Hi,

Asterisk: 13.29.2
FreePBX: 13.0.197.31

I have the following scenario and I would like to know if I can target that with Asterisk.

Requirement: I would like to record all calls incoming and outgoing including all the announcements and, hold music, etc.

Hypothesis: Recording all calls coming and going through my Trunk.

So I started to read all the documentation I could find, and I found the following below saying that Call Recording on Applications could do that.

“Call Recordings provide the ability to force a call to be recorded or not recorded based on a call flow and override other recording settings. If a call is to be recorded, it can start immediately. This will incorporate any announcements, hold music, etc. prior to being answered.”

https://sangomakb.atlassian.net/wiki/spaces/PG/pages/22642752/Call+Recording+Module+User+Guide

But after have it configured as requested:

Yet I dindt’ see any recording of that.

So I googled a bit more and I could find some people saying that to make it work I need to edit my Dial Plan for my trunk.

I would like to know if this is the right way?
And if yes, so how I should do it properly?

Would be something around here?

1 post - 1 participant

Read full topic

Recently I installed FreePBX on a GCP VM from GCP Marketplace for evaluation. I wish to configure Asterisk RESTful Interfce (ARI) details and expected to see a section named Asterisk Rest Interface. But, I am seeing only Asterisk Manager soon after "Asterisk mini-Http Server**

I am not able to see that section (Asterisk Rest Interface), even after I made the following changes (Screenshot attached)

1. Read-Only Setting - Yes,
2. Override Read Only Settings - Yes
3. Submit, Apply Config (Red Button)
4. Restarting Asterisk

It looks like the Free PBX v 15.0.23 with Asterisk v16.10.0 Admin UI is different than whatever explained in the official documentation and forums.

Can someone help me to enable Rest Interface from FreePBX UI?

1 post - 1 participant

Read full topic

I have an on-site install of FreePBX that is primarily used for internal calls between about 40 buildings. There was a separate phone system for the few locations that needed to make/receive external calls for the overall organization. The parent organization is transitioning to RingCentral as their VoIP provider. What I’m hoping to do is create a bridge between the two systems.

In Ring Central I can create an extension (say, #123) and get a set of SIP credentials for it. I’d like to use those credentials to register a trunk in FreePBX, such that if you were to call #123 from the Ring Central system, it would connect via the trunk to FreePBX and land in the main FreePBX IVR. Likewise, on the FreePBX system, you could dial a Ring Central extension (say, #456) and use outbound routes to drive that extension over the trunk to RC. I would imagine that calls from FreePBX to RC would always appear to be coming from that one extension (#123) which is fine for my purposes.

Am I barking up the wrong tree with this? Apparently RingCentral has a “cloud connector” for connecting to legacy PBX systems but it runs to thousands of dollars a month.

Any experience using Ring Central as a trunk in FreePBX? Suggestions?

Thanks!

1 post - 1 participant

Read full topic

This is an odd one I’ve been battling for a few weeks. This system is running FPBX 15 so a bit older. I can go onto the UCP page enter my credentials and click login, however it comes up with a very generic error message.

All ports are opened correctly and I am just at a loss for next steps in troubleshooting this. The answer may be to upgrade to FPBX16 but that’ll be a little bit away.

Thank you!

1 post - 1 participant

Read full topic

(topic deleted by author)

1 post - 1 participant

Read full topic

I have successfully configured FreePBX 15.0.37.4 ‘VoIP Server’ in an AWS EC2 instance and we can communicate through a softphone as well as my device to my mobile phone.
At present, the device only allows for speex format and communication is not audible on both sides during a call between the device and the mobile phone.
I have already enable speex, speex16, speex32, alaw, ulaw and gsm in Truck and IAX setting. What are the additional configurations or modules that we need to configure or install? Please help me.

1 post - 1 participant

Read full topic

I have lots of remote extensions, they all work fine, except for one odd ball.
This one device is remote, uses SIP over UDP, and it’s on a poor network.
I cannot fix those things as they are out of my control. I also cannot remotely administer this one device. I have to drive there to make changes. Before you say just get a better phone/network, I can’t in this case.

