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SIP Trunk Issue

June 6, 2020, 2:26 pm
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I wanted to get back to the group to explain what happened in case anyone is searching in the future. Client had a sip trunk coming in on a port with data. Vendor required him to use one of his public IP for the SIP trunk. No authentication, they limit the SIP traffic from his public IP only as their security. Had the normal turnup call. Because of the lockdown, I wasn’t on site, relying on local IT to plug the wire from the ETH1 on the PBX to the 5 port basic switch that his data connections already were plugged in. Its his backup data from this vendor. Interface showed connected, I setup the static IP, etc. Some traffic seemed to move, some didn’t. I could ping the SIP gateway address fine. Calls still weren’t moving in or out. Vendor was sending option pings and not showing a connection. My side showed “unreachable” when I would set “qualify=yes”.

End result was that the switch was bad. I had local IT plug a PC into the switch and assign the public IP to it and he couldn’t go anywhere either. Then had him remove the switch and plug the PBX directly into the AdTran and calls instantly started moving fine. So, the switch showed a connection, seemed to ping OK, but wouldn’t work long enough to get the trunk up and running. Thanks to all that helped, sometimes the simplest thing is the easiest.

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Working Software for Android?

June 6, 2020, 2:34 pm
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No audio on outbound call transfers

June 6, 2020, 2:38 pm
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This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.

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Voice message

June 6, 2020, 3:31 pm
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hi,
I’m creating a table management system and i need the system to call the customer on his mobile and run a voice message " your table is ready, please proceed" .
I need to use my landline number.
please advice.

Thanks in advance.

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The call is cut when transferring

June 6, 2020, 3:38 pm
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Good with everyone.

I am getting this error:
When transferring the call it is cut.

The CDR shows:

lastapp: Hangup
dispositions: BUSY

I am doing the transfer with this command from Nodejs:

exec (`Asterisk -rx" channel redirect $ {channel} $ {context}, $ {extAgent}, 1 "`, (code, stdout, stderr) => {
    console.log ('Redirect:');
    console.log (stdout);
});

The console message is this:

Redirect:
Channel 'SIP/703-00000002' successfully redirected to outgoing,700,1

Your help please.
Thank you.

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SCCP Manager - Cannot Apply Settings

June 6, 2020, 3:50 pm
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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

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Setting up OBI110 for FXO Gateway

June 6, 2020, 4:12 pm
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I’m not sure what you are asking. Do you want to dial e.g. 1 800 437 7950 (a caller ID test number) and send 800 437 7950 to magicJack? If that’s correct, just add a Dial Pattern to your Outbound Route with:
prepend: (leave blank)
prefix: 1
match pattern: NXXNXXXXXX

leaving existing Dial Patterns as they are.

If you still have trouble, make a failing test call and look at the Asterisk log. You should see a line with
Executing [18004377950@from-internal:1]
if that’s not there or the number is wrong, check the phone’s dial plan.
And there should be a line with
Called SIP/OBiTRUNK1/8004377950
if that’s not there or the number is wrong, check your route and trunk dial patterns.

If both of the above are correct, check the digit map for the Line port in the OBi.

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Reload failed because retrieve_conf encountered an error: 255

June 6, 2020, 4:22 pm
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No Audio Changing default TLS SIP port

June 6, 2020, 6:36 pm
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Hi folks,

I have been testing SIP over TLS and it work great for me in the default port 5061, but when I try to change the port , for example 5666, the extension register well but don’t have audio in the call.

I don’t think is RTP forward issue because it work well in the default port.

FreePBX 15
Asterisk 16.9

Any idea?
Thanks,

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Sending SMS Messages via voip.ms

June 6, 2020, 6:39 pm
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This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.

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Address family not supported by protocol

June 6, 2020, 7:02 pm
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This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.

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Parking between two Freepbx machines

June 6, 2020, 7:11 pm
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I just thought of something. Putting from-internal fixed my issue, thank you. But by adding that line will I still be able to get outside calls to both PBX’s?

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FreePBX yum update fails with Error: Multilib version problems found

June 6, 2020, 7:53 pm
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@cynjut Thanks for your reply.

I finally got around to re-installing FreePBX using the five steps I proposed above. Fortunately everything seems to have gone without a hitch, at least as far as I can tell at the moment. The System Overview screen on the Dashboard is showing all green, and my weird and motley collection of ancient Cisco, Aastra, Nortel and Sangoma IP phones all have come up, and I can make outgoing and receive incoming calls.

I decided to “live dangerously” again and installed FOP2 and Webmin. So far, so good.

Thanks for those who helped out.

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UCP log MySQL server has gone away code 2006 error

June 6, 2020, 7:57 pm
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Hello, community.

