FOOD FOR THOUGHT Time for upgrade but what to choose?

[dbaddour]

HI all,

It is time for update. I have been using PBX in flash 1.7.5.6 with Asterisk k 1.6.2 for many years now and no place to get updates on the system anymore.
I have the option now from my company to recourse an updated or... but now when I research so many options and so many are no longer an open source.
I would love to stay with PBX-Asterisk if that is possible for an open source. what I have now is analogue Digium card (with 8) and analogue phone lines.

Any suggestion on what to go for? what version of PBX/Asterisk I can use?

Please and Thank you

 

[dghundt]

How large is your setup?
Are you paid to provide the Pbx setup, or trying to run lean do-it-alone?

 

[smarks]

I really like Issabel. A fork of Elastix v4. Supports Asterisk v13 now.
https://www.issabel.org/

Asterisk 1.6 is ancient. If you don't need the new features that's fine but there are probably critical security updates that were never backported to 1.6

 

[dbaddour]

Hello,
in one location I have 8 analogue lines with 51 extension
another location I have 4 lines with 20 extension
and another location 1 line with 4 extension.
they are all connected Via a VPN that we make internal extension calls.

May in the future if with this 3 location find the right solution we can expend it to couple more locations as well.
I like to keep it in house, self admin

Thank you

 

[jerrm]

If your doing it yourself and it's not the primary job focus, I'd probably go with IPBX 13-13 unless there is a feature you can't live without in Wazo/Issabel/3CX/etc. IPBX is probably has the smallest learning curve.

Probably worth examining if you still need the analog lines. Lots of potential savings with VOIP (at U.S. prices anyway) that would quickly pay for any new equipment or cloud services. That probably depends on the reliability of the lines vs the internet connection.

Sounds like there are three PBXs currently? If internet connections are solid, one (or two for failover) in the cloud might be simpler.

 

[dbaddour]

Hello,
Actually I was looking into the IPBX 13-13, not too fussy about the extra features that others offers. 8 or 9 years ago when installed and worked with PBX in flash, was fun and still fun and challenging. I am the only one that works with the phone systems within my company. I tried install the IPBX13-13 twice, both times when login to the GUI I see an error on the top of the page with (can not connect to Asterisk). I believe that I will go this way for now, with IPBX13-13 and hope the 8 ports Digiium card will be happy with it.

I looked into the VOIP ( I am in Canada, east coast) for the cost my bosses wanted to stay for what we have for now. Phone lines are ok. the Internet we have FiberOps. so not bad.
Yes I have 3 PBX now, but all in different locations, one in east coast, the other on the west coast and the last is in Australia. but I do have a secondary system in the east coast as failover

again thank you so much for your reply, good to know.

Thank you

 

[dghundt]

I'm in a similar situation but only one facility.
I'd give IPBX 13-13 another go. I just installed it into my spare physical box and works great. Just follow Ward's blog step by step.
Way back when, it seemed voip providers came and went (except vitelity to name one) and call quality was just ok at times. These things appeared to have changed for the better. I'm setting up my new box so we will only have one pstn line for our main number. Everything else will roll over to voip DIDs.
I'm using T38.com for FOIP with a grandstream FXS device and our old physical fax machine and love it. The Obi had some issues. Wish we could dump faxing, but it's a HIPPA thing.
I have some old Rhino analog cards that work great but, support has fallen by the wayside and I just don't have the time or gumption to relearn some Linux to get them rolling in the new boxes. I'll still try, but I plan to just use a grandstream fxo gateway instead. It's cheap and I only have to configure sip/voip. It's also portable - I will configure my hot spare Pbx to look for the same gateway trunks/extensions. That means for all future Pbx's I build, I won't have to fuss with installing a physical analog card in each one.
Maybe Digium provides better support if you need it.

Good luck!
Dump some of those analog lines though.

 

[wardmundy]

New Nerd Vittles article will be out on Monday addressing all the current (free) options.

 

[redstonemason]

I have a number of clients where I am running PIAF GREEN under Centos and 1 Ubuntu. I need to upgrade the Ubuntu based system because it is sort of the orphan. So I was happy to see IPBX13-13 yesterday. (Not fond of 3CX).

I got it up and running on VULTR (my production clients are on AWS).

IPBX13-13 certainly has my vote as the go forward option for me.

Just not sure if I can run incredible backup under PIAF GREEN on Ubuntu and restore to IPX13-13 under CentOs.

 

[wardmundy]

IPBX13-13 certainly has my vote as the go forward option for me.

Just not sure if I can run incredible backup under PIAF GREEN on Ubuntu and restore to IPX13-13 under CentOs.

Cut-and-paste is your best bet. Ubuntu approach to MySQL is radically different than the CentOS/SL platform.

 

[redstonemason]

Thanks Ward. I had a feeling that a fresh start would be the only way to go. I am experiencing a few problems with a basic test of the GUI and DB by attempting to modify extension 701 but I'll post those issues in the appropriate spot. Not sure if you have a forum of IPBX13-3 yet.

 

[dbaddour]

Thank you for your reply,
I will be looking into the VOIP providers in near future for sure. ( In Canada). my main concern at first is the company that i work with uses the fiber network connection for crazy amount of data that goes thru in and out.. unless i have a dedicated network line only for phones that will make the traffic much easier on the network.

I sure will post again if we go that way,.