Here’s the problem. The phone works, but it goes “Unavaliable” every few minutes, then comes back to Available. I would like to just change a timer setting to relax this. I need the warnings to go away because they are triggering alerts (50 times a day).

I will be making a trip tomorrow night to that site.
Can someone suggest which setting/timer I should change in SIP Settings?

Also, the device is a Snom M3 handset, and I use PJSIP on FreePBX 16.
Thanks in advance.

3 posts - 2 participants

Read full topic

Hi,

I’m still trying to interconnect a FreePBX to my main PBX using a SIP trunk. That works so far, but I’m facing an issue.

This is from the SIP INVITE when the main PBX passes a call to FreePBX. The caller ID is the “017212345678” - but FreePBX does not take it and handles the call as anonymous. In the call event list, the caller ID is listed with “COMtrexx2” which is the username for the authentication.

From: "017212345678"<sip:COMtrexx2@10.32.4.5>;tag=6f5641d

Moving the Caller ID to the second part (<sip:017212345678@10.32.4.5>) does not work as the FreePBX then returns an “401 not authorized”.

Any ideas how to tell FreePBX to take the CID from the right place?

2 posts - 2 participants

Read full topic

FreePBX 15
Asterisk 16.30
all modules latest release.
Sangoma D65/D62 phones using REST Parking

Have a client with a busy dr office. They just asked me to put in a queue to handle the calls as this works well in their other offices.

Seems to be fine except about 10% of the calls they try to park get immediately disconnected. I have no logs as of yet because they just reported it and it’s apparently so random I couldn’t get it to happen when testing quickly yesterday afternoon. I’ve asked the users to keep a running record of calls that disconnected and the time it happened. They did say it seems to happen more when the phones are busy but that’s also when you would naturally park more calls so grain of salt there.

Has anyone seen this before or have an idea of what may be causing it? Processor overload?

1 post - 1 participant

Read full topic

Good afternoon,

Any pointers on how to implement the cdr reports to show up on webmin or an external software that captures the logs?

Thank you

2 posts - 2 participants

Read full topic

I’m having very hard time to configure du.ae SIP Trunk on FreePBX. Here is SIP Trunk settings from DU.ae ISP:

ID Range: 971436509xx
Pilot number: 97143650900
IP: 10.15.34.23x
GTW: 10.15.34.23x
MASK: 255.255.255.252

Username: 97143650900p
Pass: xxxxx

Host: fixedimsmey.duentdxb.duvoip.fmc
Domain: du.ae
DNS IP: 10.62.215.44
PORT: 5060

Please community help with this. Your time greatly appreciated.

2 posts - 2 participants

Read full topic

Hi I have a new freepbx installation on a raspi 4.

I have setup a paging group (20) which has two extensions assigned to it.

I am able to make calls between the two extensions, however when I make the intercom call by dialling (20), the initiating extension acts if the call has been answered, but neither of the two extensions in the page group pick up. I have tried the same with the (*80100) 100 being one of the extensions in the ring group which answers on speaker as expected.

Asterisk log for paging call & intercom call:

Page group, extension & yealink intercom configs:
configs

The endpoints I have are Yealink T33-G and the desktop app “PhonerLite”.

In the log for the paging call I have the "res_agi.c: Connecting to '127.0.0.1:4573' failed for url 'agi://127.0.0.1/page.agi': Connection refused ". warning. I have googled this and the suggestions I’ve found were to restart the fwconsole.

I did this with the fwconsole pm2 --restart core-fastagi command but with no luck.

Any help pointing in the right direction or further reading would be apricated thanks.

Jeff

1 post - 1 participant

Read full topic

I have moved property and my Freepbx was not updated. I installed a fresh copy which is 16.0.40.7
Recreated my previous setup. I can dial between extensions and receive calls from my SIP provider.
I am unable to make outgoing calls. I get the response “I am sorry that is not a valid extension” when dialling out. Seems odd to me that the system thinks I am dialling an extension.
My dial plan is et to X. ie the period after X
I thought it might be Firewall, it is behind a Unifi Dream Router. I turned off the rules to Drop but no difference.
Any suggestions please?

