Trying to get FusionPBX working, Newbi questions

J

jeroentjed

New Member
Aug 5, 2025
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Hi there,
First of all: thank you to the admins for approving my membership.
In the past I had a FreePBX setup. As I thought to have a better product with 3CX I made the change. For several reasons I like to leave the 3CX system as soon as possible. It looks like FusionPBX ticks all the boxes for me.

I'm using this video as a guide to install:

Unfortunately, not everything goes as should be. When I try to refresh or reload on the SIP status page, I get a "Connection to event socket failed".
I tried to reboot the VPS, that didn't change anything. Googling it didn't gave me a solution either.

So now I hope to find some answers/help here :-)
Thanks in advance for taking time to read this!

Jeroen
 
What operating system did you use? And what install script or where did you get the install script?
 
Today I did a clean install and that made the problem disappear :cool:
I also managed to connect to my sip provider and to a sip phone. Somehow I can't make or receive a call but that's the next thing to focus on. I think it needs some fiddling with in- and outbound routes.
 
Here are some tips that may help.

Inbound
  • Make sure to add your VoIP provider IP addresses to the providers access control list from Advanced -> Access Controls. Then got press the RELOAD button from the Access Control list.
  • To make your inbound routes use Dialplan -> Destinations.

Outbound
  • Make sure your extension has the outbound caller ID number set correctly
  • Create a Gateway to your VoIP provider
  • Use the Dialplan -> Outbound Routes to route calls to your VoIP provider's Gateway
 
Tried a lot today but somehow it isn't going well. I have a gateway connected to my SIP provider. On both ends it says it's registered. I have two phones, both say it's registered.
I can call them from outside, but I can't make call's between them or outbound.
I followed all the tips above from @markjcrane.
At least i'm getting somewhere now :-)
Any new tips?
 
Have you added anything to the access control list other than what you have been told to? Also, have you added any dialplans that catch more than they should?
 
I checked and I did. I deleted them. Didn't change it. When i Add the outbound route, it automatically adds another one called "call_direction_outbound_00d197" Deleting that one doesn't make a difference.
For outbound I tried with different number formats like +316460XXXXX and 00316460XXXXX. Same result: "Not found". Same for direct call between the extension.
 
This dialplan call_direction_outbound_00d197 sets the call direction to outbound. In other words, don't worry about it. "Deleting that one doesn't make a difference." Its supposed to be there, it won't fix your issue.

The providers access control list is only supposed to be for VoIP providers. Don't add phone IP addresses in that list. After making changes to the providers access control list, make sure to press the RELOAD button or go to Status -> SIP Status and press the RELOADACL button.
 
Thanks, I cleared everything exept the providers IP and did the RELOADACL. Now it all works!
Only outside call needs to be in a specific format (00316460XXXXX) where this should be 06460XXXXX But that's probably fixable in the dialplan.
 
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Well, outside calls work like they should and I created a new domain and all works well.
At this moment I have two questions, I hope you can help with:
- how/where can I set for an extension witch outbound routes it can use?
- I added a stream. When I press play, it works. But when I use it as music on hold on a extension, it remains silent

Any ideas?
 
:confused: Of course I did. And I didn't found or understand enough to answer above questions myself. That's why I ask them here. Feel free to not answer any question if you don't want to. But any reaction as "reed documentation" or "Google it" is a waste of time for both of us and isn't what a forum is ment for.
Thanks for understanding!
 
jeroentjed said:
Well, outside calls work like they should and I created a new domain and all works well.
At this moment I have two questions, I hope you can help with:
- how/where can I set for an extension witch outbound routes it can use?
- I added a stream. When I press play, it works. But when I use it as music on hold on a extension, it remains silent

Any ideas?
Click to expand...
Usually, you set your outbound trunks per domain (for all extensions in the domain) or global (all domains and all extensions) in the dialplan. You can go more granular and do it per extension in the dialplan too but in many cases it's an overkill unless you have a particular use case.

I've had issues with streams before. Look at the FS_CLI to see what error message you get while trying to open the stream. If you can't figure it out yourself, post the error here and we will try to help. Heads up, playing in the browser is very different from playing in Freeswitch. Freeswitch must understand the stream and be able to use it.
 
Thanks pbxgeek,
There is a particular use case: on one domain there are several people who can internally call each other but all have there own gateway/phone number. How can I do this per extension/user?

