You can enter a feature request but to be honest I don’t know how to setup a calendar like that in EWS so we wouldn’t be able to help you. The code is open source however so if you find a solution you are more than willing to add patches and submit them back
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Calendar module sync to EWS
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Manager, task processor
Probably not. You should think about filtering commands your AMI client sees. You can do this from the asterisk side.
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UCP with Firefox and Chrome
Simple solution.
Your server is on IP: 10.10.10.1
Buy a domain: psdk.com
Utilize: *.internal.psdk.com
Go buy a wildcard certificate for *.internal.psdk.com
Point pbx.internal.psdk.com to 10.10.10.1.
Go into the PBX and upload the wildcard certificate to your PBX in certificate Manager. Make the certificate default
Setup this certificate in Sysadmin for HTTPS.
“fwconsole restart”
Go to “https://pbx.internal.psdk.com”
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Can’t get FreePBX to fully work with ht503
hi there i have a problem with the ht503 and free pbx something really strange is happening and i cant get incoming calls whenever the fxs port is registered is like when one of the fxo or fxs is registerd the other doestnt work here are the ttraces any help will be apriciated thanks in advance chris
here are the traces
https://1drv.ms/u/s!AiNbApVnzQCP8QhsIs2MGschlxK-
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Different Buttons for Extensions of Same Endpoint Manager Model
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Can’t get FreePBX to fully work with ht503
Chances are that you need to use a separate port for each interface
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Bulk Handler problem
i try to add some extension via CSV file, but i have one problem.
All of my extensions are inserted to sip_additional.conf but all of the recently inserted extensions squared ( [ ] ) part is empty. So these extensions are not registering to server.
[]
callerid=device <515>
…
When i connect to freepbx and when i submit one of recently added extension without changing anything so [ ] part is filled. So these extensions register to server.
[515]
callerid=515 <515>
…
How can i fix this problem?Is there anything in my CSV file headers?
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Bulk Handler problem
Ok solved the problem. I just exported existing file and edit it.
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"No matching endpoint found" on some extensions, but not all
I have 2160’s and 2170’s on 1.0.9.69 that run on ChanSIP here. You could post the other configuration screens for the account and I could look for anything that seems out of whack.
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Maximum Channels on Trunk - Not Working
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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Can’t get FreePBX to fully work with ht503
In fact, HT503 has separate UDP ports for the FXS and FXO ports. Check how they are configured, but IIRC FXS is 5060 and FXO is 5062. Also make sure that lifeline is set to either auto or diabled and fxo ring through is set to disabled
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Can’t get FreePBX to fully work with ht503
Hi yes I have the fx0 at 5062 and fxs at 5060 everything is happening when I put at the unconditional forward to VoIP instaid of 5062 where it never works for me ( I don’t get incoming calls to fpbx) 5060 then it works I have incoming calls but the fxs port doesn’t work after its unregistered here is the log
2018-05-10 16:27:16] VERBOSE[2324] res_pjsip/pjsip_configuration.c: Contact 100/sip:100@192.168.0.187:5060 is now Reachable. RTT: 1516.848 msec
[2018-05-10 16:27:16] VERBOSE[2324] res_pjsip/pjsip_configuration.c: Endpoint 100 is now Reachable
[2018-05-10 16:27:18] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:27:19] VERBOSE[2324] res_pjsip/pjsip_configuration.c: Contact 000000000000/sip:000000000000@192.168.0.20:5062 is now Reachable. RTT: 13.683 msec
[2018-05-10 16:27:19] VERBOSE[2324] res_pjsip/pjsip_configuration.c: Endpoint 000000000000is now Reachable
[2018-05-10 16:27:39] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:28:01] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:28:22] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:28:43] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:29:05] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:29:26] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:29:47] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:30:03] VERBOSE[2285] asterisk.c: Remote UNIX connection
[2018-05-10 16:30:03] VERBOSE[3252] asterisk.c: Remote UNIX connection disconnected
[2018-05-10 16:30:03] VERBOSE[2285] asterisk.c: Remote UNIX connection
[2018-05-10 16:30:03] VERBOSE[3254] asterisk.c: Remote UNIX connection disconnected
[2018-05-10 16:30:03] VERBOSE[2285] asterisk.c: Remote UNIX connection
[2018-05-10 16:30:03] VERBOSE[3256] asterisk.c: Remote UNIX connection disconnected
