similar to: confBridge in Asterisk 1.6.2.0-rc1 doesn't stable

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 video_mode = sfu is >> apparently not valid in either confbridge.conf or

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 as follow_talker so I know everything

Hi guys, I'm trying to move away from meetme to loose the dependency on dahdi. ConfBridge seems to be a good fit but I can't get it going. The document sounds like an easy to use app. Am I missing any bridge_ modules? Asterisk 1.6.2.0~rc2-0ubuntu1.2 -- Executing [521 at outbound:1] Answer("SIP/109-b877a8c8", "") in new stack -- Executing [521 at

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 confbridge.conf in the >> default_bridge section: video_mode

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 profile bridge default_bridge" I see: Video Mode: no video I can change it

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 Channel: PJSIP/1003-00000003 Variable: CONFBRIDGE(bridge,sound_join) Value: en/confbridge-join Does

Formerly on a thread called [asterisk-dev] Where to find the code of application Bridge On Wed, May 6, 2009 at 7:38 AM, Tzafrir Cohen <tzafrir.cohen at xorcom.com> wrote: >> Can someone please tell me in which file the code for the application to >> be found? I was not able to find a file named app_bridge.c in the folder >> apps. > > app_bridge.c ? app_confbridge.c ?

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 <profilename>". It's an all-SIP scenario with RFC2833 as the DTMF protocol.

We are looking to migrate from AMI to ARI. We currently rely heavily on ConfBridges for multiple party support. Is it possible to add more than 2 channels? If so, is there a limit? Or a way to configure the limit? Have a great day! Dan -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 at conferences I join the ices user to the confbridge with a call file: Channel: Local/1000 at conferences MaxRetries: 2 RetryTime: 60 WaitTime: 30

I'm looking at what would be involved in converting from MeetMe to ConfBridge and there seems to be a lot of missing administrative things, but I hope I'm just missing it. We all know about the missing realtime linkage. That's a major nuisance, but can be worked around. More serious is that the CLI command to display users in a ConfBridge don't show the caller ID information, so

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All Asterisk users are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also

The Asterisk Development Team is pleased to announce the first release candidate of Asterisk 10.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All Asterisk users are encouraged to participate in the Asterisk 10 testing process. Please report any issues found to the issue tracker, https://issues.asterisk.org/jira. It is also

Thanks for your (fast) reply ! Le ven. 18 janv. 2019 à 16:32, Joshua C. Colp <jcolp at digium.com> a écrit : > On Fri, Jan 18, 2019, at 11:22 AM, Olivier wrote: > > Hello, > > > > I've just read in [1] about SIP MESSAGE addition to both chan_pjsip and > > ConfBridge. > > It seems very interesting addition as it brings the capability to mix > >

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, 2017, at 06:25 PM, Carlos Chavez wrote: >>>> On 11/14/17 3:55

Hello, We use Asterisk extensively for conferencing - for the last 8 years or so this has been the 1.4/1.6/1.8 releases running chan_sip and meetme for up to around 350 concurrent users. Right around that number DAHDI hit's a hard coded memory limit and kicks allocation errors in the log. [Jun 22 10:04:13] WARNING[9095] app_meetme.c: Unable to open DAHDI pseudo channel: Cannot allocate

I'm trying to migrate from MeetMe to ConfBridge: [conferences] exten=>_8[1-9],1,Answer() ;;exten=>_8[1-9],1,Meetme(${EXTEN},1Ms,1234) exten=>_8[1-9],2,ConfBridge(${EXTEN},1Ms) exten=>_8[1-9],n,Hangup And that works. Also changed the hints: ;;exten => 81,hint,MeetMe:81 exten => 81,hint,ConfBridge:81 ;;exten => 82,hint,MeetMe:82 exten => 82,hint,ConfBridge:82 ;;exten

Le 2017-07-05 18:51, Joshua Colp a ?crit : > On Wed, Jul 5, 2017, at 01:45 PM, Jean Aunis wrote: > >> Hello, I am struggling with a problem which I thought would be an easy one : bridging several channels together in a *smart* bridge. I emphasize *smart* : I want my bridge to be a native_rtp one when only two channels are involved, and switch to softmix technology when a third

running Asterisk 1.8.9.0-rc2, what are the ways to interface with ConfBridge ? I see the CLI command 'confbridge' documented for asterisk 10, but i dont see how to interface with confbridge on 1.8 What I'm trying to do is keep track of conferences that are used. I tried something like the below, but not only does Confbridge not return, but i'd need something that erases the

Hello, I've just read in [1] about SIP MESSAGE addition to both chan_pjsip and ConfBridge. It seems very interesting addition as it brings the capability to mix voice, video and text in conferencing. On an other hand, there are some softphones (Zoiper, Bria, ...) that tout voice, video and chat capability. Though Enhanced Messaging solution described in [1] seems more attractive to me in the