. I xxx'ed out phone numbers and Google

Re: [asterisk-users] Confbridge GUI?

Hi, > On 17 Oct 2017, at 18:30, Richard Kenner wrote: > >> If you can provide details, even vague ones, about how you did it, I >> can update the WMM package. > > See http://asterisk.gnat.com/meetme.tgz > > That's a gzipped tar of our working directory plus the relevant

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On 11/15/17 11:36 AM, Joshua Colp wrote: On Wed, Nov 15, 2017, at 01:30 PM, Carlos Chavez wrote: Here is more information from the browser about the session: https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/af36iLlljtbYkbF On Asterisk I have icesupport=true in rtp.conf and

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On Wed, Nov 15, 2017, at 01:30 PM, Carlos Chavez wrote: > Here is more information from the browser about the session: > https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/af36iLlljtbYkbF > > On Asterisk I have icesupport=true in rtp.conf and ice_support=yes on the > endpoint. I have

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On 11/15/17 11:10 AM, Joshua Colp wrote: On Wed, Nov 15, 2017, at 01:05 PM, Carlos Chavez wrote: On 11/14/17 5:23 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote: Trace with 3 clients. We can hear each other but no video.

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On Wed, Nov 15, 2017, at 01:05 PM, Carlos Chavez wrote: > On 11/14/17 5:23 PM, Joshua Colp wrote: > > > On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote: > >> Trace with 3 clients. We can hear each other but no video. > >> > >>

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On 11/14/17 5:23 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote: Trace with 3 clients. We can hear each other but no video. https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz Do you see anything in the Javascript console of the browser?

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On Tue, Nov 14, 2017, at 07:19 PM, Carlos Chavez wrote: > Trace with 3 clients. We can hear each other but no video. > > https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz Do you see anything in the Javascript console of the browser? We are adding the needed media streams

Re: [asterisk-users] Confbridge SFU for Asterisk 15

Trace with 3 clients. We can hear each other but no video. https://pbxoficina.telecomabmex.com/nextcloud/index.php/s/X0PQ5FrYeqCwwkz On 11/14/17 5:06 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 07:03 PM, Carlos Chavez wrote: On 11/14/17 4:27 PM, Joshua Colp wrote: On Tue, Nov 14,

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On Tue, Nov 14, 2017, at 07:03 PM, Carlos Chavez wrote: > On 11/14/17 4:27 PM, Joshua Colp wrote: > > > On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: > >> On 11/14/17 3:55 PM, Joshua Colp wrote: > >> > >>> On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: > I followed

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On 11/14/17 4:27 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: On 11/14/17 3:55 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: I followed the blog post and I can get video from the conference if I configure the bridge

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On Tue, Nov 14, 2017, at 06:25 PM, Carlos Chavez wrote: > On 11/14/17 3:55 PM, Joshua Colp wrote: > > > On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: > >> I followed the blog post and I can get video from the conference if > >> I configure the bridge as follow_talker so I know

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On 11/14/17 3:55 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: I followed the blog post and I can get video from the conference if I configure the bridge as follow_talker so I know everything is working on the pjsip side. The only problem is that

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On Tue, Nov 14, 2017, at 05:47 PM, Carlos Chavez wrote: > I followed the blog post and I can get video from the conference if > I configure the bridge as follow_talker so I know everything is working > on the pjsip side. The only problem is that video_mode = sfu is > apparently not valid

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On 11/14/17 3:38 PM, Joshua Colp wrote: On Tue, Nov 14, 2017, at 05:23 PM, Carlos Chavez wrote: I am trying to get the "Mega Phone" demo working on my office PBX but there seems to be a problem when trying to set the default bridge to sfu mode. I have the following configuration in

Re: [asterisk-users] Confbridge SFU for Asterisk 15

On Tue, Nov 14, 2017, at 05:23 PM, Carlos Chavez wrote: > I am trying to get the "Mega Phone" demo working on my office PBX > but there seems to be a problem when trying to set the default bridge to > sfu mode. I have the following configuration in confbridge.conf in the > default_bridge

[asterisk-users] Confbridge SFU for Asterisk 15

I am trying to get the "Mega Phone" demo working on my office PBX but there seems to be a problem when trying to set the default bridge to sfu mode. I have the following configuration in confbridge.conf in the default_bridge section: video_mode = sfu but when I do a "confbridge show

Re: [asterisk-users] Confbridge GUI?

