similar to: Reporting Utility

Olivier If you installed asterisk from source, you need to recompile it after kernel version upgrade. This will compile & install asterisk modules with latest installed kernel sources. -- regards, abdul basit On 19 December 2017 at 08:01, Ron Wheeler <rwheeler at artifact-software.com> wrote: > Linux x.y.com 3.10.0-693.5.2.el7.x86_64 #1 SMP Fri Oct 20 20:32:50 UTC > 2017

Sorry for the empty message. Pressed the wrong button. I have been wrestling with a pretty generic Asterisk configuration (version 11.11.0 ) set up with FreePBX. The trunk SIP is setup to allow ulaw,alaw,gsm, Video is disabled. I was using Eyebeam and am now trying Jitsi. Jitsi has a number of codecs enabled - opus, SILK, G722, speex,PCMU, PCMA, iLBC, GSM, G723 and telephone-event The

Hello Ron, Which kernel do you run Asterisk/Freepbx with ? Cheers 2017-12-14 16:57 GMT+01:00 Ron Wheeler <rwheeler at artifact-software.com>: > CentOS 7 works well with Asterisk. > Install latest CentOS7 with updates install asterisk > > I am running FreePBX on CentOS 7. > > Ron > > On 14/12/2017 10:38 AM, Olivier wrote: > > Hello, > > I'm used to

Hey all We have been working with SIP for years. It has the potential to be better than Skype. It is really all in the implementation. Not all SIP soft clients are equal nor are the networks and computers they are running on. I will not bash Skype. We have tested it and in most cases choose not to use it. It has it's place and is good for the user that meets it's specific target

I have connected a PSTN line to a Digium FXO card. There is also an ordinary analogue phone attached to the same line. The Asterisk answers the line on the first ring. I would like it to wait for a few seconds so that someone can answer the PSTN line with an analogue phone. This would allow a person to directly pick up the line if they wanted to or if not, let it go to the Asterisk where it

Why not just bite the bullet and move to a supported Linux? - you can be assured that it works - updates are tested - help and support is readily available. - only takes a few minutes to install the whole setup - configuration should port easily. There is almost no Linux administration required once it is set up so getting deep into the actual OS is not required. I have used CentOS (5.x and 6.x)

Hello; I certainly appreciate your response. In fact, I used that exact solution for three of the incoming lines. I setup ring groups and a silent ringtone for each phone. Unfortunately, the last incoming line is more complicated and uses an IVR with multiple input choices, so the solution is not as clear cut as for the other ones. That's why I was trying to look at other options. Best

You might check your phones as well. We had this problem early on with a softphone and it was a setting in the phone that was set to hang up after 30 seconds of inactivity "in case of network disruption". For some reason it was detecting "network disruption" in every call even when the calls were proceeding normally. Unchecking this box solved the problem. It may not be

Your characterization may be true but Skype works much better than SIP when it comes to sound quality. I have SIP softphone with Asterisk server and Skype on the same workstation. Skype just works better over the same network. Ron On 12/03/2015 9:26 AM, A J Stiles wrote: > On Thursday 12 Mar 2015, Thufir wrote: >> I'm testing Asterisk at home, crummy connection. Skype works fine

What are you using for the database - SQLite? I am using mysql (mariadb). I am not familiar with SQLlite. Can you access the database from the console - look up the list of tables - display the contents from a table? Anything to see if your SQLite is working and has asterisk data in it. From your Asterisk console, |CLI> core show help database| should give you a list of commands that you

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

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

On Thu, Feb 12, 2015 at 8:52 AM, D'Arcy J.M. Cain <darcy at vex.net> wrote: > On Thu, 12 Feb 2015 09:43:33 -0500 > Ron Wheeler <rwheeler at artifact-software.com> wrote: > > Why not just bite the bullet and move to a supported Linux? > > If all I had was a phone switch that might be an option but this is > just part of a multi-server system that needs to be

On Thu, 12 Feb 2015 09:43:33 -0500 Ron Wheeler <rwheeler at artifact-software.com> wrote: > Why not just bite the bullet and move to a supported Linux? If all I had was a phone switch that might be an option but this is just part of a multi-server system that needs to be able to move services back and forth so the underlying OS has to be the same for everything. Besides, I am a NetBSD

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

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.

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

Hi, I have a def in my controller see below where i want to find the page object based on a parameter value and then use the page_id to then find all the contents that belong to that page in the same method. Controller def aboutus @page = Page.find_by_name(params[''About Us'']) @contents = Content.find(:all, :conditions => "page_id = #{@page.id}") end