similar to: Why no CentOS 7 repos?

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

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

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

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

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

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 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

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

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

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

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

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

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.

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

B.H. Hello, all :-) We have a cluster of Asterisk (v. 11.9) servers that host IVR applications. The servers work behind SIP proxy (kamailio) for load balancing. All servers are in 2 processor configuration, 8-10 cores per CPU. When a particular server gets about 500 concurrent calls, the sound quality begins to degrade, the sound plays slowly and with clicks. As far as i understand, it's

Hello, I am new on asterisk and do some tests on freepbx. I have 2 SIP provider: Provider1: In-/Out- Flatrate, only 1 Number Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers On Asterisk site i have 3 phones (branch ??, don't know how its called in asterisk) Is it possible to do something like: Phone 1: Incoming Call: Number1/Provider1 Outgoing Call:

>>> On Jan 30, 2017, at 11:55 AM, Motty Cruz motty.cruz at gmail.com wrote: >>> Fresh installed CentOS 7.3 and Asterisk 13.13.1. Download Asterisk from here: http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz >>> I continue to see errors like this: >>> [2017-01-30 08:37:17] WARNING[2332]: chan_sip.c:4061 retrans_pkt:

As I did two years ago, "I'm posting a new thread with the "Top Posting" subject" rather than hijacking the "Paging for Praying" thread. Two questions: 1. Steve K: What do you mean by "/coat"? 2. How do we change rule #5? --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -------------- next part

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

List users, Did anyone else recently receive spam from DIDForSale with the subject "DIDForSale 2012 achievements"? I suspect that they are using this list to harvest email addresses and think they should be called out on this poor business practice if that is the case. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer