similar to: Asterisk 1.8.7.0 connectivity to Avaya SM

Thanks for the reply John. About 85-90% of what this box has to do is just handle calls, but it also has options to transfer calls to the main phone system, which up to now has been another asterisk box. For example, you can hit 6 to be transferred to the Lost & Found Department. I do have allowguest set to “yes” already, but of course I also have type=peer and the other stuff for a sip

We have an old Asterisk 1.8.7.0 system desperately need to keep alive for another 6 months or so. We had all kinds of hardware problems, so we virtualized it. Now, random extensions ring once and go straight to voicemail. I went to one of the affected extensions and changed the ring time from the default (20) to 26. Still one ring. I changed it to 30. Now I get two rings. Other extensions ring

On 1/14/19 4:02 PM, Duncan Turnbull wrote: > > > Sent from my iPad > > On 15/01/2019, at 10:34 AM, Thomas Peters <TPeters at mcts.org > <mailto:TPeters at mcts.org>> wrote: > >> Duncan: >> >> You may have it right—I took one phone and set the ring time to 60 >> seconds. I now get about 4 rings on that one. >> >> I wonder how I

Duncan: You may have it right-I took one phone and set the ring time to 60 seconds. I now get about 4 rings on that one. I wonder how I can change the timing source. Thomas M. Peters | Sr. Systems Administrator | tpeters at mcts.org<mailto:tpeters at mcts.org> Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org<mailto:helpdesk at mcts.org> Milwaukee County Transit System

Carlos and Stefan (and other who have helped): I DON'T HAVE the res_timing_timerfd.so file. Can I build it? Recompiling Asterisk is unrealistic in my position but I wonder if I can build the one module. Here's what I do have: apbx:~ $ locate *res_timing_timerfd* /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.makeopts /usr/src/asterisk-1.8.23.1/res/.res_timing_timerfd.moduleinfo

I'd like to start configuring my Avaya 9608G phones for use on Asterisk / FreePBX / PBX-In-a-Flash. I'm using a variety of other phones on my system without major issues. I've read the discussion back in March, May and August of 2016, but unfortunately, my difficulty is much more basic. I think it has to do with DHCP, specifically, what options I'm offering the phone via DHCP.

Ok, commented out that line. It's still doing it. Reloaded dialplan. Please don't tell me I have to restart asterisk. Thomas M. Peters | Systems Administrator | tpeters at mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpdesk at mcts.org >>> Tiago Geada <tiago.geada at gmail.com> 6/26/2015 12:07 PM >>> messages => error states to log error messages to

Switched from Asterisk 1.8 to 13.3.2. Now it logs to /var/log/asterisk/full (good) as well as /var/log/messages (not good). Anyone know why? # grep -v "^;" logger.conf [general] [logfiles] console => notice,warning,error messages => error full => notice,warning,error,debug,verbose,dtmf,fax Thankfully, the .../full logs are rotating properly now (thanks Dale) but we don't

Ah I see, in theory it's possible then. We don't have any IVRs or anything which requires key presses, there isn't even voicemail on this particular phone system so I don't think it will be too much of a problem. I've also updated the firmware on the Cisco phones that have had the issue, just to see if that solves the issue but as it's been going on for a while, I'm

Hi Tom, Thank you for your informative and helpful reply. I had considered using the relaxdtmf setting but held off this due to not using any physical connection hardware -Asterik uses both SIP in and out from an upstream provider (Gradwell.com). Is it still possible to set this when using SIP trunks only and not physical hardware? The box does have a Digium ISDN card but the ISDN is no longer

Ok, to review, I'm trying to get Avaya 9608G to come up in a pure Asterisk environment-- no Avaya SBC or gateway or any other Avaya gear in sight. I have the phone working to the point where it boots up properly, then displays a Username and Password prompt, and says its extension is 123 and the time is 4:57p, which is wrong. But please don't tell me the only way to program up each

The line you commented out was writing errors to /var/log/asterisk/messages The problem you are having is the logging to /var/log/messages via syslog. It appears that Asterisk is sending verbose logging out to syslog even though logger.conf does not have syslog configured. I am not sure why Asterisk is doing that, and I do not want to play on a production system. I added a filter to the

Hello folks, We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyance. I've tried changing the DTMF tones on the phones (512G's running firmware 7.5.5) from In-Band to every other possibility,

messages => error states to log error messages to 'messages' log file On 26 June 2015 at 17:50, Tom Peters <TPeters at mcts.org> wrote: > Switched from Asterisk 1.8 to 13.3.2. Now it logs to > /var/log/asterisk/full (good) as well as /var/log/messages (not good). > Anyone know why? > > # grep -v "^;" logger.conf > > [general] > [logfiles] >

Hi Rowland. The command (samba-tool user enable 'user') is used to enable a user account that has been disabled in AD, but it is not functional to unlock a user account that has been locked by wrong password. Anderson Hoffmann do Carmo MCP | MTA | MCDST | MCTS | MCSA | MS | MOS | ITIL-F | ISFS | CLOUDF | CI-SCS | VCA-DCV | 2016-08-01 13:51 GMT-03:00 Rowland penny <rpenny at

We are running Asterisk version 1.4.23-1, libpri-1.4.9 and zaptel-1.4.12.1 and two Digium TE220Ps. Debugs are set to 10. We have a T1 PRI connected to the telco. Over the last 4-5 days, we have getting Yellow/Red alarms coming from the T1 PRI. The other two ports in use are connected to internal test switches (Avaya Legend/Avaya Definity), and are not showing any errors.

Hi for all! It's a simple question, but I did not find the answer! How unlock domain user after the account blocked by wrong password? How to do this by samba-tool or any other tool in Linux_AD? Or is this possible only by Windows RSAT_Tool? Anderson Hoffmann do Carmo MCP | MTA | MCDST | MCTS | MCSA | MS | MOS | ITIL-F | ISFS | CLOUDF | CI-SCS | VCA-DCV |

Hi Jason/Rowland Great news! the following procedure worked perfectly... I added at the end "Remove manually Windows DC entries in DNS" The script used in step 9 was " https://gallery.technet.microsoft.com/scriptcenter/d31f091f-2642-4ede-9f97-0e1cc4d577f3 " *Very thanks for all!* Anderson Hoffmann do Carmo MCP | MTA | MCDST | MCTS | MCSA | MS | MOS | ITIL-F | ISFS | CLOUDF

As part of a project to move a clients Cisco phones to SIP to work with an Avaya system, I'm taking a Cisco 7961 we bought and adding it to our Asterisk setup. Now, I've gotten the firmware files from the site, the latest version is 8.4 which contains the following files: apps41.8-4-3-16.sbn cnu41.8-4-3-16.sbn cvm41sip.8-4-3-16.sbn dsp41.8-4-3-16.sbn jar41sip.8-4-3-16.sbn

Hello all, first post, need help. I'm running a complex asterisk 1.8 install with five machines. I inherited it and don't fully understand it, nor the deep mysteries of asterisk either. I would appreciate any insight you might have. I scoured the 'net and the Digium wiki and my Google-Fu has failed me. I've been asked to somehow prohibit transfers to extension 3232. It has to be