similar to: Logging SIP connection status for review

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

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

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

This past holiday weekend has resulted in some real groaners when it comes to bugs in our dialplan, making obvious the need for some changes in our procedures. First, our hours of operation for Christmas Eve, Christmas, Boxing Day and New Year's Eve had changed with little to no notice. Okay, fine, whatever, I fix. Our Christmas Eve hours (made worse by being Monday this year) dialplan

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

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

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

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

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

All, We are in the process of trying to setup our network to use Verizon's SIP "trunking" product. They say that since Asterisk is not on their certified list of approved devices, we need to go through a field trial to get it approved before allowing us to use their service. Where we are at is getting the design approved. We are trying to watch our budget at the same time. We

Hi, I compile a .cpp with cmd: clang++ -emit-llvm -c -g -O0 -w pbzip2.cpp -o pbzip2.bc -lbz2 llvm-dis pbzip2.bc One function in .cpp is consumer_decompress. However, I look inside pbzip2.ll. The function name is changed to "define i8* @_Z19consumer_decompressPv(i8* %q) #0 {" Why clang adds a "_Z19" prefix and "Pv" suffix? Thanks,

I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

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

Hi, I am dealing with very large datasets and it takes a long time to save a workspace image. The options to save compressed data are: "gzip", "bzip2" or "xz", the default being gzip. I wonder if it's possible to include the pbzip2 (http://compression.ca/pbzip2/) algorithm as an option when saving. "PBZIP2 is a parallel implementation of the bzip2

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

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