GOOD NEWS Incredible PBX 11 for Elastix 3.0

arztde

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I like elastix but did use a old version
the reason i did try it was the xmpp server openfire
but openfire have since version 9.1 problems and are not realy fixed.
Another thing i likethat you prepare for any cases and be also open for freeswitch. Gives some hope that you make a Gemeinschaft Version of incredible pbx.
I prefere this for the moment.
until its time i use pbxes.org what for my little private use is also good and free. maybee is the year 2015 realy a good time to look a lot arround and search the right partners as you mentioned in yours article. in incredible pbx its realy difficult and PIAF as well to search and look through the code for me, because mainly are downloads first of the complete packets. Its difficult for the sources if someone isnotto much deep inside. I do not see any branches an dcandivide differences easy. Mabee its there butdid not foundor i am to much stupid.
 

billsimon

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Is this their multi-tenant version?
 

vic555

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Elastix 3.0 had no manual in english...., as of last week; so any distribution in English, featuring a secure Centos 6.4 elastix installation, with a new GUI, is eagerly awaited.
 

bobmats

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I was running elastix 2.4 before. Update to 2.5 was not possible according to their support. Since elastix had some hack problems the last years. I also looked at elastix 3, instaled it on a vps but was not impressed. compared to freepbx11 there are many limitations. Elastix 3 does not rely on freebpx anymore so they started from scratch. It might work in some time but for now I installed incredible pbx 11 on centos which works fine for now.

The advance of incredible pbx will be at least to make elastix 3 more safe which it's currently lacking.
 
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Twilight Sparkle

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O.M.G xd

i so cant wait...........

i used there old versions years ago... but keep telling me *No Dice, when i try to install other apps lol..... hated it so i left and discovered PBX in a Flash & loved it since..........
 

smarks

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The current version of Elastix MT is a really nice starting point for testing multi-tenant. I consider it late Alpha/early Beta software right now.

The more people testing and filing bug reports the quicker it will improve.

http://bugs.elastix.org/main_page.php
 
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wardmundy

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Once you begin with the Nerd Vittles tutorial, you'll quickly discover that beauty is only skin deep unfortunately. Remember what they say about pioneers...



New build (version 11.13-02) will require a fresh start... if you wish to make any outside calls. ;) This new build gets you the very latest CentOS and Elastix MT updates as part of the initial Incredible PBX install. You're well advised to leave yum alone thereafter until further notice.

Trunks are still a work-in-progress at the Elastix end. Haven't gotten incoming calls to work AT ALL yet, but outgoing calls will function with a little jiggery pokery.

Here's an example using VoIP.ms:

1. Create the trunk in PBX -> PBX -> Trunks and fill out the General tab with Name, CallerID, and enable incred.pbx as an authorized Organization. Fill out Peer Settings form with your credentials and do the same with the Registration tab in traditional format (username:[email protected]):



2. Create a New DID entry in PBX -> PBX -> Inbound Numbers. Use 7 digit number, then choose Country Code, and Area Code. Save.

3. Enable the DID for your organization in Manager -> Organization -> Organization by clicking Assign DIDs beside the incred.pbx tenant. Then click Add DID. Drag the DID from the left to BELOW the Add DID heading on the right and Save it.

4. Create a New Inbound Route in PBX -> PBX -> Inbound Routes. Click Add Incoming Route and fill in the blanks. Dont' forget to Set Desintation for incoming calls. Save.

5. Create a New Outbound Route in PBX -> PBX - Outbound Routes. Click Add Outbound Route.

In Settings, give the Route a unique name, e.g. voipms-outgoing.

In Dial Patterns, add a Prefix if you want outbound calls to work at all, e.g.



In Trunk Sequence, drag your trunk into the Trunk Sequence field. Then Apply Changes.



6. So far, so good. Well, almost. The Elastix folks forgot to include the context name or something?? in realtime so you need to add the trunk settings context in /etc/asterisk/sip_custom.conf. Otherwise, Dial SIP/voipms/phonenumber fails. So add something like this to the file using your own credentials (the same ones you used in Step #1). This assumes you named your peer in Step #1: voipms!

Code:
[voipms]
disallow=all
username=123456_subacct
type=friend
trustrpid=yes
sendrpid=yes
secret=yourpassword
qualify=yes
nat=yes
port=5060
insecure=port,invite
host=atlanta.voip.ms
fromuser=123456_subacct
context=from-pstn
canreinvite=nonat
allow=ulaw
6. Restart Asterisk: asterisk-restart

Now you should be able to make outbound calls using VoIP.ms with a dial string like this: 8NXXNXXXXXX.
 

smarks

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Yes, incoming DID is broken. Sometimes it will work. No rhyme or reason when or how. I am finding Kamailio to be a PiTA to troubleshoot. Even with debug enabled the logging is almost useless. No easy way to trace a call through the configuration script.

Really makes you appreciate Asterisk logging.
 

smarks

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Elastix MT uses Asterisk realtime so no need to populate sip.conf and other config files and no need to reload on changes. At least not if it was working properly.

To verify realtime from asterisk CLI:

> core show config mappings

Config Engine: odbc
===> queue_log (db=asteriskcdrdb, table=queue_log)
===> meetme (db=elxpbx, table=meetme)
===> iaxusers (db=elxpbx, table=iax)
===> iaxpeers (db=elxpbx, table=iax)
===> musiconhold (db=elxpbx, table=musiconhold)
===> voicemail (db=elxpbx, table=voicemail)
===> queue_members (db=elxpbx, table=queue_member)
===> queues (db=elxpbx, table=queue)
===> sippeers (db=elxpbx, table=sip)
===> sipusers (db=elxpbx, table=sip)
If you named your trunk voipms then view settings as follows

> realtime load sippeers name voipms
or by username
> realtime load sippeers username 12345_someusername
or by domain name
> realtime load sippeers organization_domain someorg.com
If you have a user admin belonging to organization domain someorg.com

> realtime load sippeers name admin_someorg.com
 
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