We have no plans at this time. Seriously 20 users for 200 is cheaper then Bria as a softphone and Zulu is much more then a softphone. To be honest we should be charging 50.00 a user for this. Your going to see pricing changes come in time as Zulu has been adding so many new features and even more coming.
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Zulu licensing
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Zulu licensing
I can totally agree that 50.00$ a license would be worth it - if it were perpetual. My point is that some people just don’t want to buy 20 licenses if they don’t need 20 licenses. I guess it’s really their loss
¯_(ツ)_/¯
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Zulu licensing
Well it would not be perpetual. No way 50.00 would be enough to let you have software that you can use forever with all the features we are adding and do on a daily basis. When I was saying 50.00 a user I was referring to yearly. Now will it go to 50 a user per year doubtful but pricing will be going up in the future.
Zulu is only offered in a yearly renewal option. For the last couple years any new commercial features in FreePBX that we have added are only available in yearly payment option and how things will be moving forward.
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BLF's not working right with Yealink 83.0.X firmware. Response from Yealink
Currently, you have to do a custom firmware. I did this for the 84 firmware.
https://wiki.freepbx.org/display/FPG/EPM-Admin+User+Guide#EPM-AdminUserGuide-InstallingCustomFirmware
The only caution is make sure you run chown to asterisk:asterisk when you upload via SCP to the folder.
It is interesting that they haven’t released any newer firmware than V81 for Endpoint manager. I do know there are tickets open for the newer firmware.
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Zulu licensing
Current MSRP on Zulu for FreePBX is 20 seats for 199 per year, so that breaks down to $9.99 per seat. In your example Zulu for 8 users would average $24.87. Zulu offers that price for desktop licenses, and now includes Android and IOS apps as well. The last time I looked at Bria or Bria stretto licenses we’re per device. So the pricing for Zulu is pretty economical for all the different ways we allow you to connect.
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Audio issues - dropped packets
I don’t have a VPN set up between offices. I have tried in the past to get it set up and never had any luck with it.
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Issue After Power Outage
This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.
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BLF's not working right with Yealink 83.0.X firmware. Response from Yealink
Let me clarify why I asked that question. I do not use the EPM, myself, anymore. I have my own provisioning server and my own templates. I will be testing this using my templates and even directly configuring this in the phone.
I asked because if everyone says “Yeah, we do” and then I cant replicate the issue then my next test would be via the EPM. If the issue can be replicated after that then it does narrow down the field.
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Audio issues - dropped packets
The audio issues could be a variety of thingsm poor bandwdith, poor connectivity, NAT, the router/network (either side) being overloaded…
What are all the common factors at each remote location that has this issue? Routers, switches, phones, etc…
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Zulu licensing
Its great that you are keen to add more and more features but if the prices go up too much you might alienate the users who just want a fully supported softphone/mobile app for FreePBX.
Maybe one price doesn’t fit all for Zulu. Maybe explore having a “Zulu Basic” package and a “Zulu Feature Rich” Package that costs an additional fee.
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Enable disable extensions for external calls
Sounds like exactly what you are asking for.
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Audio issues - dropped packets
I should have plenty of bandwidth and the connection passed a VoIP quality test. Both locations have the ISP and service level.
Both locations also have the same router (Netgear AC3200). Previously, both locations had Asus NT66U routers, I replaced both with the Netgear routers while troubleshooting. The remote location phones run through an unmanaged switch. At the primary location where the PBX server is, everything is hooked directly into the router.
I’m starting to think it’s an issue with my PSTN connection since internal calls between offices are fine. I’ve created a support ticket with my trunk provider (1-Voip) to see if they can help me fix the issue.
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Zulu licensing
Just to be clear right now we don’t have a decision or plan on changing pricing but overtime something will change and feedback from our partners will be taken in account when that time comes.
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HOW TO - Flowroute Trunk with Proper Use of IP Auth and new PoPs
This is how I’ve done it. I’m not sure it is completely proper in all of its intricacies. It is a work in progress. Please correct me if I’m doing something incorrectly. But this setup does seem to work.
Note: This tutorial assumes you have a static IP for your server. If this is not the case, then don’t setup IP Authentication for your Trunk.
First, set your preferred PoP within Flowroute.
Go to Flowroute.com and Log In.
Go to Interconnection -> Registration. Set your preffered PoP.
Then create a PJSIP trunk in FreePBX:
(Obviously, replace the “Outbound CallerID” with your DID Number.)
Then you need to get your “Tech Prefix” from the Flowroute Dashboard:
And put the Tech Prefix followed by an * in the “Outbound Dial Prefix” setting:
Set Authentication and Registration to none. And the “Sip Server” to whatever your preferred PoP is set to:
Change DTMF Mode to RFC 4733, set the From Domain to your Preferred PoP, and add the following to the Match (Permit) Line: 147.75.60.160/28,34.210.91.112/28,147.75.65.192/28,34.226.36.32/28
Save all of this so far.
Add the following IPs to your firewall. If you’re using the built-in FreePBX firewall, the settings will loos like this:
Then go to Asterisk Sip Settings -> PJSIP. And change the “Domain the transport comes from” to your preferred PoP:
Go back to the general tab under Asterisk Sip Settings, make sure “Allow Anonymous Inbound IP Calls” and “Allow Sip Guests” are set to “No”, and take note of your IP:
Within Flowroute, go to Preferences -> Fraud Control -> Outbound SIP Credentials
Click “Disable Credentials”. It should now appear like:
Now go to Interconnection -> IP Authentication and add your server IP without a port (which you noted on the “Asterisk Sip Settings” general tab):
Still within Flowroute, go to Interconnection -> Inbound Routes and add a route to your server. This time, include the port which you are listening on for PJSIP (Default is 5060)
Now, go to DIDs -> Manage. And set the route for your DID to the one you just created:
Let me know if I’ve made any mistakes here. It’s a little hard to make sure I got everything down here.
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Flowroute New POPs & Firewall
Here you go. Let me know if anything needs to be adjusted:
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Audio issues - dropped packets
Well 41ms of Jitter is not great. You can start seeing problems at over 30ms. You will need to run one of these when you are actually experiencing the problem. Does any of those results change? Particularly, the Jitter or Latency.
This is the downside to intermittent issues, you have to do a bunch of things but you can only do them when it is happening.
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Can't forward call outside to cell phone number
Yes but on forward it can’t call the number ?
What do you suppose me for a GSM Gateway device that I can make calls to be forwarded on schedules ?
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Can't forward call outside to cell phone number
I can’t call on 049681222 but if I forward to 038XXXXXX it will call that number because the same it’s configured on IP-PBX 038XXXXYY
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All outbound calls going to cannot-complete-as-dialed
Thank You for pointing that out, I bought a bundled package of modules for park, and fax and extension module was included in that pack! Thanks Hopefully this will help someone else that is thrown off by it!
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Disable Intercom Auto Answer
No no, I didn’t meant to accuse you of anything, I’m acctually glad you wanted to help.
I found out the solution by myself and well… Maybe I shouldn’t have called “*80” all the time but just the extension number. Thank you!
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