[2015-Apr-29 11:23:18] [PHP-WARNING] (/var/lib/asterisk/agi-bin/enc/vmnotify-newvm.php:0) - License check failed!
[2015-Apr-29 11:23:18] [PHP-WARNING] (/var/lib/asterisk/agi-bin/enc/vmnotify-newvm.php:0) - No license for this product (PBXact) - make sure zend_loader.license_path is properly configured in your ini file!
[2015-Apr-29 11:23:18] [PHP-WARNING] (/var/lib/asterisk/agi-bin/enc/vmnotify-newvm.php:0) - License check failed!
[2015-Apr-29 11:23:18] [PHP-WARNING] (/var/lib/asterisk/agi-bin/enc/vmnotify-newvm.php:0) - No license for this product (PBXact) - make sure zend_loader.license_path is properly configured in your ini file!
[2015-Apr-29 11:23:18] [PHP-WARNING] (/var/lib/asterisk/agi-bin/enc/vmnotify-newvm.php:0) - License check failed!
[2015-Apr-29 11:23:19] [PHP-WARNING] (/var/www/html/admin/libraries/php-asmanager.php:343) - fsockopen(): unable to connect to localhost:5038 (Connection refused)
[2015-Apr-29 11:23:19] [CRITICAL] (admin/bootstrap.php:122) - Connection attmempt to AMI failed
[root@localhost asterisk]# nano modules.conf
GNU nano 2.0.9 File: modules.conf
; KDE console is obsolete and was removed from Asterisk 2008-01-10
;
noload => pbx_gtkconsole.so
;load => pbx_gtkconsole.so
noload => pbx_kdeconsole.so
;
; Intercom application is obsoleted by
; chan_oss. Don’t load it.
;
noload => app_intercom.so
;
; DON’T load the chan_modem.so, as they are obsolete in * 1.2
; Trunkisavail is a broken module supplied by Trixbox
noload => app_trunkisavail.so
; Ensure that format_* modules are loaded before res_musiconhold
;load => format_ogg_vorbis.so
load => format_wav.so
load => format_pcm.so
; format_au.so is removed from Asterisk 1.4 and later, remove ; to enable
;load => format_au.so
; This isn’t part of ‘Asterisk’ iteslf, it’s part of asterisk-addons. If this isn’t
; installed, asterisk will fail to start. But it does need to go here for native MOH
; to work using mp3’s.
; Note that on a system with a high number of calls, using a compressed audio format for
; musiconhold takes CPU resources. Converting these files to ulaw/alaw makes the job
; much easier for your CPU.
load => format_mp3.so
load => res_musiconhold.so
;
; Load either OSS or ALSA, not both
; By default, load no console driver
;
noload => chan_alsa.so
noload => chan_oss.so
;
noload => app_directory_odbcstorage.so
noload => app_voicemail_odbcstorage.so
noload => cdr_mysql.so
noload => cdr_radius.so
noload => cel_radius.so
I love FreePBX its AWESOME! and community is the best
For all small - medium size businesses, I recommend FreePBX exclusively.
My only complaint on the technology side is the lack of multi-site dial plan support, to be able to separate groups of phones like example:
“San Diego office” “Chicago office” then based on that phone group, San Diego office calls would only see the san diego dial plan, when calling outbound, calls would go out the San Diego Trunk
This concept is called “Calling Search Spaces” and “Partitions” in Cisco’s CUCM
I’ve accomplished this by adding prefixes to the phone dial plan, so for example San Diego phones prefix “001” to all calls after they are dialed and then freepbx is able to route 001xxxxxxxxx out the San Diego Trunk
Without a better solution for multi-site, freepbx/asterisk cant compete against Cisco’s CUCM for multi-site businesses
I will continue to recommend FreePBX and spread the word! I also try to give back to the community as much as I can by helping others in the forum. I try to keep a ratio of helping others more than I ask for help.
If there is another way I can help, let me know!
Also, is there an app I can get for my phone. I would be more active on the forum if I had an app
Hmmm…I was trying for sarcasm there, but apparently it was lost - I don’t think you can buy a Cisco Pen for $39.00 - let alone anything having to do with VoIP - So much “Conventional Wisdom” around Cisco products and yet we have replaced LARGE (100+ phones) Cisco systems many times and once we do, people are always amazed what Asterisk/FreePBX can do that is included, versus a hefty fee for every little thing on the Cisco.
Even if you bought every paid module including HA on even a small (40+ phones) system, you might approach the cost of a BASIC Call Manager.
An open challenge - Post what you think Call Manager can do that Asterisk/FreePBX can’t either on this thread, or in a new thread and let the community weigh in - I think you will be proven wrong.
Just now saw this (yes 2 years later lol) but yeah FreePBX is great, it isn’t CUCM though
Name a feature?
Multi-tenancy – cant even have overlapping patterns or extensions in different groups in FreePBX (Asterisk). FusionPBX (FreeSWITCH) fixes that though so they are my CUCM open source replacement. FreePBX is great for small businesses.
Another major feature, SRST. But there are literally hundreds (probably thousands) of minor features as well which CUCM has that aren’t matched by FreePBX. They are two different markets though, no need to compete. CUCM cannot be configured properly without years of training and practice, whereas FreePBX can be setup easily.
One more CUCM feature to name drop, Checkout SAF. Its like a routing protocol to dynamically advertise and route to blocks of phone numbers, its pretty awesome.
Extension Routes is equal to Cisco’s “Class of Restriction” feature. Its not the same thing at all, a common misconception, and I think the module name itself is misleading, should be renamed to Extension Restrictions
By the way, 100+ phones isn’t large for the Cisco VAR i work for, that’s a small client, which I agree would be better served by FreePBX. Try connecting 125,000 phones to a FreePBX HA, that is where CUCM shines with clusters of servers all connected, SRST for internet failures reverting to PRI/POTS lines etc
The small business market is wide however, and Cisco has largely stopped competing in it (their smallest product, CUCMBE, is even overkill for under a few hundred phones IMO). They also have Meraki but its no comparison to FreePBX (FreePBX is better)