Cheers

 

[dbaddour]

Hi again,
With the older version that i am using, most of the configuration especially the trunks are ZAP Trunk (Dahdi compatibility Mode). since i am still using the Digium card for analogue lines.
now with the new version, it has many options and flavors. I will be using the same kind of Digium card for the analogue lines, but what to chose from the trunk options: add SIP (chan_pjsip)? SIP (chan_sip)? DAHDI Trunk? Dundi Trunk? what is the best of these option would work with the situation that i am in.

Thank you

 

[dghundt]

The analog phone cards use the Dahdi trunk.
Your voips extensions and trunks would use SIP.

I've been experimenting with pjsip trunks/extensions, for a long winded reason, and they take a little more time to configure. My snom phone won't work with pjsip, or at least I haven't figured it out without Fail2Ban locking me out eventually.

 

[dbaddour]

Thank you,
with Incredible 13-13 Dahdi module is not installed by default. I should install it from the Mod admin and config?

 

[dghundt]

The dahdi module is not required, but can make it easier to manage.
I never could get my rhino card working . For some reason chan dahdi.so does does not exist/load.

 

[dbaddour]

Ok well here what i have done...
without the Dahdi module, I connected my analogue lines to the DIgium card and hopped for the best. unfortunate that did not work, no trunks were registered on the dashboard and when i tried to make a call: just a part of the log:
2018-03-13 07:14:37] WARNING[5852][C-00000002]: channel.c:6262 ast_request: No channel type registered for 'DAHDI'
[2018-03-13 07:14:37] WARNING[5852][C-00000002]: app_dial.c:2525 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [s@macro-dialout-trunk:24] NoOp("SIP/1026-00000002", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 66") in new stack
-- Executing [s@macro-dialout-trunk:25] GotoIf("SIP/1026-00000002", "0?continue,1:s-CHANUNAVAIL,1") in new stack
-- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1)
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:1] Set("SIP/1026-00000002", "RC=66") in new stack
-- Executing [s-CHANUNAVAIL@macro-dialout-trunk:2] Goto("SIP/1026-00000002", "66,1") in new stack
-- Goto (macro-dialout-trunk,66,1)
-- Executing [66@macro-dialout-trunk:1] Goto("SIP/1026-00000002", "continue,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/1026-00000002", "TRUNK Dial failed due to CHANUNAVAIL HANGUPCAUSE: 66 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] ExecIf("SIP/1026-00000002", "1?Set(CALLERID(number)=1026)") in new stack
-- Executing [902xxxxxxx@from-internal:11] Macro("SIP/1026-00000002", "dialout-trunk,1,902xxxxxxx,,off") in new stack

so decided to install the Dahdi Module hoping for this to work. after the install (via the Module Admin). unfortunate all kinds of error start to appear,
-
section

File /etc/modprobe.d/dahdi.conf does not exist.
File /etc/dahdi/system.conf does not exist.
File /etc/dahdi/modules does not exist.

when attempt to configure the DAHDI. more error appeared and could not reload or restart the Dahdi module
.
so back to zero now and not sure what to do?

THX

 

[dghundt]

Same errors I got.
You can see my post titled Rhino card, at the end of the thread are my sample scripts. I installed quite a bit.
However chan dahdi.so does not load/exist . I couldn't figure it out there. I'd need paid support at that point.
The freepbx dahdi module crashes and prevented any freepbx reloads. I had to delete the dahdi module via winscp and fiddle with it.

 

[dbaddour]

Excellent, thank you.. Just following your solution post. but you just lost me on "use winscp to add rcbfx to the /etc/dahdi/modules file" i cannot find rcb

 

[dghundt]

Sorry,
The rcbfx is the name for the rhino analog card driver. This will not apply to you.
The dahdi install scripts, I recall, recognize the Digium cards already. No additional driver installation needed.
So I believe your steps would only be,

I installed dahdi 2.6.1
cd /usr/src/
wget https://downloads.asterisk.org/pub/...plete/dahdi-linux-complete-2.6.1+2.6.1.tar.gz
tar -zxvf dahdi-linux-complete-2.6.1+2.6.1.tar.gz
cd dahdi-linux-complete-2.6.1+2.6.1
make
make install
make config

What's interesting with 2.6.1, is I see during the dahdi installation it sees my rhino card!

Then installed dahdi 2.11.1 again
cd /usr/src/
cd dahdi-linux-complete-2.11.1+2.11.1
make clean
make
make install
make config

Now there is no mention during dahdi installation about my rhino card.... Nuts.

Then fix pointer reference with two commands
echo “/usr/local/lib” > /etc/ld.so.conf.d/usr_local.conf
/sbin/ldconfig

cd

Then the following you may or may not need to do
service dahdi restart
dahdi_genconf
dahdi_cfg
service dahdi restart
***freepbx error etc/dahdi/modules not owned by asterisk
****etc/dahdi/system.conf not owned by asterisk
****/etc/modprobe.d/dahdi/conf not owned by asterisk
fwconsole chown
then change permissions for modprobe.d/dahdi.conf to asterisk in winscp
service dahdi restart

However, dahdi still does not load for me because chan dahdi.so is missing.
Freepbx dahdi config module still does not work, I assume as a result of chan dahdi.so missing.
Not sure how to fix that if it happens to you.
Please post your results, esp successes!