On another domain is a radiostation. I like to use their stream as music on hold. I looked at the log viewer and this is the result, maybe you can understand it:


e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.543218 97.97% [DEBUG] sofia.c:7493 Channel sofia/internal/200@pbx.nuradio.nl:5060 entering state [ready][200]
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.503218 97.97% [DEBUG] sofia.c:7493 Channel sofia/internal/200@pbx.nuradio.nl:5060 entering state [completed][200]
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] sofia.c:8449 Processing updated SDP
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:8640 Audio params are unchanged for sofia/internal/200@pbx.nuradio.nl:5060.
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5854 sofia/internal/200@pbx.nuradio.nl:5060 Set 2833 dtmf send payload to 101 recv payload to 101
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5796 Set telephone-event payload to 101@8000
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5436 Set telephone-event payload to 101@8000
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5524 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5524 Audio Codec Compare [G722:9:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5586 Audio Codec Compare [G722:9:8000:20:64000:1] ++++ is saved as a match
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5524 Audio Codec Compare [G722:9:8000:20:64000:1]/[G722:9:8000:20:64000:1]
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5524 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMA:8:8000:20:64000:1]
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5524 Audio Codec Compare [G729:18:8000:20:8000:1]/[PCMU:0:8000:20:64000:1]
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5524 Audio Codec Compare [G729:18:8000:20:8000:1]/[G722:9:8000:20:64000:1]
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5586 Audio Codec Compare [PCMA:8:8000:20:64000:1] ++++ is saved as a match
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5524 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5524 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5524 Audio Codec Compare [PCMA:8:8000:20:64000:1]/[G722:9:8000:20:64000:1]
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5524 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMA:8:8000:20:64000:1]
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5586 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5524 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1]
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.483221 97.97% [DEBUG] switch_core_media.c:5524 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[G722:9:8000:20:64000:1]
2025-08-12 08:45:29.363266 97.97% [ERR] switch_core_file.c:346 Invalid file format [https] for [a7.asurahosting.com:7550/radio.mp3]!
7219f54c-ced5-47df-8383-01928c4fe17f EXECUTE [depth=0] sofia/internal/100@192.168.178.11:39517 playback(https://a7.asurahosting.com:7550/radio.mp3)
7219f54c-ced5-47df-8383-01928c4fe17f 2025-08-12 08:45:29.363266 97.97% [DEBUG] switch_ivr.c:632 sofia/internal/100@192.168.178.11:39517 Command Execute [depth=0] playback(https://a7.asurahosting.com:7550/radio.mp3)
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.223220 97.97% [DEBUG] switch_channel.c:1994 (sofia/internal/200@pbx.nuradio.nl:5060) Callstate Change ACTIVE -> HELD
e21a6522-4770-41a9-b3e4-a39c7104b27f
e21a6522-4770-41a9-b3e4-a39c7104b27f a=ptime:20
e21a6522-4770-41a9-b3e4-a39c7104b27f a=sendonly
e21a6522-4770-41a9-b3e4-a39c7104b27f a=fmtp:101 0-15
e21a6522-4770-41a9-b3e4-a39c7104b27f a=rtpmap:101 telephone-event/8000
e21a6522-4770-41a9-b3e4-a39c7104b27f a=rtpmap:9 G722/8000
e21a6522-4770-41a9-b3e4-a39c7104b27f a=fmtp:18 annexb=no
e21a6522-4770-41a9-b3e4-a39c7104b27f a=rtpmap:18 G729/8000
e21a6522-4770-41a9-b3e4-a39c7104b27f a=rtpmap:8 PCMA/8000
e21a6522-4770-41a9-b3e4-a39c7104b27f a=rtpmap:0 PCMU/8000
e21a6522-4770-41a9-b3e4-a39c7104b27f m=audio 11866 RTP/AVP 0 8 18 9 101
e21a6522-4770-41a9-b3e4-a39c7104b27f t=0 0
e21a6522-4770-41a9-b3e4-a39c7104b27f c=IN IP4 192.168.178.90
e21a6522-4770-41a9-b3e4-a39c7104b27f s=SDP data
e21a6522-4770-41a9-b3e4-a39c7104b27f o=- 20022 20026 IN IP4 192.168.178.90
e21a6522-4770-41a9-b3e4-a39c7104b27f v=0
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.223220 97.97% [DEBUG] sofia.c:7503 Remote SDP:
e21a6522-4770-41a9-b3e4-a39c7104b27f 2025-08-12 08:45:29.223220 97.97% [DEBUG] sofia.c:7493 Channel sofia/internal/200@pbx.nuradio.nl:5060 entering state [received][100]
2025-08-12 08:45:28.743226 97.97% [WARNING] switch_core_file.c:463 File has 2 channels, muxing to 1 channel will occur.
 
@jeroentjed as I suspected it can’t play it. Freeswitch gives you an error that it can’t read the stream. It’s probably because it’s not a stream but just a file judging by the extension at the end. For steams you need to use shout:// and not playback.
Here is an article with examples how to play files and streams. Unfortunately it’s missing an example for icecast but it may put you in the right direction.

mod_shout | FreeSWITCH Documentation

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developer.signalwire.com developer.signalwire.com

For your other issue. If I understand correctly you just want to assign a different external caller ID to each extension. You can do it in the extension settings. You don’t need a separate gateway for each caller ID.
Summary: Use one gateway for all extensions. Assign a specific external caller ID to each extension.
 
Thank you @pbxgeek
I think I let it go for now with the stream. It looks very difficult.

I need to use different gateways because these users all have there own sip account. But I think I get how it works now :-)
 
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