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How to pull phone reports
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NAT: global, per extension or only in the trunk definition?
In FreePBX there is the NAT setting for each extension, but also global under Asterisk SIP Settings. What are these two attitudes to each other? In the trunk, NAT can be defined also. Can NAT be deactivated in the global settings and enabled in the trunk?
My FreePBX is behind a router and thus behind NAT. The phones are in the same network as the FreePBX and thus they do not need NAT. Therefore I assume NAT in the global setting should be true and on the extensions false.
But I was confused by the following comment from the sip.conf. He seems to say that if the global option NAT is turned on, it also has to be switched on for the extensions NAT and vice versa. Otherwise it could be a security problem.
IT IS IMPORTANT TO NOTE that if the nat setting in the general section differs from
; the nat setting in a peer definition, then the peer username will be discoverable
; by outside parties as Asterisk will respond to different ports for defined and
; undefined peers. For this reason it is recommended to ONLY DEFINE NAT SETTINGS IN THE
; GENERAL SECTION. Specifically, if nat=force_rport in one section and nat=no in the
; other, then valid peers with settings differing from those in the general section will
; be discoverable.
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Huge "nodebug" log files
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NAT: global, per extension or only in the trunk definition?
Assuming that your External Address and Local Networks are properly set, the PBX emulates being on a public IP. The trunking provider is of course also on a public IP, so there is no need for Asterisk to do NAT and it can be turned off globally. If this causes a problem, post details.
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Recording lenny, strange
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Audio problem between two sip trunk interfaces and wan / lan
Dears, I have a problem and I hope you can help me
I have two network interfaces on my server
eth0 -> LAN AND INTERNET
eth1 -> to my sip provider
previously I could make or receive calls without problems from and to the internet with softphone or by VPN with the public ip that goes to the eth0 network, this was until I changed the parameter (external address) because I changed my analog lines to digital and in this parameter I had the public ip that goes to the eth0 network and I changed it to the public ip that has the inferred eth1 due to audio problems, obviously I preferred to use the ip of eth1 since there are cell phone calls coming and going , local phones, other companies etc.
now the phones that are on the internet or vpn register to the server, make and receive calls, but do not have audio, it is probably a routeo / nat problem but I don’t know how to solve it.
Has anyone had any experience with this problem?
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Audio problem between two sip trunk interfaces and wan / lan
If Asterisk must present a public IP address to external extensions and also present a different public IP address to the trunking provider, you will need to have separate transports. One way is to use chan_sip for the trunk and pjsip for extensions (or vice versa); it is also possible to set up multiple transports on pjsip.
Alternatively, if you can use VPN for all external extensions, no NAT is required and Asterisk can be configured with the eth1 public address.
Or, if the trunking provider’s SIP and media servers are on private addresses accessible from eth1, it should be possible to set up Local Networks to include those addresses and only the eth0 public address could be used.
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FreePBX architecture
Hello guys,
I have been working with Asterisk for many years but just started looking into FreePBX for one of my work.
Basically I am interested in knowing FreePBX core architecture, how it interacts with Asterisk and Database.
I did search on forum and found this post where tonyclewis said
FreePBX writes everything to a database tables. Then when you run the Apply Config from the GUI it takes everything in the database, writes out .conf files and reloads asterisk.
Is there any other details available on architecture other than this?
Thanks,
P.S. I’m a new user and hence not allowed to put any link in my post.
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Daylight Saving Problem
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FreePBX architecture
New to the team and project, but this seems to be very much the case. I will say that the core of FreePBX is made up of many modules, each can be found in their own git repository. All of these modules are protected with signatures, so any updates needs to be signed to protect the module from being tampered with. It does take a bit of time to wrap your head around.