[2018-05-10 16:30:09] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:30:31] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:30:52] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:31:14] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:31:35] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:31:56] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:32:18] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:32:39] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:33:00] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:33:22] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:33:43] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:34:04] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:34:26] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:34:47] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:35:08] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:35:30] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:35:51] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:36:12] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:36:34] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:36:55] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:37:16] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:37:38] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:37:59] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:38:20] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:38:42] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:39:03] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:39:24] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:39:46] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:40:07] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:40:28] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:40:50] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:41:11] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
[2018-05-10 16:41:32] WARNING[2319] res_pjsip_registrar.c: AOR ‘200’ not found for endpoint ‘000000000000’
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Install XFCE on FreePBX OS
I understand your train of thought and I knew these comments might come up so that’s why I mentioned it. However this is a home connection and I rather not mess with the router settings since they might get reset in the future and the network has dynamic IP addresses. It will make things a bit more complicated, at least for me.
My colleagues all don’t have an IT background so I can’t tell them to troubleshoot it, and I will be in a different country so I can’t access the physical box anymore. Teamviewer is a lot more user friendly in that aspect. This being said I will this look into your solution and potentially use both.
Besides, any reason why it shouldn’t be possible to install a GUI on the box? It might be not the “optimal” solution for a server but I don’t see any downsides to it.
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Install XFCE on FreePBX OS
You have a Catch22
If you open 443 (selectively) then you can get to the FreePBX GUI, if you don’t , you get neither The FreePBX GUI nor TeamViewer.
Also there is no reason wht you cant add any gui desktop of your preference , I do it myself to get to recalcitrant lan phone http servers when necessary, just dont have it running by default, (wasteful) and add one of Xforwarding , xrdp or perhaps vnc . Whichever you are comfortable privisioning and firewalling
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Install XFCE on FreePBX OS
Thanks! I will keep this in mind, could’ve stumbled over that issue.
Which OS are you using and which desktop environment? What installation instructions worked for you?
When I last used these solutions the performance was much worse than teamviewer.
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CID Superfecta not caching - always showing "CID Superfecta!"
Hi everyone… as per the title subject, if I add the CID Superfecta cache to the scheme, it will always show incoming calls as “CID Superfecta!” even though I have OpenCNAM setup in priority to the superfecta cache.
Anyone see this behaviour?
… and is there any way to flush the superfecta cache?
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UCP with Firefox and Chrome
Thanks i got the certificate. But I got again the same error. I opened the 80,443 and 81 also.
Is there any other port? like 8089 ?
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TrueCNAM - Cannot send Spam Call to destination, only append "SPAM"?
I have a TrueCNAM paid account setup in the CID Superfecta scheme. Although it will prepend “SPAM” to the CID of incoming calls (if it breaches the spam threshold), it is not respecting the option to "Send Spam Call to (and in my case, voicemail) as setup in the scheme default settings.
Send Spam Call to voicemail works for PhoneSpamFilter and Who Called.
Anybody know how to confirm so that it will send spam calls to the configured destination?
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Announcement Won't Play
Hi Forum
I’m using a newly installed FPBX 14 with chan_dongle and latest updates.
Can’t get my Announcement to play.
Here’s some vital config settings.
Settings, Advanced Settings
Default Language (for webUI) = en_US
Admin, Sound Languages
English and English - Australia are both installed
Admin, System Recordings
I have a sound file called Exterminate-call.mp3
She says “The correct number for XYZ is XXXXXX. The number you have dialled is not connected. Please check the number and try your call again.”
I used Admin, System Recordings to upload the mp3 file and make all audio conversions wav, g722, sln, sln16 and sln48.
The same sound file has been used previously with FPBX 12. It works.