1 post - 1 participant

Read full topic

I have a strange problem…

I work with both US and Canadian customers and decided that I want to set up both a US and Canadian DID on my PBX to accommodate for any restrictions or extra charges the customer may have to deal with.

First off, I am using FreePBX 16.0.40.7 with Asterisk 18.20.2 through a static IP. Calls come in and are routed to a ring group which rings a few IP phones in my office.

I have an account with Skyetel and created the two DIDs, the Canadian number’s area code is 514 and the US, 321. I’ve set up an inbound and outbound trunk based on Skyetel’s instructions here: Trunk. I then created an outbound route for each number and based on dialing plans, I just need to dial 999 ahead of any number and the party I’m calling will see a 321 number… otherwise if I only dial the number, the party will see a 514 caller. This works flawlessly… so far so good.

The inbound… not so good. As I am based in Canada, inbound calls to the 514 number have been working well since I set this PBX up several years ago. Now that I added the 321 number, it seems that I can’t configure FreePBX so that anyone in North America can call the 321 number… BUT anyone with a 514 area code can call my 321 number!!! No US (including 321) or Canadian numbers will get through, even other area codes that are assigned to the same city as 514 don’t work. When I call from a 514 number, my call group rings and the call goes through, but when a call is placed to the 321 number from any other area code, there is a voice that answers indicating “technical difficulties with this number”. I think this voice comes from a Skyetel service, but I can’t confirm… yet.

I have tried creating a different IP group at Skyetel for the 321 number, no joy. I even tried sending the second IP group to a different WAN IP but my PBX got confused. I tried creating a different trunk… no joy.

If I watch at the packet level (tcpdump) at my router’s interface and FreePBX when dialing the 321 from a 514 number, I see 5060 traffic communicating between Skyetel/my router and the PBX happily back and forth. When I watch for the same traffic when dialing from any other area code, I see the traffic coming in from Skyetel/my router to FreePBX but the PBX appears to not respond…???

Even stranger or perhaps expected, if I look at the logs in FreePBX, calling from 514 will add some 300+ lines of logs but from another area code not a single entry in the logs and the CDR report will show that no one called which in part leads me to believe that the voice message comes from Skyetel.

This makes no sense to me. I’m no guru with FreePBX but I did get this far on my own… what can be causing this issue? Is there some kind of filtering going on in FreePBX or at Skyetel? I have searched high and low in both areas and come up with nothing. I’ve gone through all the rules in my router but I don’t believe my router could create a rule that can segregate traffic at that level. Quite simply, all I have is a rule for 5060 UDP/TCP pfwd to FreePBX.

Any guidance would be very much appreciated… I need to sleep!

Thanks!

2 posts - 2 participants

Read full topic

Hi -
I’m having an issue were Zoiper softphones on the internet aren’t communicating in to a PBX on my LAN. They aren’t registering with a Request Timeout (408). The same phone works fine on the LAN.

What works:
• Zoiper phone inside the Lan same LAN the freepbx box is on
• Sip Trunk to Tynlex
• Inbound/outbound calls to Zoiper phone on the Lan
• IAX Zoiper on the internet communicating to the PBX on the LAN, Inbound and outbound
What doesn’t work -
• Registering Zoiper SIP phones outside the lan from the internet
• Error is Request Timeout(408)

I’m really confused because the SIP Trunk works, Telnyx has given me sip.telnyx.com so I sort of assume it’s using 5060

General SIP settings NAT settings set to External IP and internal network addresses
Advanced Settings SIP NAT = Yes

OPNsense Firewall rules for port forwarding
• 5060 TCP/UDP
• 5061 TCP/UDP
• 4569 TCP/UDP for IAX
• 9000-20000 UDP

I looked at siproxd on OPNsense, but it looks like it’s for SIP phones going out of OPNSense, I’m trying to go in from WAN via OPNsense to get to the PBX on my lan.

Any suggestions are appreciated
Thanks.