There are few other tools you can use defined here including dev tools:
https://wiki.freepbx.org/display/FOP/Setting+up+a+Development+environment+from+the+FreePBX+Distro
Module Signing:
https://wiki.freepbx.org/pages/viewpage.action?pageId=29753662
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FreePBX 14.0.13.6 - No menu bar - possible hacked
I have a client that uses Flowroute for their SIP provider. He got a message asking if he wanted to block calls to the UK. He is in the US and wanted to know if that was spam or something else. I logged into his Flowroute account and noticed that over the last two days there were a lot of 2 minute phone calls. I pulled a CDR report and noticed that the origin IPs were not from his PBX. Ok, this looked interesting to me. I wondered . . . Hmm. Did his PBX get hacked. I went to log into his PBX and saw the missing menu bar. I get the system overview and feed modules, but the blue bar above has NO menus.
Ok, that seemed odd, so I logged into the box. I did a yum update then a yum upgrade then a fwconsole ma updateall. Still no menus. Then I saw another thread and the person suggested doing a fwconsole ma upgrade framework. I got a message stating I had the latest framework.
So I realize that this is really a two part question and they are somewhat related.
- How do I get my menu back?
- How do I determine if the server was hacked into. I have gone to Flowroute and whitelisted US calls only.
Thanks,
dave
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Line appearances similar to analogue?
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FreePBX architecture
Big Module Object (BMO) is also something worth mentioning:
https://wiki.freepbx.org/pages/viewpage.action?pageId=19498386
BMO provides provides methods to install, upgrade, remove, backup, handle AGI, among other things for a module.
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FreePBX 15 is Released!
Are you really releasing FreePBX 15 with all these modules still in EDGE status?:
root@FreePBX:~# fwconsole ma showupgrades
No repos specified, using: [standard,extended] from last GUI settings
Up to date.
root@FreePBX:~# fwconsole ma showupgrades --edge
Edge repository temporarily enabled
No repos specified, using: [standard,extended] from last GUI settings
Upgradable:
+----------------+---------------+----------------+
| Module | Local Version | Online Version |
+----------------+---------------+----------------+
| announcement | 15.0.3.6 | 15.0.3.9 |
| arimanager | 15.0.3.3 | 15.0.3.6 |
| asteriskinfo | 15.0.4.1 | 15.0.5 |
| backup | 15.0.8.61 | 15.0.8.66 |
| blacklist | 15.0.2.6 | 15.0.2.9 |
| calendar | 15.0.4.6 | 15.0.4.16 |
| callback | 15.0.5 | 15.0.7 |
| callforward | 15.0.8 | 15.0.10 |
| callrecording | 15.0.7.8 | 15.0.7.10 |
| callwaiting | 15.0.4.1 | 15.0.4.2 |
| cdr | 15.0.11 | 15.0.15 |
| cel | 15.0.11 | 15.0.15.4 |
| certman | 15.0.10 | 15.0.14 |
| cidlookup | 15.0.7 | 15.0.13 |
| conferences | 15.0.6.5 | 15.0.7.4 |
| contactmanager | 15.0.8.9 | 15.0.8.23 |
| core | 15.0.9.44 | 15.0.9.48 |
| customappsreg | 15.0.11 | 15.0.13 |
| dahdiconfig | 15.0.5.1 | 15.0.5.4 |
| dashboard | 15.0.1.1 | 15.0.3 |
| daynight | 15.0.7 | 15.0.11 |
| dictate | 15.0.4 | 15.0.6 |
| directory | 15.0.12 | 15.0.15 |
| disa | 15.0.4.6 | 15.0.4.7 |
| donotdisturb | 15.0.5 | 15.0.6 |
| fax | 15.0.13 | 15.0.18 |
| filestore | 15.0.3.3 | 15.0.3.4 |
| findmefollow | 15.0.13 | 15.0.16 |
| hotelwakeup | 15.0.5.1 | 15.0.5.4 |
| iaxsettings | 15.0.5 | 15.0.6 |
