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On Wed, Sep 12, 2018 at 6:59 AM, Martin Kletzander <mkletzan@redhat.com> wrote: > On Mon, Sep 10, 2018 at 02:38:48PM -0400, David Vossel wrote: > >> On Wed, Aug 29, 2018 at 4:55 AM, Daniel P. Berrangé <berrange@redhat.com> >> wrote: >> >> On Tue, Aug 28, 2018 at 05:07:18PM -0400, David Vossel wrote: >>> > Hey, >>> > >>> > Over in KubeVirt we're investigating a...

On Fri, Sep 14, 2018 at 6:55 PM David Vossel <dvossel@redhat.com> wrote: > > > On Wed, Sep 12, 2018 at 6:59 AM, Martin Kletzander <mkletzan@redhat.com> > wrote: > >> On Mon, Sep 10, 2018 at 02:38:48PM -0400, David Vossel wrote: >> >>> On Wed, Aug 29, 2018 at 4:55 AM, Daniel P. Berrangé <berr...

...ew.php?id=19165. If you start using the ConfBridge application and find that you are interested in writing a new feature for it, feel free to use me as a resource by email or IRC. I'm happy to review your code and anything else I can to do make this application successful. Thanks! -- David Vossel Digium, Inc. | Software Developer, Open Source Software 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com & www.asterisk.org The_Boy_Wonder in #asterisk-dev

On Wed, Aug 29, 2018 at 4:55 AM, Daniel P. Berrangé <berrange@redhat.com> wrote: > On Tue, Aug 28, 2018 at 05:07:18PM -0400, David Vossel wrote: > > Hey, > > > > Over in KubeVirt we're investigating a use case where we'd like to > perform > > a live migration within a network namespace that does not provide > libvirtd > > with network access. In this scenario we would like to perform a live...

Running build 1.8.5.0 (compiled from source) I seem to be having an issue with codec negotiation. I have a Grandstream HT503 FXO port connected to a pstn line, a Polycom SP501, and a SIP trunk with callwithus. What I'm essentially looking to accomplish is for ulaw or g729 (preferably ulaw) to be used to the Grandstream FXO or any other internal endpoint, and for g729 only to be used outbound

On Mon, Sep 17, 2018 at 02:17:39PM +0200, Fabian Deutsch wrote: >On Fri, Sep 14, 2018 at 6:55 PM David Vossel <dvossel@redhat.com> wrote: >> Any chance we can get the safety check removed for the next Libvirt >> release? Does there need to be an issue opened to track this? >> > >Regardless of Martin's answer :): Please file one. >Please file an RFE requesting the change...

On Mon, Sep 10, 2018 at 02:38:48PM -0400, David Vossel wrote: >On Wed, Aug 29, 2018 at 4:55 AM, Daniel P. Berrangé <berrange@redhat.com> >wrote: > >> On Tue, Aug 28, 2018 at 05:07:18PM -0400, David Vossel wrote: >> > Hey, >> > >> > Over in KubeVirt we're investigating a use case where we'd like to...

Hey, Over in KubeVirt we're investigating a use case where we'd like to perform a live migration within a network namespace that does not provide libvirtd with network access. In this scenario we would like to perform a live migration by proxying the migration through a unix socket to a process in another network namespace that does have network access. That external process would live

On Tue, Aug 28, 2018 at 05:07:18PM -0400, David Vossel wrote: > Hey, > > Over in KubeVirt we're investigating a use case where we'd like to perform > a live migration within a network namespace that does not provide libvirtd > with network access. In this scenario we would like to perform a live > migration by proxying the mi...

Dear all, First off, I am very new to asterisk so forgive me if any of my comments or questions seem trivial. Thanks to [this post](http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/) and [this post](http://www.davidvossel.com/?p=28), I have GV set up on asterisk through jabber.conf and gtalk.conf. I can successfully dial out from asterisk. I'm trying to set up an auto-attendant on asterisk. I am doing a basic "Hello world" example. My config: jabber.conf: [general] debug=yes autoprune=no autoregis...

Hi, I am trying to use ConfBridge application, but it throws "Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw)" error. Please see console output below. -- Executing [501 at services:9] ConfBridge("SIP/OpenSER-00000005", "1001") in new stack [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404 join_conference_bridge: Trying to find conference

Hi, When starting Asterisk (1.8.5.0) I see in messages: [Jul 18 15:47:50] WARNING[15491] loader.c: Error loading module 'chan_gtalk.so': /usr/lib/asterisk/modules/chan_gtalk.so: undefined symbol: ast_aji_get_client [Jul 18 15:47:50] WARNING[15491] loader.c: Module 'chan_gtalk.so' could not be loaded. Yet I do have iksemel installed: ls -l /usr/local/lib/libik* -rw-r--r-- 1