similar to: asterisk.conf ignored?

Than you very much. I use asterisk 14, and yes, menuselect shows me the need for generic_odbc(E), res_odbc_transaction(M) and ltdl(E) but what does this imply under debian ? I have unixodbc installed an tested and too libltdl-dev ! But what am I missing ? On 04/19/2017 10:10 AM, Marcelo Terres wrote: > What version of Asterisk are you using? > > When I go to cdr_adaptative_odbc in

Thank you, but unixodbc and odbcinst are installed... end even unixodbc-dev But I get the same need for "generic odbc(E)". On 17/04/2017 10:48, Marcelo Terres wrote: > You need unixodbc and odbcinst packages too, to configure the odbc. > > []s > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br >

Hello. Anybody in the list is using this IP phone? Regards, Marcelo H. Terres <mhterres at gmail.com> IM: mhterres at jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres

Hello. Is someone here using VoIPmonitor? I am using just the sniffer and I found some pcap files that contain some odd streams. For example, I have a file with 3 streams, but the weird stuff is that 2 streams are the same (e.g., have the same source address and port and same destination address and port). Example: "Source Address","Source Port","Destination

One more thing about my last email: I think that you forgot to update the configs/samples/res_odbc.conf.sample file, because it still contains idlecheck and limit parameters. Regards, Marcelo H. Terres <mhterres at gmail.com> IM: mhterres at jabber.mundoopensource.com.br https://www.mundoopensource.com.br https://twitter.com/mhterres https://linkedin.com/in/marceloterres On Mon, Sep 19,

On Tue, Feb 14, 2017, at 09:57 AM, Marcelo Terres wrote: > Same problem with me. > > I downloaded the file in 2 different places and had the same error... An issue was filed for tracking this[1] and it will be resolved later today. [1] https://issues.asterisk.org/jira/browse/ASTERISK-26791 -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL

I want to use corosync and installed it via ubuntu repository. I guess there is a version 1 and 2 of corosync. For some reason ./configure for Asterisk (13) isn't recognizing I have corosync installed. I can't enable the res_corosync module in menuselect. Any ideas? Thanks! Travis -------------- next part -------------- An HTML attachment was scrubbed... URL:

yessir From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Marcelo Terres Sent: Wednesday, July 19, 2017 10:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] corosync and Asterisk 13 Did you installed the dev package? corosync-dev Marcelo H.

I use debian stretch and I have installed unixodbc-dev but I have a dependency on genreric_odbc in make menuselect What am I missing ? Is there an howto ? Thanks PX

The Asterisk Development Team has announced the first release candidate of Asterisk 14.0.0. This release candidate is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 14.0.0-rc1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues

I have an 14.04 server with Asterisk 13.5 that will recognize it. It has corosync and corosync-dev version 2.3.3 on it. The one I can?t get to recognize it is 16.04 ubuntu, with corosync and -dev version 2.3.5. Won?t be recognized by ./configure for Asterisk 13.17, 13.16, or Asterisk Certified 13.13. I REALLY need some help figuring this out. ? Thanks! From: asterisk-users-bounces at

Can anyone put light on whatsapp features and how it can be operated . What are the technology that need to be installed , Regards Yves -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160729/62da2470/attachment.html>

On 7/19/17 2:37 AM, Marcelo Terres wrote: > This is the pjsip library. > > Is it possible to you to update pjsip for the latest version to test > if it solves the problem? > > On 18 Jul 2017 3:52 pm, "Carlos Chavez" <cursor at telecomab.mx > <mailto:cursor at telecomab.mx>> wrote: > > I am getting frequent segfaults on a new Asterisk

Hello, I'm a faithful reader of this mailing list, for several years now. Lately, I'm receiving emails asking me to re-enable my list subscription due to "excessive bouncing". What does this exactly mean and why am I receiving this ? Beside re-enabling my subscription, what can I do to improve things ? Regards -------------- next part -------------- An HTML attachment was

Ditto; a Gmail issue? Andrew On 12 June 2017 at 16:00, Marcelo Terres <mhterres at gmail.com> wrote: > It is happening the same with me. > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres > https://linkedin.com/in/marceloterres > >

hi. i am running asterisk 11 and i am stuck with the feature codes. how do i setup them. Now the system has. PBX*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # # Attended Transfer One Touch Monitor Disconnect Call * * Park Call One Touch MixMonitor Dynamic Feature Default Current --------------- ------- ------- (none) Feature

Me too, also gmail. I emailed the list owner a couple of days ago, but no reply. Is everyone else affected also forwarding to another email address (gmail or not)? Could be wrong, but I'm guessing there may be an incorrect DMARC policy somewhere - although this is the only fail I could find in the headers. bounces at lists.digium.com; dmarc=fail (p=NONE sp=NONE dis=NONE)

Hi Marcelo, Thanks for replying, I do not know what this branch is. Could you please let me know. Also, I enabled google cloud speech API only from the console. Do I need more API enabled? On Wed, Jul 19, 2017 at 3:41 PM, Marcelo Terres <mhterres at gmail.com> wrote: > Did you already tried the cloud_api branch? > > Regards, > > Marcelo H. Terres <mhterres at

Hello in sip.conf I have ; videosupport=yes Kind regards. J. On 20-04-17 13:09, Marcelo Terres wrote: > I suppose that you enable the video support on sip.conf, right? > > Regards, > Marcelo H. Terres <mhterres at gmail.com> > IM: mhterres at jabber.mundoopensource.com.br > https://www.mundoopensource.com.br > https://twitter.com/mhterres >

Hello, I'm familiar with Pickup/PickupChan for taking a ringing call, but does anyone know how a phone can "steal" an already answered call from another phone? Our users have decided that call parking is too long-winded and don't want to use that. For example: phone A calls phone B, phone B answers the call, phone C dials something to "steal" the call from B, and