Hi!
Can you try
What are your results?
If you get the same thing described in the message that follows, try this:
(which is, by the way, safe, @GameGamer43 confirmed it...)
Good luck and have a nice day!
Nick
and the few messages that follow
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Fresh SNG7 install, DAHDI fails to load
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Auto dialing a number
thanks ill give this a go
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Failed open stream permission denied asterisk.conf
I just installed from wiki.freepbx.org FreePBX 13 CentOS 7 following the direction exactly and it i was able to access the web interface at first however and restarting services I am getting:
On the web page
Whoops \ Exception \ ErrorException (E_WARNING)
file(/etc/asterisk/asterisk.conf): failed to open stream: Permission denied
In freepbx log:
CRITICAL admin/bootstrap.php:270 - Connection attmempt to AMI failed
This is my second install getting the exact same result. I assumed i did something wrong during the first install so I installed centos7 fresh and performed the install again.
I have run fwconsole chown
I have performed every step as root user
fwconsole reload / restart are successful
/etc/amportal.conf, /etc/freepbx.conf /etc/asterisk/*.conf are owned by asterisk:asterisk and are 664 -rw-rw-r--
I am not sure where to go from here and am not having success finding a resolution for this problem. I would appreciate any help you could throw my way. Thank you in advance.
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Failed open stream permission denied asterisk.conf
sudo -u asterisk cat /etc/asterisk/asterisk.confps aux | grep httpd↧
Disaster Relief Vehicles - Phone Solutions
Hello,
You asked a really loaded question here, as the scenarios are almost endless. I am the IT guy at a radio communications company, and (along with my Ham Radio training), there are arguments for radio communications. If all the chips are down, a true emergency, is it critical for the infrastructure to support point to point communications, whereas a radio system might suffice? Your customer will be able to identify if full duplex calls are required, or if a two-way radio system will suffice.
To cover all the emergency bases, you should assume that public resources will get shut down. This includes the cellular network. I work with international folks who have suffered in other areas of the world the loss of GSM networks to control widgets in the power transmission lines and sewer systems, and when a terrorist incident occurred, the government shut down the cellular networks. There went the controls for the power and sewer.
Your group may need to plan completely around the public infrastructure to ensure communications. This likely means your own radio system, including microwave. You will also need to identify who these people want to talk to. All in the same group? Perhaps you can host a server at the tower, and make all connections there, and treat the system as a big office. But if calls need to leave the organization, you will need a trunk to the outside phone network, assuming it is still working.
You may need to look into radio solutions, such as TETRA, which will support telephone calls using a device that looks like a walkie talkie. TETRA is the public safety standard everywhere outside of the US... you may have heard of P25... that is what the US uses, but it doesn't support full duplex telephone calls. TETRA is the wrong medium to use, however, if you are looking to replicate an office environment with lots of full duplex calls.
I apologize if I added to the complexity of your mission, but to be truly sufficient in a "anything possible" disaster, a plan around public infrastructure is required. Remember that in any battle, the enemy gets a vote (natural enemy or human one) and your system will need to respond to it. You may find that an easy simple two way radio is best to get the messages out. All depends on the customer requirement.
Christian
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Llamadas de salida se cortan a los 15 min
Ya aumente los tiempos del RTP en la configuracion general de SIP y ahi mismo agrege la linea de session-timers=refuse. Pero aun con estos cambios sigue igual. Anexo el archivo /etc/asterisk/sip_general_additional.conf que en la consola es el que recomienda modificar en lugar del /etc/asterisk/sip.conf.
--------------------------------------------------------------------------------;
; Do NOT edit this file as it is auto-generated by FreePBX. ;
;--------------------------------------------------------------------------------;
; For information on adding additional paramaters to this file, please visit the ;
; FreePBX.org wiki page, or ask on IRC. This file was created by the new FreePBX ;
; BMO - Big Module Object. Any similarity in naming with BMO from Adventure Time ;
; is totally deliberate. ;
;--------------------------------------------------------------------------------;
accept_outofcall_message=yes
auth_message_requests=no
outofcall_message_context=dpma_message_context
faxdetect=no
vmexten=*97
useragent=FPBX-13.0.192.16(11.23.1)
disallow=all
allow=ulaw
allow=alaw
allow=gsm
allow=g726
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
limitonpeers=yes
session-timers=refuse
context=from-sip-external
rtpend=20000
rtpstart=10000
callevents=yes
tcpenable=no
bindport=5060
jbenable=no
tlsbindaddr=[::]:5061
notifyhold=yes
tlsclientmethod=sslv2
notifyhold=yes
tlsclientmethod=sslv2
allowguest=yes
tlsenable=no
srvlookup=no
defaultexpiry=120
rtpholdtimeout=1800
g726nonstandard=no
videosupport=no
maxcallbitrate=384
canreinvite=no
rtptimeout=300
registerattempts=0
rtpkeepalive=5
checkmwi=10
notifyringing=yes
registertimeout=20
minexpiry=60
maxexpiry=3600
nat=no
ALLOW_SIP_ANON=no
callerid=Unknown
localnet=192.168.0.0/24
language=es
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Auto dialing a number
This sounds like predictive dialling which would normally use something like http://www.vicidial.com/
Appointment reminder can call al list, play a message and the users can press 1 to speak to an agent etc.
You can use call files with a custom context and do AMD which is hit and miss.
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Upgrade to FreePBX 14 from 13 failing
It's not the smoking gun because /etc/init.d/dahdi status redirects to systemctl status dahdi.
