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How to change the licence duration for a commercial product?
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EPM - Export Basefile Edits
That’s a thought! I will try it.
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Telekom Germany SIP Trunk SRV Records
Thanks for the response!
If I understood the link from you correctly, this basically means that Asterisk does not support the SRV setup from Telekom yet, and one should do the resolution to an IP address manually and enter it permanently, right?
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Configure phones to connect to an existing VPN server in FreePBX?
It looks like FreePBX uses OpenVPN for it’s VPN server. I already have a OpenVPN server on my network that I would like to use instead. Is there a way in FreePBX to configure the phones to use my existing VPN server?
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PJSIP Qualify fails where SIP Qualify works
Trying to use PJSIP as Chan-SIP is going away - but I have a customer with 15 remote extensions behind a SonicWALL firewall on Comcast - If I set them with PJSIP extensions, about 6 of them eventually go to unreachable and stay that way - if I set them to SIP Extensions, they work perfectly - they stay registered, reachable and functional.
I have the SIP qualify set to 30 Seconds.
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PJSIP Qualify fails where SIP Qualify works
Do you have SonicWall rules specific to your SIP port? You will have to update those to include the port bind to PJSIP as well
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PJSIP Qualify fails where SIP Qualify works
Of course - both ports are open and forwarded - It’s not a firewall thing, I just don’t think the PJSIP qualifies frequently enough. And I can’t see where to adjust that.
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PJSIP Qualify fails where SIP Qualify works
In Asterisk land it is referred to as the “qualify_frequency”. I don’t know where that is in FreePBX or what it is called. Otherwise you’d need to show a log.
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PJSIP Qualify fails where SIP Qualify works
It’s in the GUI for PJSIP extensions, field is called Qualify Frequency. Otherwise, /etc/asterisk/pjsip.aor.conf stores that setting per extension.
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PJSIP Qualify fails where SIP Qualify works
The default UDP timeout for SonicWall is 30 seconds, way too short. Set this to 300 seconds; see
https://www.sonicwall.com/support/knowledge-base/how-can-i-increase-the-udp-timeout-value/170505468738370/
Also see
https://community.freepbx.org/t/happiness-with-sonicwalls-it-can-happen/39463
Set the registration expiry in the phones to 120, so registrations alone will keep the NAT association alive, even without qualify. Also, if for some reason the NAT association is lost, it will be reestablished in no more than two minutes.
If the SonicWall doesn’t have a public IP address on its WAN interface (the Comcast modem is configured as a gateway), put the modem in bridge mode.
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PJSIP Qualify fails where SIP Qualify works
It most certainly is - Sorry I missed that - I will experiment!
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PJSIP Qualify fails where SIP Qualify works
What do you mean by that? If there are 15 phones on the same public IP address, then the SoncWall will assign 15 different source ports. Forwarding any of these should not be required (and in general is not useful), because incoming calls should appear as ‘replies’ to the REGISTER requests and be automatically routed to the proper phone.
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PJSIP Qualify fails where SIP Qualify works
Didn’t open any ports at the remote site - I meant that the two ports for PJSIP and SIP were open on the Firewall that the PBX is behind (Also a SonicWALL) - Phones registering to it from the remote site force a connection to the PBX and the qualify keeps it open (or is supposed to…)
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PJSIP Qualify fails where SIP Qualify works
That is not robust; a momentary internet outage, Asterisk restart, etc. would cause the connection to be closed.
With the default 30 second UDP timeout on SonicWALL and 30 second qualify interval, slight timing variations will lose the connection. And you should allow for occasional loss anyhow, by using a short registration interval to ensure that any outage is brief.
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PJSIP Qualify fails where SIP Qualify works
Had the same problem with a SonicWall with remote endpoints, after switching to Chan-Sip, still had problems with NAT. Finally threw the SonicWall in the trash, I then recouped some sanity after moving to a Firebox at the remote location.
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PJSIP Qualify fails where SIP Qualify works
Normally with SonicWall I create a LAN > WAN rule with an address group containing all the phones (IP range, MAC addresses, whatever’s clever) as the source and the external phone server as the destination. That way you can set UDP timeout specifically for the phones, BWM, etc.
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Allowing outside callers to dial the queue number directly as if it was an extension
Best suggestion IMO.
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Telekom Germany SIP Trunk SRV Records
Asterisk absolutely supports DNS SRV records. Has for a long time. As noted use PJSIP for your trunk type and things should just work.
You can manually to a SRV lookup to get the IP that are being used and then block them one at a time in a firewall rule to see FreePBX route to the next one.
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DPMA upgrade for 13
Really it comes down to the version of Asterisk you are using, you don’t actually need to install the module via yum, digium/sangoma have updated it for various versions of Asterisk as well. If you have Asterisk 13 (which I believe FreePBX 13 supports), you could manually install the latest DPMA for Asterisk 13 which appears to have been updated (today?).
https://downloads.digium.com/pub/telephony/res_digium_phone/asterisk-13.0/
https://wiki.asterisk.org/wiki/display/DIGIUM/DPMA+Installation
But yeah I agree that you should look at updating. Also - make sure the phones have the latest firmware installed, I had issues with a couple that wouldnt connect and a firmware update fixed them.
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Applying config Error
When I hit apply config I get the following error
Unknown Error. Please Run: fwconsole reload --verbose
I have installed a fresh version of FreePBX and after all the updates including mod and app this is what I get.
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