Hi!
Since your IP is static if it is the only IP which should be communicating with that PBX block everything else...
No need to leave a port open to the world if it's not needed...
Good luck and have a nice day!
Nick
Hi!
Since your IP is static if it is the only IP which should be communicating with that PBX block everything else...
No need to leave a port open to the world if it's not needed...
Good luck and have a nice day!
Nick
I have been searching and trying various different solutions I have found but they are all for older versions of freepbx.
I just installed a new load with the following:
10.13.66-64bit
Release Date: 2016
FreePBX 13 • Linux 6.6 • Asterisk 13
I have a current version of trixbox that is currently working and I am trying to upgrade to latest freepbx.
I have dedicated IP for new freepbx.
I have setup extensions and calling between works fine, and outbound calls work with no issues.
Problem I am having is inbound calls don't route at all. Below is what I am getting.
log_failed_request: Request 'INVITE' from '"XYZ Company " ' failed for '216.82.224.202:5060' (callid: 138451311_100636576@192.168.18.40) - No matching endpoint found
From what I can see the system is trying to route from the callid not the in the signaling.
I have had bandwith.com remove the +1 for incoming and outgoing.
Any ideas, or if you need anything else let me know
Bună ziua Dan!
(Please forgive me if I am wrong about this...)
If it was not a new install I would suggest crashed tables but my guess is an ODBC problem (missing libraries, etc...) or incorrect user/password/database.
As far as config is concerned I would look into /etc/odbc.ini and /etc/asterisk/res_odbc_additional.conf (this one is under the control of FreePBX, you should not modify it directly. If needed this can be changed under advanced settings.
There are also references to the database in /etc/asterisk/cdr_adaptive_odbc.conf...
Your logs should give you some clue about what is going wrong (missing dependency, bad password, etc..) when you place/receive calls...
La revedere!
(Once again please forgive me if I am wrong about this...)
Nick
Hi Lorne!
Sorrrry for the delayed reply...
I checked and I found the "getter"...
I tried it and, surprisingly, it was empty...
I tried hardcoding what it was returning and that did worked...
I will take a look at it again this weekend if I have time to figure out why it ends up being empty... I see where it's populated but something on my PBX seems to keep from being properly initialized...
Thank you and have a nice day!
Nick
What is answering that connection ?
I appreciate the responses and will heed the advice. As I had stated, I've seen other posts with similar issues where they've blamed Netgear routers. I will, however, pursue this with a bit of packet sniffing since I'd like to be sure I'm not missing anything (ie, hacked PC, rouge copier, etc.). God knows we need to protect ourselves from enemies foreign and domestic.
Thanks again, everyone.
Hi, in a new server with a minimal install of FreePBX 13 in CentOS 7 cli displays:
[2017-03-08 14:16:31] ERROR[2448]: netsock2.c:305 ast_sockaddr_resolve: getaddrinfo("provider ip address", "(null)", ...): Name or service not known
[2017-03-08 14:16:31] WARNING[2448]: acl.c:800 resolve_first: Unable to lookup 'provider ip address'
It seems the above error and warning is coming from:
/var/www/html/admin/modules/core/page.trunks.php
Is there perhaps something missing from the minimal install which is causing this?
Modules Installed:
fwconsole ma upgrade framework core voicemail sipsettings infoservices
featurecodeadmin logfiles callrecording cdr dashboard music conferences
Thank you for any help with this.
Did you actually try what Lorne (lgaetz) told you to do?
Yes, you have less choices than if you had Fax Pro but once you have activated fax for a user you can use it as destination of your DID (if it's a fax DID only) or as a fax destination if you enable the detection of faxes...
I don't have Fax Pro and I receive faxes without problems...
Nick
not yet, OK. I will try it. Thanks a lot!
Hi!
Even if you're doing it just for you, the value of your time versus the price of the add-on is reasonably positive.
There is something to gain by doing it at least once the hard way, you learn how things actually work a lot better than if you use something which is already all setted up for you...
I agree that for most people it's preferable to use the commercial solution but doing it the hard way gives a deeper knowledge of how things work...
It's like provisioning phones... For anything more than a few phones it's preferable to go with the commercial solution but learning how to do it manually gives you a deeper knowledge of how things work...
Have a nice day!
Nick
Just wondering is the a ETA on chat? We are doing a lot of testing with zulu and the crm integration for suitecrm at this time and the last part is chat. At this time we use slack but the zulu chat looks promising. Also one last question is there any work being done on allowing other providers to work with the sms portion of zulu?
Hi!
not yet, OK. I will try it. Thanks a lot!
You are welcome!
Have a nice day!
Nick
The 216. Ip is bandwidth sending the call to the freepbx.
From the log it looks like freepbx is using the callid from bandwidth.com to determine the inbound DID not the (to:....) like our existing pix does.
Again, which process in FreePBX do you have answering udp/5060?_
I have a Reseller account and multiple customers under my account. When I go to register a customer FreePBX (on FreePBX) the page never changes. As in, it looks like it's going to work and then I the page refreshes to show sign up for a free trial. In the Sangoma Portal I can see under the customer deployment that a license was generated but for some reason just shows this as the license key bnVsbA==
Am I going about this the wrong way?
It would be pretty cool if there was a module admin update. Like I could choose multiple PBXs and send the command to check and update all modules.
Another cool feature would be groupings. So if I have different SLAs based on customer I could drop them in different "buckets"
And lastly, alert thresholds and online/offline status. So if my node goes down it fires off an email alert. Likewise I could set alerts if a trunk goes down or is showing extremely high utilization by a number a calls I set.
@Altocloud1 Thanks for the feedback!
We're definitely looking into some of those 'management' features aside from monitoring and your feedback will help us greatly to decide which direction to take.
Threshold customization is quite fair and something I agree wholeheartedly that is needed. I can't comment specifically on new features, but we'll keep you informed of new updates and features as they come along.
Thanks again for the feedback!
@Altocloud1 You're hitting a known issue. In fact, we just deployed a few minutes ago changes that will fix this. However, they're currently hidden under a feature flag as they need a bit more testing before going live. You can work-around this by creating a user under the organization of each deployment, or, you can wait a day or two and we'll have it fixed so you don't need to do anything else.
I'll update this thread once this use case works.
Sorry about the inconvenience.
turn off IP calling on the phones. That will fix it.
I haven't tried this but does the PBX determine the ring at the time of the call or is it dynamic?
For instance, if a ring group had three phones and an incoming call comes in and someone is on extension 3 - phones 1 and 2 will ring.. If the person on extension 3 hangs up, will their phone start ringing?
I think that's FN_GM's question.