HI’ve got a bunch. About 1/2 of the machines I have I put public ip right on the box (physical firewalls process traffic rules) and then free pbx distro using their fw and f2b. Works Like a charm.
I put 2 nic on each VM, 1 for wan and 1 for lan.
Other machines share IPs with nat, no problems. Hyperv can handle full failover no problems
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Freepbx in HyperV
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Can not get Cisco Phones to Register
Haven’t gone through your config file in detail or the other responses here but one thing I just picked up was the fact that FreePBX default port for chansip is 5160 and not 5060 - port 5060 is used for pjsip by default. So unless you changed the default settings via the Advanced option in FreePBX your phone’s sip registration will not work.
As I said, haven’t looked at the Wireshark traces or anything else but will go through them a bit later but double check the chansip port setting to start with.
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Can not get Cisco Phones to Register
This is most likely because you are not running ntpd on 192.168.1.15 - Cisco 79xx phones will not update the time or anything if there are errors in the SEPXXX.cnf.xml file. On the phone check the status messages and you will most likely find errors in reading SEPXXX.cnf.xml file - Unfortunately it does not tell you what the error is - so one has to painfully go through each line to work out the error. This could be also the reason why time does not get updated. Will go through the cnf.xml file and get back to you if I see any potential errors.
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Pm2 install problems
Didn’t you already get that by running npm i debug yourself
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Can not get Cisco Phones to Register
We know that the phone attempted to get the time – the NTP request was seen with Wireshark.
It’s indeed possible that ntpd is not running, but we can’t tell because a firewall blocked the attempt to access it. I confirmed that CentOS firewall does (by default) send Destination unreachable (Host administratively prohibited) ICMP in response to a UDP packet directed to a closed port. I also confirmed that if the port is open but ntpd is not running, the response is Destination unreachable (Port unreachable).
I don’t know whether the 7960 will not try to register if it can’t get the time. Given that we see NTP packets but no SIP packets emitted by the phone, if the lack of SIP is caused by something wrong with the XML file, it’s not an error that the phone detected when loading the file.
I thought it worthwhile to solve the firewall issue first (if it’s not terribly hard) because it would likely also affect SIP (once we get that flowing). If it’s too much trouble, changing the XML to use a public NTP server should get past the time issue. Then, if there are still no SIP packets emitted, we’ll need to use the debug tools on the phone.
I agree that there may also be a problem with the chan_sip bind port and/or conflict with pjsip, but we won’t know until some SIP appears in Wireshark and some entries show up in the Asterisk log.
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Why is a girl photoshopped into the picture of the server cabinets on the community page?
Uh, guys - that pic you have on the front page - you forgot to photoshop in her reflection when you photoshopped her in.
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Remote CDR feature
That’s it!
I decided those two steps of Wiki instruction, I specified above, are not independent of each other 
Thanks a lot, solved!
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Column not found: 1054 Unknown column 'time_mode' in 'field list'
Dear support ,
could you advise why i can’t create outbound route and get error
Column not found: 1054 Unknown column ‘time_mode’ in ‘field list’
thank you in advance
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What would you like to see added in FreePBX 15?
I just completed my first FreePBX install. I did not use any of the commercial add-ins (like the telephone provisioning modules) As it was my first and somewhat experimental and for use in my own IT business I used a number of different phone models to see how things would work out. I used Polycom Soundpoint 550’s one SP 335, one Yealink, one Aastrix, and just for fun I hooked in an old Bell Telephone model 500 rotary dial phone on an old Sunrocket ATA that I had to dig around for instructions on how to break into. It’s right in the lobby and it’s a real conversation piece everyone who comes in wants to dial an extension on it. And I brought my existing POTS trunks into a voicecard in a Cisco 1700 and FreePBX is running on an ESXi 5.5 system in a virtual server.
Anyway as I intend to enter the market supplying softPBX-based phone systems what I’m going to say is from my own would-be customers POV not mine.
First - if I was selling someone a commercial FreePBX system I would definitely have them buy the Endpoint Manager. And I would use Cisco phones. Why? Simple. Users associate Cisco with higher end stuff and assume that just because it’s a Cisco phone on their desk, that it’s a Cisco PBX. Managers who are reluctant to go with an “open source” phone system that is inexpensive can be told “if this doesn’t work out then I’ll sell you a Cisco UCS and you won’t have to buy all new phones”
I did not use EM in my system even though I could have cheated by installing it then having it generate the provisioning configs during the trial period then uninstalling it because I felt as though that was cheating and because I wanted to learn the nitty gritty of what was actually happening. And I have a lot of sympathy for whatever developer works on EM because if what I’ve seen with the models I’ve used is the norm, well what a nightmare. All my phones are running the latest firmware from their manufacturers.
I love Sangoma as a company and I’d happily resell your blue box PBX hardware if the customer wanted to use a PRI but not your phones.
Second - do not assume that analog lines are dead. I have some customers who are addicted to faxing. Don’t ask me why and yeah yeah everyone says faxes are dead but they aren’t. They want to put a document into their printer scanner and key in the phone number at the printer scanner and hit send. So I need to have the analog fax line from the printer plugged into something. You need to make sure your endpoint manager has good support for ATAs.
