search for: ronaldmwheeler

...ty: > > - Is it recommended to upgrade kernel version(s) (ie moving from linux > 3.10 to 4.3) just after OS installation ? > > Best regards > > > > > -- > Ron Wheeler > President > Artifact Software Inc > email: rwheeler at artifact-software.com > skype: ronaldmwheeler > phone: 866-970-2435, ext 102 <(866)%20970-2435> > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: https://communi...

...version(s) (ie moving from linux >> 3.10 to 4.3) just after OS installation ? >> >> Best regards >> >> >> >> >> -- >> Ron Wheeler >> President >> Artifact Software Inc >> email: rwheeler at artifact-software.com >> skype: ronaldmwheeler >> phone: 866-970-2435, ext 102 <%28866%29%20970-2435> >> >> >> -- >> _____________________________________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> Check out the new Asterisk...

...ts such as the one from Camrivox that interfaces with different CRM packages or Outlook, but I was wondering if a way was available to show the calls on their phones. Thanks in Advance; John V. -- Ron Wheeler President Artifact Software Inc email: rwheeler at artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20171211/bf48eb2a/attachment.html>

...empt at interoperability is truly shocking. > > For connecting two Asterisk installations to each other over the Internet, IAX > is better than SIP -- that's what it was designed for. > -- Ron Wheeler President Artifact Software Inc email: rwheeler at artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102

...ne rings, the Asterisk PBX has already answered the call and starts the "You have reached..... Dial ...." and tries to dispatch the call. This makes it hard to carry on a conversation. Ron -- Ron Wheeler President Artifact Software Inc email: rwheeler at artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102

Hello, I'm used to install Asterisk on Debian stable platforms. A customer is asking how I would proceed on a CentOS platform. After a short research (see [1] as an example), I'm wondering what are general kernel practices on CentOS regarding Asterisk and when targeting stability: - Is it recommended to upgrade kernel version(s) (ie moving from linux 3.10 to 4.3) just after OS

On Tue, Mar 14, 2017 at 06:03:33PM +0100, Jean Aunis wrote: > Hello, > > Did you disable selinux ? It usually causes troubles when starting asterisk > as a service. You can do this with : setenforce 0 (this will not totally > disable selinux, but switch it to a permissive mode). Generally before advising that, check if this is the error: tail -f /var/log/audit/audit.log and

All; I have an interesting scenario where I have a small office with maybe half a dozen phones and several incoming lines. The calls are routed based on the DID that people call. What they would like is when a call comes in to a single phone to have all the phones show the CallerID. That way they can decide if they should pick up the call or not using call pickup. I've been looking at

...For connecting two Asterisk installations to each other over the > Internet, IAX > > is better than SIP -- that's what it was designed for. > > > > > -- > Ron Wheeler > President > Artifact Software Inc > email: rwheeler at artifact-software.com > skype: ronaldmwheeler > phone: 866-970-2435, ext 102 > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asteri...

I was wondering if anyone can give me any pointers or insights of whether or not to have an asterisk server behind a firewall. I have always ran Asterisk on a public IP but was wondering if I should move it to a local IP behind a firewall. I am looking to set up a location with 300 SIP phones. Normally, I would put the Asterisk server on one public IP and let the SIP phones get DHCP from a

Is anyone using something to log SIP results (connected/not, latency) that they really like? We do some logging using simple scripts writing the results of sip show peers to a text file if customers report issues, but it would be nice to have a tool that logs all the time and lets us do some better reporting. For example, graphs of latency in a time range, or a list of unreachable phones within

Hi. i am running asterisk in a low powered machine (alix2d13 from pcengines) without any gui. the machine works fine to route all my calls for the office. the problem is the management of the CDRs. i can see the master.csv file, but it is not very friendly for the secretary of this office to manage the calls. is there a way to have a nice way to see the CDRs?Since the machine is very small on

...ut > it still stops every few days or so. I have a cron job that tests for > it and restarts it when necessary. > > Anyone else have experience on non-Linux systems? > > Cheers. > -- Ron Wheeler President Artifact Software Inc email: rwheeler at artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102

Any particular reason CentOS 7 repos aren't available? I'm finding integration issues with CentOS 6's ancient versions of MySQL and PHP with third party applications. -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Hello, we plan to move a PBX to asterisk and searching for Support and a Phonehardware Reseller in Germany. The should be no license costs per User / Server. - Install Configure Asterisk for our specification - Install FreePBX or similar (optional) - Resell Hardware Thanks for any suggest. Best Regards, basti

I know but this is not my sole decision. On 12.12.2017 16:17, Ron Wheeler wrote: > If your phone system goes down and you can not get it back up until > tomorrow afternoon because your support person is on another project, > you may wish you had an SLA.

...the data files, produce nice looking reports and automatically upload them to a portal might be helpful. I would be grateful for any comments about content, format or my skill as an artist.:-) Ron -- Ron Wheeler President Artifact Software Inc email: rwheeler at artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102

I wonder if anybody know how to hire Alice or some professional voice-artist. I need to record 12 messages for a customer.

Size: - one location - 15 IP Phones ( 1 dect) - Create new voip trunk (current are ISDN) (30 number block) - LTS is important - an SLA is optional at the moment there is no one On 11.12.2017 22:31, Ron Wheeler wrote: > You might want to add some details > - size of the project > ?-- number of locations > ?-- number of extensions > - are you converting your trunks? > - what are

...tempt at interoperability is truly shocking. > > For connecting two Asterisk installations to each other over the Internet, IAX > is better than SIP -- that's what it was designed for. > -- Ron Wheeler President Artifact Software Inc email: rwheeler at artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing li...