> If you can provide details, even vague ones, about how you did it, I > can update the WMM package. See http://asterisk.gnat.com/meetme.tgz That's a gzipped tar of our working directory plus the relevant parts of extensions.conf. I xxx'ed out phone numbers and Google interface data. This

Re: [asterisk-users] Confbridge GUI?

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard Kenner Sent: Friday, October 13, 2017 2:14 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Confbridge GUI? > I have a very old ser

Re: [asterisk-users] Confbridge GUI?

> I have a very old server that is used only for conferences on > Meetme. To manage the conference rooms we use Web Meetme. Now it is > time to upgrade everything but since Meetme is no longer available I > need to find a replacement GUI to manage the conference rooms. Anyone > know a

[asterisk-users] Confbridge GUI?

I have a very old server that is used only for conferences on Meetme. To manage the conference rooms we use Web Meetme. Now it is time to upgrade everything but since Meetme is no longer available I need to find a replacement GUI to manage the conference rooms. Anyone know a solution

[asterisk-users] ConfBridge increase talking volume as standard

Hello, is it possible to increase talking volume for caller in ConfBridge as standard without need to press buttons after joining an conference room. best regards Thomas -- _ -- Bandwidth and Colocation Provided by

[asterisk-users] Confbridge or Bridge

Hi, an agent should wait in an confroom and hear some music or tones. On an website he should put in an tel number for calling to somebody. This is working wih phpagi very well. I have problems with Agent should hear ringing when callee is called, busy for may be 5 s that callee is busy or

Re: [asterisk-users] ConfBridge function slight change from 11 to 13

On Wed, Mar 29, 2017 at 9:40 AM, Michaël Gaudette wrote: > Hi, > > > > I have been using ConfBridge since Asterisk 11, and I recently upgraded a > server to 13. While everything that needed fixing seems fixed, I have an > issue that does not seem documented anywhere. > > >

[asterisk-users] ConfBridge function slight change from 11 to 13

Hi, I have been using ConfBridge since Asterisk 11, and I recently upgraded a server to 13. While everything that needed fixing seems fixed, I have an issue that does not seem documented anywhere. The way I used ConfBridge is that I have a standard bridge profile, user profile and menu that

Re: [asterisk-users] confbridge setup

Hello, Thanks for getting back to me. I didn't know that the conferences wouldn't show up on the list until they were "active;" I thought that was meant to show the defined conferences. However, when I try to dial into the conference room that I (think) have defined, I see: -- [Apr 18

Re: [asterisk-users] confbridge setup

You can't see them until someone joins the bridge, might be able to put in db using the asterisk live setup On Apr 16, 2016 1:36 PM, "Mike Diehl" wrote: > Hi all, > > I'm trying to configure a few conference bridges. I've started with the > very > basic: > > [general] >

[asterisk-users] confbridge setup

Hi all, I'm trying to configure a few conference bridges. I've started with the very basic: [general] [default_bridge] type=bridge [default_user] type=user [default_bridge] type=bridge [5340] type=bridge However: confbridge list Conference Bridge Name Users Marked Locked?