FreePBX 15.0.16.53
Asterisk 16.9.0

I´m getting endless MySQL server connection errors in /var/log/asterisk/ucp_err.log:

2020-06-03 07:44 -03:00: { Error: MySQL server has gone away code: 2006 }
2020-06-03 07:44 -03:00: There was an error with MySQL Connection
2020-06-03 15:44 -03:00: { Error: MySQL server has gone away code: 2006 }
2020-06-03 15:44 -03:00: There was an error with MySQL Connection
2020-06-03 23:44 -03:00: { Error: MySQL server has gone away code: 2006 }
2020-06-03 23:44 -03:00: There was an error with MySQL Connection
2020-06-04 07:44 -03:00: { Error: MySQL server has gone away code: 2006 }
2020-06-04 07:44 -03:00: There was an error with MySQL Connection
2020-06-04 15:44 -03:00: { Error: MySQL server has gone away code: 2006 }
2020-06-04 15:44 -03:00: There was an error with MySQL Connection
2020-06-04 23:44 -03:00: { Error: MySQL server has gone away code: 2006 }
2020-06-04 23:44 -03:00: There was an error with MySQL Connection
2020-06-05 07:44 -03:00: { Error: MySQL server has gone away code: 2006 }
2020-06-05 07:44 -03:00: There was an error with MySQL Connection
2020-06-05 15:44 -03:00: { Error: MySQL server has gone away code: 2006 }
2020-06-05 15:44 -03:00: There was an error with MySQL Connection
2020-06-05 23:44 -03:00: { Error: MySQL server has gone away code: 2006 }
2020-06-05 23:44 -03:00: There was an error with MySQL Connection
2020-06-06 07:44 -03:00: { Error: MySQL server has gone away code: 2006 }
2020-06-06 07:44 -03:00: There was an error with MySQL Connection
2020-06-06 15:44 -03:00: { Error: MySQL server has gone away code: 2006 }
2020-06-06 15:44 -03:00: There was an error with MySQL Connection
2020-06-06 23:44 -03:00: { Error: MySQL server has gone away code: 2006 }
2020-06-06 23:44 -03:00: There was an error with MySQL Connection

But UCP and the system in general seems to be working just fine.

How can I further debug this?

Thank you.

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UCP log MySQL server has gone away code 2006 error

June 6, 2020, 8:17 pm
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Step 1) from a shell

systemctl status mariadb

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Network setup

June 6, 2020, 11:03 pm
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I’m getting confused as to the typical setup for FreePBX

Please tell us about the application. New system? If so, why on-site, rather than in the cloud? If not, what are you replacing and why? Approximate size (number of extensions, number of simultaneous calls)? Any non-VoIP trunks (POTS, PRI, GSM gateway, etc.)?

Yes a new system, we already have a VM server on site, in order for me to start learning the system (and probably breaking it a few times) i have the ability to try and try again with an on site VM.

I’d say about 30 handsets on desks and 30 softphones with a single receptionist. No more than 5 simultaneous calls. no non voip trunks

Why is a single NIC connected to your existing LAN unsuitable?

because if the firewall is turned on that is a double firewall situation which isn’t recommended. if the firewall is turned off i’m having to set up quoite a few rules on our firewall and im still getting issues with calling but no audio between zulu for example

I want the wan nic to not go via our PFSense router

That implies either multiple public IP addresses or double NAT. Why do you want to do this? What equipment do you have now between the pfSense and your ISP(s)?

there’s only an edge switch between pfsense and isp.
At the moment it’s isp router --> edge switch --> pfsense --> freepbx --> softphones
but in that setup i’d be disabling either freepbx firewall or pfsense firewall.

every time i enable both NICs on FreePBX i get locked out of both WAN and LAN interfaces …

Is the machine virtual or physical? If virtual, which platform? If physical, can you connect a keyboard and monitor to it so you can troubleshoot when the network isn’t working?

virtual VMWARE EXSI and yes i can reboot and disable / enable nics very easily or rebuild from scratch, hence the reasons i prefer it on site.

so it boils down to. in a more traditional office setup with existing firewalls / vlans is it preferred to plug FREEPBX into a network switch port the same as a phone or computer and have clients contact it via an internal ip, or should it be wan facing and everything go out the building and back in again?

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Custom UCP-URL in outgoing emails and translations

June 6, 2020, 11:16 pm
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This topic was automatically closed 31 days after the last reply. New replies are no longer allowed.

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Voice message

June 6, 2020, 11:20 pm
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I am not sure how you want this to happen, I imagine that you’ll need a custom dialplan and integrations with your CRM.

FreePBX has a commercial module Xact Dialer which allows you to run campaigns. If that helps…

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Cisco 7975G not acting as expected with screen timeouts

June 6, 2020, 11:39 pm
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I have a Cisco 7975G setup with SIP firmware and registered with Asterisk. So now I am just tweaking the XML setup for my use using this site: http://usecallmanager.nz/sepmac-cnf-xml.html

So far so good, but I am having an odd thing happen with the screen timeouts you can set. I have them configured like this:

 <daysDisplayNotActive></daysDisplayNotActive>
 <displayOnTime>06:00</displayOnTime>
 <displayOnDuration>17:00</displayOnDuration>
 <displayIdleTimeout>00:10</displayIdleTimeout>
 <displayOnWhenIncomingCall>1</displayOnWhenIncomingCall>
 <backlightOnTime></backlightOnTime>
 <backlightOnDuration></backlightOnDuration>
 <backlightIdleTimeout>00:10</backlightIdleTimeout>
 <backlightOnWhenIncomingCall>1</backlightOnWhenIncomingCall>

The active display settings seem to work fine, however when they kick in any calls to the extension are immediately sent to voicemail. You can, however, pickup the phone and make a call with no problem.

And then there is the backlight settings here, they don’t seem to do anything.

My goal here is to basically have the screen on for most of the day, but have the backlight turn off after being idle for 20 minutes, but otherwise the phone should work as it usually does and not kick calls to voicemail after the screen is turned off.

Is this the expected behaviour? If so, then why is there an option to turn on the screen when a call comes in?

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The call is cut when transferring

June 7, 2020, 12:32 am
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