28 posts - 3 participants

Read full topic

Hello everybody,

First, let me clarify that topic: Yes, I’m fully aware that my ITSP should be answering the following question, but they are a bit uncommunicative at this moment. I’m working on that. Until then, I’ll appreciate your help with the following scenario:

I’m currently have an internet connection using that ITSP (which is also my ISP) that is arranged in this way:
ISP gateway → PfSense → Switch → Vlans. I get a public IP from them and nothing more, at this moment.

FreePBX is currently connected to one of the Vlans, having an internal IP of 192.168.40.8. Everything works just as fine (Internally, obviously, as there is no trunk yet, currently testing).

I’ve asked them for a SIP trunk, for which they gave me only this info: “Set your PBX IP to 10.57.2.245/32” - Am I correct to assume that I’m missing some critical info here? I mean, it’s a /32, only a single address, and what about gateway, etc?

I am trying to get a hold of them, but I wanted to make sure I’m not making a fool out of myself - There is some key info missing here that should have been provided by them, other than that single private IP, right?

Anyway, that’s it for today… Thanks a lot. Appreciate your help.

1 post - 1 participant

Read full topic

How to connect dialogflow from google to freepbx?

I found this but i dont know how does it works GitHub - nimbleape/asterisk-dialogflow-rtp-audioserver

1 post - 1 participant

Read full topic

Sorry but again someone down the rabbithole, a question about setting up a Cisco CP-8841.

Goal: Connect a Cisco CP-8841 (with enterprise firmware) to FreePBX.
(and have regular features like, call, call_forward, Voicemail, BLF)

Background:
I have been running a freePBX server for a while with a bunch SPA504G in SIP mode
and thought to replace one or a few phones with the CP-8841.
As many before me I discovered that under the hood there are two versions of the CP-4481.
One with Cisco Enterprise firmware and the 3PPC. I have the Enterprise firmware.

So even though I’am aware I can convert them, and will if this is the only option.
I felt confident enough to try to get them to play nice with FreePBX and in Sccp mode.

So I installed a fresh FreePBX in a VM.

PBX Version:16.0.40.7
PBX Distro:12.7.8-2306-1.sng7
Asterisk Version:18.20.2
Core_sccp 4.3.5
Sccp Manager v14.5.0.4
Firmware phone: sip88xx.11-0-1-11

Enabled TCP transport in FreePBX

Installed Chan-sccp per the github wiki:
https://github.com/chan-sccp/chan-sccp/wiki/FreePBX_Installation

and then the sccp_manager plugin in the GUI.

Provisioning goes throught the FreePBX build-in TFTPServer, and the phone picks-up the
SEPmac file and carries it out. Because when I send it a SEPmac file with SIP registration.
It does work (although the CP-4481 on SCCP firmware is very limited in SIP mode).

What doesn’t work is getting it to register with Chan-SCCP.
The good news is I don’t see any error’s in the Asterisk logs,
but that’s also the bad news. I have no idea where/what to troubleshoot.
The phone just reports “Unprovisioned” after reading the SEPmac, AppDialRules and dialplan.

Could some of you guru’s “dicko”?, give me some leads how to proceed.
Or query me for some logs/screenshots for details

3 posts - 2 participants

Read full topic

How use rest api wakeup call v15 ? endpoint ‘…/rest/hotelwakeup’ is right?
test in postman result error access_denied

1 post - 1 participant

Read full topic

Hi,

I have big delays when I try to playback from CDR. CDR table has around 2.5mil entries.
After I press to play, it can take up to 20 seconds to start playing the Recording. If I choose to download, it starts instantly.
If that makes any difference, it is a VM on on-premise vSphere.

Incidentally, I also have the Call Recording module on this deployment and if I choose to playback through Call Recordings, it starts almost instantly, usually no more than 2 seconds.

I experience the same behavior on another FreePBX isntance as well with 2.1 mil entries on CDR table.

My first assumption is obviously that the perpetrator is the amount of entries on CDR table. But, is there anything else to try before proceeding with deletion of old CDR?
How can I see what delays the what-I-would-only-guess-as buffering when pressing playback from CDR list? Meaning that if for example is a resource problem I can always assign more resources to the VM.

1 post - 1 participant

Read full topic