Re: [asterisk-users] ConfBridge play message to all in conf

To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] ConfBridge play message to all in conf I would like to figure out using confbridge how to play a file after the conf is built. not really a per user thing - just conf is up and ready and need to play a file to all

[asterisk-users] ConfBridge play message to all in conf

I would like to figure out using confbridge how to play a file after the conf is built. not really a per user thing - just conf is up and ready and need to play a file to all in the conference. (I am creating my conf on the fly and bringing in other devices on the fly). How can that be

[asterisk-users] confbridge play tone before speaking

I use both confbridge to bring several devices into a receive only or listen mode, then allow the one person on the phone to speak live over those devices. Works great. However - now I would like to be able to play a tone into the conference before the person speaks. How might that be

Re: [asterisk-users] Confbridge

Take a look here: http://asteriskfaqs.org/tag/confbridge/page/2 Am 02.12.2014 03:37, schrieb Bryant Zimmerman: I am doing dynamic conference bridges using confbridge in asterisk 11. Is there a way to toggle off an on recording of an ongoing conference call I have figured out how to record a

Re: [asterisk-users] Confbridge

I am doing dynamic conference bridges using confbridge in asterisk 11. Is there a way to toggle off an on recording of an ongoing conference call I have figured out how to record a conference if it is turned on when someone enters. Also I have noticed that when setting music_on_hold_class

[asterisk-users] ConfBridge / internal_sample_rate=auto / warning

Hi there, I am running Asterisk 11.9.0 WANPIPE Release: 7.0.10 DAHDI Version: 2.9.0 Echo Canceller: HWEC libpri version: 1.4.12 When I start the ConfBridge application I get the following warning: [2014-10-24 14:36:21] WARNING[29177][C-6934]: config_options.c:790 uint_handler_fn: Attempted

[asterisk-users] Confbridge options

Hi, I'm doing an evaluation of Confbridge (migrating from Meetme). Looking at: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 Under the heading User Profile Configuration Options the option announce_only_user is present. The sample config looks like this: -- ;announce_only_user=yes

Re: [asterisk-users] Confbridge options

2014-04-04 22:01, Johan Wilfer skrev: Hi, I'm doing an evaluation of Confbridge (migrating from Meetme). Looking at: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 Under the heading User Profile Configuration Options the option announce_only_user is present. The sample config looks

Re: [asterisk-users] Confbridge options

2014-04-04 23:33, Johan Wilfer skrev: Also - setting quiet=yes still plays join/leave sound. My current work-around is: sound_join=silence/1 sound_leave=silence/1 But this seems a bit ineffective... In Meetme the quiet-flag also disabled join/leave sounds. Is this by design or an oversight?

[asterisk-users] ConfBridge 11.7 and 11.8

11.7 is working fine for me. I put on 11.8 and my confbridge that should be muted to users now sounds un-muted or like I am getting feedback. Did something change there??? Jerry -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] ConfBridge speak wave file in conf

On 15/02/14 20:05, Jerry Geis wrote: I have a confbridge in asterisk 11. I am using an AGI to bring people in the conf automatically. I want to speak a pre-recorded wave file message into the conf to all users. how might I do that? Thanks, Jerry You could initiate a call which would

[asterisk-users] ConfBridge speak wave file in conf

I have a confbridge in asterisk 11. I am using an AGI to bring people in the conf automatically. I want to speak a pre-recorded wave file message into the conf to all users. how might I do that? Thanks, Jerry -- _ --

[asterisk-users] ConfBridge on asterisk 11

I believe I am running an AGI (to put users in a conf) before the confbridge is built. So the users are not really get in the conf... exten X,1,run agi to put users in conf exten X,n,ConfBridge() How do I have in the dial plan ConfBridge() and someplace run an AGI that brings the users I want

Re: [asterisk-users] ConfBridge on asterisk 11

On Friday 14 Feb 2014, Jerry Geis wrote: I believe I am running an AGI (to put users in a conf) before the confbridge is built. So the users are not really get in the conf... exten X,1,run agi to put users in conf exten X,n,ConfBridge() How do I have in the dial plan ConfBridge() and

[asterisk-users] confbridge - play different sounds to caller and bridge at same time?

