This works for us. We create the main DIDww trunk in the GUI and then setup inbound trunks
for each additional IP in sip_custom.conf like below all pointing to the same context "from-trunk-sip-DIDww".
This will also open up the ports in the integrated firewall.
;DIDww
[46.19.209.11]
host = 46.19.209.11
type=peer
disallow=all
allow=alaw
context=from-trunk-sip-DIDww
If the phones were on the WAN side, I would agree.. but having UDP INVITE packets floating around inside the LAN makes me a bit nervous.
This helped me get iSymphony to work on a fresh install of FreepPBX 13, too. Thanks for sharing!
I have also had this happen in 2 occasions. Both times I was able to go I to advanced settings of the extension and set it up there. Voila it reappeared.
I'd say your first effort should be to lock down your PBX. The brute force attacks, however lame, will fill your logs in no time.
You might also try the extension number for the Authentication Username.
the phones are behind a NAT but the SIP port is exposed to the open Internet which means you can receive IP to IP calls i.e. phantom calls. It's a known issue and changing the SIP port 5060 to something else would fix this but the better solution imho is to turn off IP calling on the phones.
Not sure what 'exposed' means.. There are no forwards that allow incoming connections to the LAN.. the phones initiates the registration to the external PBX.
Sorry dude, but i couldn't resist. . .
A rouge copier no less 
FreePBX: 13.0.190.19
Asterisk: 13.11.2
Server is running as a VM on ESXi 5.5 host.
We seem to be experiencing random, brief (2-5s) drops in audio, on both internal and external calls. I can't reliably reproduce the problem, wondering what direction to head in terms of troubleshooting this issue.
Will log files on the server show anything useful, or am I better off trying to get a packet capture of a call where the issue occurs? Or is there another avenue of troubleshooting I should look into?
The bug is you have customer orgs under you in the portal as orgs but no users for that org in the portal and RMS expected the org the deployment belongs to would have a user always.
All feature request should be just like all other FreePBX products. Opened at issues.freepbx.org under RMS project
Chat is due out server side later this week in edge.
As far as SMS no it uses FreePBX SMS and only provider for that is SIPStation.
We made a video outlining the process of open Feature Request, useful if you have never opened one.
Greetings,
I am creating a new system for a client. I made a time condition for their business hours and also assigned a BLF to monitor and manually change. It dials / subscribes to *271.
Setting the time condition to end a couple of minutes ahead of the current time I can call and test that the condition toggle does flip and I am sent to the proper IVR to announce that the business is closed. However, the BLF hint does not update at the indicated time. It only updates after a call has been made.
I checked the CLI using: core show hints
Before call is received, but after time condition is flipped:*271@timeconditions-: Custom:TC1 State:Idle Presence:not_set Watchers 1
After call is received:*271@timeconditions-: Custom:TC1 State:InUse Presence:not_set Watchers 1
Was originally on current stable version 13.0.33.3 of the Time Conditions module. Tried updating to EDGE version 13.0.33.3, and that didn't seem to help. Tried downgrading to 13.0.32.4 & .3 no help either.
FreePBX 13.0.190.20
Any ideas / help is greatly appreciated.
Am trying to register my Ubuntu server in the portal as sipstation support says I need paid support to fix my SMS issue, they feel my Freepbx is mis-configured.
dear community,
first of all: my Trunk is up and running.
have 2 extensions 10 and 20.
i can made outbound calls, and inboud calls, without problems.
on both extensions i have a snom phone.
so now i can: call out from 10 and 20 to PSTN.
call from PSTN to 10 and 20.
and now here comes my problem:
if i redirect one of my both phones with the forwarding key,
if i call from external, all is redirecting fine (i get the correct extension).
in this case i have forwarded my phone 20 to external number.
if i now call from 10 to 20, the call breaks with 480.
so now i call here from ext 20, and not from +PUBLICNUMBER20.
i have done sip debugging, the diversion header = .
What is my problem with them?
please help, thank you very much
Hi Andrew
I've followed up with Leo; but still no reply to my communication around the en_NZ files I have provided to Sangoma.
Digium closed https://issues.asterisk.org/jira/browse/ASTERISK-26626 today - having accepted our en_NZ files.
Keen to ensure that FreePBX/Sangoma are progressing as well.
Cheers
David
Using EPM for Sangoma S500, I modify the Horizontal Soft Keys and add an action "REST-call forward"
The issue is the button does not show up the phone
I created a group and gave it rights to phone apps and then make the user is a member of this group
When I checked http://server:84/applications.php/callforward/main?user=EXT106 The result was
{"application_name":null,"application_display":null,"page_name":"main","type":"display","exitPath":null,"layout":[],"action":[],"error":[{"reason":400,"display":"Phone Apps module not licensed."}]}.
I don't have the license for the Phone Apps but my understanding, it's included on Sangoma Phones
The S500 has the latest firmware. S500 2.0.4.26. I'm using FreePBX 13.0.190.19
The log file show the following warning and notice
2017-03-09 01:52:20] WARNING[5405] chan_sip.c: Purely numeric hostname (106), and not a peer--rejecting!
[2017-03-09 02:00:14] WARNING[5405] chan_sip.c: Purely numeric hostname (106), and not a peer--rejecting!
[2017-03-09 02:00:14] NOTICE[1343] res_pjsip_exten_state.c: Extension state subscription failed: Extension 9921*106 does not exist in context 'from-internal' or has no associated hint
[2017-03-09 02:00:14] NOTICE[1343] res_pjsip_exten_state.c: Extension state subscription failed: Extension 9924*106 does not exist in context 'from-internal' or has no associated hint
I'm using Custom Contact. I was wondering if custom contact is not compatible with phone apps because of the error : does not exist in context 'from-internal'
Appreciate any info
We don't provide support on non FreePBX Distro systems.
The BLF only gets updated when a call comes in or based on a interval. That default.interval is 10 minutes. You can change this in advanced settings module.