After upload, I can play the recording in the browser using the triangle button.
Here’s more vital stuff (ie Mono not stereo)
$ file/var/lib/asterisk/sounds/en_AU/custom/Exterminate-call.wav
Exterminate-call.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
Applications, Announcements
Selected the recording, After Playback Terminate the call.
Connectivity, Inbound Route, (for this DID), Set Destination = Announcements, Select Exterminate Call.
BUT when I call, there’s no announcement played.
After 6 rings, she says “This line is busy, please leave your message after the tone."
And the logs suggest the announcement is being played, but its actually not being played.
[2018-05-13 14:00:08] VERBOSE[26741][C-00000005] pbx.c: Executing [04XXXXXX@from-trunk:10] NoOp(“Dongle/dongle0-0100000005”, “”) in new stack
[2018-05-13 14:00:08] VERBOSE[26741][C-00000005] pbx.c: Executing [04XXXXXX@from-trunk:11] Set(“Dongle/dongle0-0100000005”, “__CALLINGNAMEPRES_SV=allowed_not_screened”) in new stack
[2018-05-13 14:00:08] VERBOSE[26741][C-00000005] pbx.c: Executing [04XXXXXX@from-trunk:12] Set(“Dongle/dongle0-0100000005”, “__CALLINGNUMPRES_SV=allowed_not_screened”) in new stack
[2018-05-13 14:00:08] VERBOSE[26741][C-00000005] pbx.c: Executing [04XXXXXX@from-trunk:13] Set(“Dongle/dongle0-0100000005”, “CALLERID(name-pres)=allowed_not_screened”) in new stack
[2018-05-13 14:00:08] VERBOSE[26741][C-00000005] pbx.c: Executing [04XXXXXX@from-trunk:14] Set(“Dongle/dongle0-0100000005”, “CALLERID(num-pres)=allowed_not_screened”) in new stack
[2018-05-13 14:00:08] VERBOSE[26741][C-00000005] pbx.c: Executing [04XXXXXX@from-trunk:15] NoOp(“Dongle/dongle0-0100000005”, “CallerID Entry Point”) in new stack
[2018-05-13 14:00:08] VERBOSE[26741][C-00000005] pbx.c: Executing [04XXXXXX@from-trunk:16] Goto(“Dongle/dongle0-0100000005”, “app-announcement-1,s,1”) in new stack
[2018-05-13 14:00:08] VERBOSE[26741][C-00000005] pbx_builtins.c: Goto (app-announcement-1,s,1)
[2018-05-13 14:00:08] VERBOSE[26741][C-00000005] pbx.c: Executing [s@app-announcement-1:1] GotoIf(“Dongle/dongle0-0100000005”, “0?begin”) in new stack
[2018-05-13 14:00:08] VERBOSE[26741][C-00000005] pbx.c: Executing [s@app-announcement-1:2] Answer(“Dongle/dongle0-0100000005”, “”) in new stack
[2018-05-13 14:00:08] VERBOSE[26741][C-00000005] pbx.c: Executing [s@app-announcement-1:3] Wait(“Dongle/dongle0-0100000005”, “1”) in new stack
[2018-05-13 14:00:09] VERBOSE[26741][C-00000005] pbx.c: Executing [s@app-announcement-1:4] NoOp(“Dongle/dongle0-0100000005”, “Playing announcement Exterminate Call”) in new stack
[2018-05-13 14:00:09] VERBOSE[26741][C-00000005] pbx.c: Executing [s@app-announcement-1:5] Playback(“Dongle/dongle0-0100000005”, “custom/Exterminate-call,noanswer”) in new stack
[2018-05-13 14:00:09] VERBOSE[26741][C-00000005] file.c: <Dongle/dongle0-0100000005> Playing ‘custom/Exterminate-call.slin’ (language ‘en’)
TIA’s for any tips or clues to get this working as intended.
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UCP with Firefox and Chrome
Thanks. I fixed it and now it’s certified.
Just I get error for web phone call about media encryption that I’m trying to solve.
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