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Voicemail Greeting Upload in UCP not working
IIRC, vm folders are not created until you dial into the VM box at least once.
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IVR. Directdial to other PBX that is connected via IAX
Allow the IVR to direct dial from a directory, then populate the directory with the remote extensions.
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UCP data stopped after upgrade to FreepBX 14
I upgraded my FreePBX 13 instance to FreePBX 14 on Friday night. I have a weird issue with TLS registration not working well, but otherwise everything seemed to be working.
Today I finally setup my UCP because I will be working from the back porch for a while and I use the UCP to start calls (I wear a DECT Headset).
Well the UCP works and I setup the widgets I want, but the data is all from prior to the upgrade.
This is a small system and I can easily blow it up and install clean, but I wanted to test the upgrade process.
I also no longer see a way to dial from the UCP. So I will have to use FOP2 for that.
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Feature code *72 and *73 goes busy
Hi,
Having issues when setting up the call forwarding. When I dial *72 and/or *73, it goes to a busy tone. Appreciate the input that you guys can give. Thanks!
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Feature code *72 and *73 goes busy
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Ring Group trouble transferring if new call is answered by another extension in ring group
No one having this issue???
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Failed open stream permission denied asterisk.conf
sudo -u asterisk cat /etc/asterisk/asterisk.conf <---successful cat the file
[directories]
astetcdir=/etc/asterisk
astmoddir=/usr/lib64/asterisk/modules
astvarlibdir=/var/lib/asterisk
astagidir=/var/lib/asterisk/agi-bin
astspooldir=/var/spool/asterisk
astrundir=/var/run/asterisk
astlogdir=/var/log/asterisk
[options]
transmit_silence_during_record=yes
languageprefix=yes
execincludes=yes
[files]
astctlpermissions=775
ps aux | grep httpd
root 477 0.0 0.3 437124 14532 ? Ss 09:35 0:00 /usr/sbin/httpd -DFOREGROUND
asterisk 631 0.0 0.2 439208 9360 ? S 09:35 0:00 /usr/sbin/httpd -DFOREGROUND
asterisk 632 0.0 0.2 439208 8428 ? S 09:35 0:00 /usr/sbin/httpd -DFOREGROUND
asterisk 633 0.0 0.2 439208 9360 ? S 09:35 0:00 /usr/sbin/httpd -DFOREGROUND
asterisk 634 0.0 0.2 439208 9360 ? S 09:35 0:00 /usr/sbin/httpd -DFOREGROUND
asterisk 635 0.0 0.2 439208 9356 ? S 09:35 0:00 /usr/sbin/httpd -DFOREGROUND
asterisk 3274 0.0 0.2 439208 8432 ? S 09:49 0:00 /usr/sbin/httpd -DFOREGROUND
root 5957 0.0 0.0 112648 964 pts/2 S+ 11:00 0:00 grep --color=auto httpd
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UCP data stopped after upgrade to FreepBX 14
The data comes from asteriskcdrdb I would look there first in the cdr table.
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Upgrade to FreePBX 14 from 13 failing
If that's the case why do I get different results between '/etc/init.d/dahdi status' and 'systemctl status dahdi'?
After a reboot, 'dahdi status' echo $? reports 0, whereas 'systemctl status dahdi' echo $? reports 3.
If I then manually start dahdi, both report 0.
Could it be that 'dahdi status' it's not redirecting correctly on my system?
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Sangoma s500 not configuring correctly?
What files am I looking for? Theres a ton in there cause we use sccp b. Looked for the mac address file but didnt see it could it be in a sub folder?
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UCP data stopped after upgrade to FreepBX 14
Hi!
Does CDR and CEL work at all from the Dashboard?
After the upgrade that no longer worked for me, see
(and the next 2 messages that follow it).
I had to recopy odbc.ini (in /etc I believe), as it was no longer present. I had to recopy the backup that was done when the RPM was updated (odbc.ini.rpmsave) to odbc.ini...
Are you having the same problem?
Good luck and have a nice day!
Nick
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Upgrade Conversion Tool Planning?
First off, THANK YOU!!!!! to the author(s) who made this upgrade conversion script! For those few of us adopting an older distro that has never seen an update, but want to get to the latest version without having to take each and every step along the way (at one or two steps per day, if we test our work as we should, for us this would take half a year), this is borderline miraculous.
I tried it. It ate all of our available disk space & almost crashed our production PBX. Fortunately for me, I was already SSH-ed in & was quickly able to du my way to the file & delete it, bringing the PBX back to life. Still a disaster narrowly averted is still a disaster, so I won't be running this script again any time soon!
To the Good, I'm always Very Impressed at how well FreePBX/Asterisk gracefully recover from disasters like this. Nice work, people!!
Question: How do I determine the amount of disk space that is likely to be used for the giant file this script creates? I promise not to complain too much about the fact that it ships that file off to some distant Server somewhere on the Internet for processing, although that does bother me a lot.
I confess I haven't yet looked at the script to see if there's a way to put that file elsewhere, but I'd have to guesstimate that at ~60% utilization on the old PBX, I don't think I have room to do this. I can handle giving the PBX access to more disk space elsewhere in the network, and of course the old .BATman (me) won't have any trouble modifying the script to use it; but I would need to know how much disk space I need to provide for this script to run. (Or is it required that the script run from the distant Server?)
Anybody have any ideas about that?
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