When a caller enters the confbridge, I want to play a sound file ('ring') for the caller and a different sound file ('type of caller') to the bridge (all participants or just the admin?) at the same time. It's OK if the bridge hears the 'ring,' but the caller should not hear the 'type of

[asterisk-users] Confbridge doesn't kick chan_local

I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial 1000_admin@conferences I join the ices user to the confbridge

Re: [asterisk-users] Confbridge doesn't kick chan_local

On 06/03/2013 08:11 AM, Chris Gentle wrote: I have a confbridge setup that feeds the conference from the ALSA microphone input (this is the conference leader) and then uses app_ices to send the conference audio to icecast. I start the conference leader like this: console dial

Re: [asterisk-users] Confbridge doesn't kick chan_local

On Mon, Jun 3, 2013 at 11:52 AM, Matthew Jordan mjor...@digium.com wrote: (1) Verify that with all 'normal' channel drivers, such as chan_sip, that the Conference tears down correctly. OK, looks like this is the problem. Taking chan_local out of the picture, I tested it with an incoming SIP

Re: [asterisk-users] Confbridge doesn't kick chan_local

On 06/03/2013 01:03 PM, Chris Gentle wrote: On Mon, Jun 3, 2013 at 11:52 AM, Matthew Jordan mjor...@digium.com wrote: snip If both (1) and (2) are successful, than there's some impact that the Ices application is having on the Local channel that is messing up the reference counting inside

[asterisk-users] Confbridge Dynamic video_mode

Hi All, I want to set the video_mode of the confbridge dynamically in the dialplan. SO say if 5 users join the conference with follow_talker mode, it should work like that (and it does). But if 6th user changes the video_mode to first_marked and gets marked in the dial plan and joins the

Re: [asterisk-users] confbridge and talker

11.02.2013 18:19, Matthew Jordan пишет: On 02/11/2013 01:20 AM, Dmitry Melekhov wrote: Hello! We use meetme, but, as I understand it will be soon removed from asterisk (already marked as deprecated), so I'm thinking about confbridge migration. Really, we use self-developed (really my ;-) )

Re: [asterisk-users] ConfBridge performance problem...?

Hi again, I did a try on my asterisk 11.2.1 compiled on Ubuntu 12.04 (64 bit) with a simple Pentium 4 CPU (Intel(R) Pentium(R) D CPU 2.80GHz). I connected 5 SIP-Users with a ConfBridge. This is my picture: Please give a a hint where I can

Re: [asterisk-users] confbridge and talker

On 02/11/2013 01:20 AM, Dmitry Melekhov wrote: Hello! We use meetme, but, as I understand it will be soon removed from asterisk (already marked as deprecated), so I'm thinking about confbridge migration. Really, we use self-developed (really my ;-) ) web interface to control meetme. We

[asterisk-users] confbridge and talker

Hello! We use meetme, but, as I understand it will be soon removed from asterisk (already marked as deprecated), so I'm thinking about confbridge migration. Really, we use self-developed (really my ;-) ) web interface to control meetme. We use cli ( over manager ) command to get users list

Re: [asterisk-users] ConfBridge performance problem...?

Hi, perhaps it is a problem with your Host-Guest-Setup? Did you try the Asterisk-Setup on a dedicated server without virtualization? -Thorsten- Am 07.02.2013 11:42, schrieb Hristo Trendev: Hi Thorsten, Thanks for your reply. I did check core show translations, but the following

Re: [asterisk-users] ConfBridge performance problem...?

Hi, perhaps it is a problem with your Host-Guest-Setup? Did you try the Asterisk-Setup on a dedicated server without virtualization? -Thorsten- Am 07.02.2013 11:42, schrieb Hristo Trendev: Hi Thorsten, Thanks for your reply. I did check core show translations, but the following

[asterisk-users] ConfBridge performance problem...?

Hi, I have been experimenting with ConfBridge from the asterisk-11 stable SVN branch (and with 11.2.0 also) for the last 3 weeks and I see a problem, which what I believe is performance related. I just wanted to ask if someone else has made any tests and what is the maximum number of participants

Re: [asterisk-users] ConfBridge performance problem...?

Did you watch the cpu usage (for example with top)? You have a board installed which does use dahdi? Did you check the command dahdi_test? Maybe a (performance) problem of the software ec? Am 06.02.2013 11:13, schrieb Hristo Trendev: Hi, I have been experimenting with ConfBridge from the

Re: [asterisk-users] ConfBridge performance problem...?