Third you need to publish a guide or something that says what versions of OS need to be defined in ESXi for what version of FreeBBX
Forth your voicemail needs to be able to be broken out when it’s saying the message - the end user VMX Locator just doesn’t work.
last you need to fix whatever you broke in version 14 that prevents it from setting up usable VpIP connection to a Cisco with a FXO card in it.
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Can not get Cisco Phones to Register
@Stewart1, not sure either - I have 7960s and a 7970 but they do not seem to be too fussed about not getting a response from a sntp server. In my case the date/time field remains blank until it is able to contact the ntp server and get an update and the inability to access the ntp server does not stop loading the config and attempting to register. OP’s using 7941 phones and knowing how Cisco changes behaviours of their phone with each model, it is anyone’s guess. I would think the attempt to get to the ntp server would time out after a few tries just like with SIP registrations.
@John45, a quick test would be specify a any ntp server such as nist.gov ( which resolves to 129.6.13.49). This will show whether inability to update time stops the registration or not.
@Stewart1, what I have observed is that these phones do not explicitly say that there is a problem with the config file - the only way I have been able to figure it out is by checking the status messages which do not even tell what the problem is - it always says, “error parsing the xml file” and expects one to go figure it out! I have also noticed that if the config file has errors, invariably the time is not updated (shows an date time a few years back) and registration fails among a few other hints.
Could not see anything obviously wrong with the sepmac.cnf.xml file but these config files are very finicky. I spent days trying to work out why my config file did not work - to find out that I had put a space between two words in the phone display name!
At this stage my suggestion is to point the ntp server to an external address and take it from there.
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Google Voice no outbound route
I have configured a google voice (motif) account under Connectivity. Inbound calls work!
However, I don’t see how to route outbound calls to google voice.
I have checked the boxes “Configure Trunk” and “Configure outbound routes” for the Google Voice channel. But I don’t see a route for Google Voice under Connectivity->Routes. I’ve seen from some older youtube video that I should see a GV_nnnnnnnnnn route in this table but it’s not there in my version. In fact, nor is there a trunk under the Trunks list (though inbound works via my catch all inbound route).
Seems like a bug! Am I missing something? How can I direct specific calls or use Google Voice as a backup route?
I’m using Rapspbx running FreePBX 14.0.2.10 and Google Voice/Chan Motif13.0.3.2. All seems up to date as possible, these seem to be the latest versions.
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Distro stuck at booting when no internet connection
I would run a job after reboot (crontab @reboot) which waits for x minutes and checks, if all services are started. if not, you could have your ip configured automatically by the script and reboot again.
regards,
astrakid
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Queue Feature code * not working
Lorne, did I forget to post something or are you still looking at it… not trying to be a pest, just wanted to make sure you didn’t forget about me… 
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Why is a girl photoshopped into the picture of the server cabinets on the community page?
Report a bug: issues.freepbx.org 
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Column not found: 1054 Unknown column 'time_mode' in 'field list'
I don’t see the full error, but try to tie a time group to your outbound route when creating it.
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What would you like to see added in FreePBX 15?
Faxing is probably still around partly due to HIPPA.
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.call file call to intercom on local phone
Hey all, was hoping to get some help.
I have created a script that will drop a .call file into
/var/spool/asterisk/outgoing
Super simple: this works , calls the extension and then plays the gsm file.
Channel: sip/511
Callerid: 5555555555
application: playback
data: OfficeSpace
What I would like to do is intercom the phone with
Channel: sip/*80511
Callerid: 5555555
application: playback
data: OfficeSpace
But I am missing something, here is the error:
[2018-04-01 09:19:35] VERBOSE[13995] pbx_spool.c: Attempting call on sip/*80511 for application playback(OfficeSpace) (Retry 1)
[2018-04-01 09:19:35] VERBOSE[13995] netsock2.c: Using SIP RTP TOS bits 184
[2018-04-01 09:19:35] VERBOSE[13995] netsock2.c: Using SIP RTP CoS mark 5
[2018-04-01 09:19:35] ERROR[13995] netsock2.c: getaddrinfo("*80511", “(null)”, …): Name or service not known
[2018-04-01 09:19:35] WARNING[13995] chan_sip.c: No such host: *80511
[2018-04-01 09:19:35] NOTICE[13995] pbx_spool.c: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?)
[2018-04-01 09:19:35] NOTICE[13995] pbx_spool.c: Queued call to sip/*80511 expired without completion after 0 attempts
I have verified that the intercom is working on the phone from another station.
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DOException (42S22) SQLSTATE[42S22]: Column not found: 1054 Unknown column 'time_mode' in 'field list'
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Can not get Cisco Phones to Register
I haven’t had time to check all of your suggestions but here is what I have tired so far.
My IPtables now all say target= RETURN prot= all opt= – source= anywhere destination=anywhere
I changed my ntp server to a nist.gov server 129.6.15.28
In the status messages in the phone it gives errors updating Locale, no trust list installed and just
“SEPMAc.cnf.xml(TFTP)” it does not say that there were errors parsing the file.
My FreePBX says my Chansip is running on port 5060 and Pjsip on 5160 in the extensions editor so I believe that it should be on 5050
I will try the other solutions when I have time later today.
Thanks again!
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Google Voice no outbound route
I downgraded the google/motif plugin to version 12 and it works. So there seems to be some bug in version 13.
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