| ivr | 15.0.14 | 15.0.20 |
| languages | 15.0.6 | 15.0.9 |
| manager | 15.0.5 | 15.0.7 |
| miscapps | 15.0.4 | 15.0.7 |
| miscdests | 15.0.2.5 | 15.0.2.8 |
| music | 15.0.11 | 15.0.16 |
| outroutemsg | 15.0.7 | 15.0.9 |
| paging | 15.0.4.10 | 15.0.4.13 |
| parking | 15.0.9 | 15.0.14 |
| phonebook | 15.0.7 | 15.0.11 |
| pinsets | 15.0.1.6 | 15.0.1.9 |
| pm2 | 15.0.3.6 | 15.0.3.7 |
| presencestate | 15.0.5 | 15.0.7 |
| queueprio | 15.0.5 | 15.0.9 |
| queues | 15.0.9 | 15.0.15 |
| recordings | 15.0.3.4 | 15.0.3.9 |
| ringgroups | 15.0.11 | 15.0.11.4 |
| setcid | 15.0.4 | 15.0.8 |
| sipsettings | 15.0.6.7 | 15.0.6.16 |
| sipstation | 15.0.4.2 | 15.0.5.3 |
| soundlang | 15.0.4.1 | 15.0.5.6 |
| superfecta | 15.0.2.13 | 15.0.2.19 |
| timeconditions | 15.0.12 | 15.0.14 |
| tts | 15.0.8 | 15.0.9 |
| ttsengines | 15.0.4.3 | 15.0.4.6 |
| userman | 15.0.8.2 | 15.0.19 |
| vmblast | 15.0.11 | 15.0.11.3 |
| voicemail | 15.0.17.5 | 15.0.18.8 |
+----------------+---------------+----------------+
Resetting temporarily repository state↧
Audio problem between two sip trunk interfaces and wan / lan
What would be the best way to solve this?
I currently have pjsip for extensions and for the truncal of my SIP provider
Should I leave the ip that goes on the eth1 interface as an external ip?
I had not had this problem before because I only used my eth0 interface for the lan since I had a gateway with fxo lines within the same network and everything worked fine
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FreePBX 15 is Released!
I’m fairly sure there are always modules in the edge repositories as development / testing is ongoing. I could be wrong, but it looks like that is what you’re seeing here.
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FreePBX architecture
Thanks Matt,
Not exactly what I was looking for but those links are useful in other ways.
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UCP - Can not update Device Management
I have a licensed for System Admin and I believe Module Admin if part of it
Under module admin, end point is showing 13.0.118.129 as the latest
How do I upgrade to 13.0.127?
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FreePBX 14.0.13.6 - No menu bar - possible hacked
This will list the IP addresses accessed your server as long as they are in the access_log:
sudo cat /var/log/httpd/access_log | awk ‘{print $1}’ | sort -n | uniq | sort -nr | head -20
These will list all denied and failed access attempts:
sudo cat /var/log/httpd/error_log | grep denied |cut -f 10 -d ’ '| sed ‘s/.{7}$//’ | sort | uniq | sort -nr | more
sudo cat /var/log/secure | grep “Failed password” | grep -E -o “([0-9]{1,3}[.]){3}[0-9]{1,3}” | cut -f 11 -d ’ '| sort | uniq | tr ‘\n’ ’ ’ | sort -nr
sudo cat /var/log/secure | grep “authentication failures” |cut -f 16 -d ’ ’ |cut -f 2 -d = | sort | uniq | tr ‘\n’ ’ ’ | sort -nr
Look into logwatch you will get fail2ban-messages, sites probed the server, su/sshd/sudo-i/sudo sessions, users connected to the server, and many more in one output (as long as they exist in the logs)
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FreePBX 15 is Released!
Module updates are published continuously. When an update is first published it goes to the edge repo, and after time and testing then goes to the stable repo. This is true of all supported versions of FreePBX and has zero to do with whether a version is in general release or not.
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