Sorry - I just read you alsways checked the cpu usage. Are all cores at 100%? Is it the atserisk process which consumes it all? Am 06.02.2013 13:54, schrieb Thorsten Göllner: Did you watch the cpu usage (for example with top)? You have a board installed which does use dahdi? Did you check the

Re: [asterisk-users] ConfBridge performance problem...?

Did you check asterisk -rx core show translation recalc 10 Am 06.02.2013 13:56, schrieb Thorsten Göllner: Sorry - I just read you alsways checked the cpu usage. Are all cores at 100%? Is it the atserisk process which consumes it all? Am 06.02.2013 13:54, schrieb Thorsten Göllner: Did you

[asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?

Hi list! ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF gets transmitted throughout the conference. I've tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source. I've confirmed that it's disabled via the CLI confbridge show profile user

Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?

On 28/09/12 06:50 AM, Markus wrote: Hi list! ConfBridge dtmf_passthrough=no doesn't seem to have any effect. DTMF gets transmitted throughout the conference. I've tried Asterisk 10.7.1 from the official RPMs and 10.8.0 compiled from source. I've confirmed that it's disabled via the CLI

Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?

Hi Leif, Am 28.09.2012 13:24, schrieb Leif Madsen: Searching the issue tracker (hint, hint) does not return any dtmf_passthrough issues other than this one[0], which doesn't look to be related. thanks for your reply. Right, doesn't look related. Is another channel connected to the

Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?

On 28/09/12 07:36 AM, Markus wrote: Am 28.09.2012 13:24, schrieb Leif Madsen: Is another channel connected to the conference receiving the DTMF? Is that what you're intending? Because from my understand that is the intention, and not simply to limit the DTMF from being in the conference in the

Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?

Hi again Leif, Am 28.09.2012 13:42, schrieb Leif Madsen: OH! I just tested with a SIP softphone (X-Lite), and DTMF does not get passed to the other users! In X-Lite I can hear the DTMF keypresses of the users connected via PSTN (incoming via SIP), but when I hit a key in X-Lite I can't hear

Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?

Markus wrote: Snipped long results list PSTN means that I've tested two times, from a regular landline and from a mobile. Always calling to the providers DID which ends up in Asterisk via SIP. In the case of ConfBridge there were always 2 participants in the conference so that I could check if

Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?

Hi Joshua, Am 28.09.2012 15:56, schrieb Joshua Colp: I think your results are sort of skewed. In the case of SIP - SIP if a local bridge occurs things will optimize and you most likely won't see DTMF related messages. They get passed through as packets and not fully interpreted. ah, ok! That

Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?

Markus wrote: Hi Joshua, Hola, My suggestion is to take a step back further. Just send incoming calls to the Read application and have it store the received DTMF in a variable. Next step have it output what was received. Ok, good idea, here are the results of Read() and SayDigits():

Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?

Am 28.09.2012 17:33, schrieb Joshua Colp: Ok, good idea, here are the results of Read() and SayDigits(): snipped results to make this email manageable How are you changing the DTMF for each provider? If you are merely changing it using dtmfmode in sip.conf this may or may not change how the

Re: [asterisk-users] ConfBridge dtmf_passthrough=no doesn't have any effect. Bug?

Markus wrote: Am 28.09.2012 17:33, schrieb Joshua Colp: Ok, good idea, here are the results of Read() and SayDigits(): snipped results to make this email manageable How are you changing the DTMF for each provider? If you are merely changing it using dtmfmode in sip.conf this may or may not

[asterisk-users] confbridge video support

Where does video support for confbridge stand? I need to be able to take in multiple video callers and have the active speaking caller displayed to all participants. Are we there yet in Asterisk 10 or 11? Thanks Bryant -- _

Re: [asterisk-users] confbridge video support

On 25/09/12 10:18 AM, Bryant Zimmerman wrote: Where does video support for confbridge stand? I need to be able to take in multiple video callers and have the active speaking caller displayed to all participants. Are we there yet in Asterisk 10 or 11? Yes, both Asterisk 10 and 11 support this

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