/var/spool/asterisk/fax
Yes.
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FaxPro module operation
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Software to send fax
Thanks @gbaughma for shedding some light on this.
Apparently there will have to be added component(s) on my existing FreePBX server before it can handle sending fax regardless.
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Commercial Voicemail Notification doesn't work
Nevermind, I got it sorted out myself. Thank you so much though
it’s under settings/voicemail admin/ settings/ Email Config
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Fwconsole reload and limit of freepbx/asterisk
Yes, it is.
You can’t, because the system needs to know about all of the parts of the system that are loading/reloading.
There used to be a “realtime” extension for Asterisk, but it wouldn’t have helped you with this.
The limit to the number of extensions is variable based on several factors, but we’re have several people try to stress the system by using several hundred phones registered to a single instance. Your concern about reload time is well-founded at this scale - a reload can take several minutes to pull in all of the data and set up the system. Having said that - I don’t think anyone has tried to set up a system this big in several months and the system has had several improvements over the past year that could change this.
The number that sticks in my head is around 800 extensions before things start to really go haywire, but that’s with a pretty typical 10:1 ratio (extensions to active calls) so your use case might change that.
With this size of an installation, your network is likely to cause more problems than the server will. Call quality issues, voicemail performance, and just registration/hints processing all take their toll on the operation of the system. Throwing additional cores at the problem isn’t likely to help a lot (it will, but not linearly).
My expectation is that the system can handle a lot, but you have to be willing/able to split the installatoin up over several servers to get the kind of performance you are sounding like you want.
Finally, remember that FreePBX Is not a good multi-tenant solution, so if this is going to be for a bunch of independent “agents” (several businesses), you might need to look at a different solution.
Try it and see what happens. Let us know when you’re done testing.
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SIP Call without Authorization Header
What goes wrong? Did you remember to have the device send to port 5160 (or whatever you have bind port set to)?
Other options:
Set up your device as a pjsip trunk w/no registration or authentication. You’ll be able to make and receive calls, but you won’t have voicemail, follow-me, forwarding, DND, etc. What kind of device is this? A trunk might be more suitable.
Set it up as a custom extension and put the config in the pjsip custom config files.
Possibly, you can modify the existing config as shown in
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Asterisk Info, Peer report of SIP Peers indicating Host as PBX server IP
The Reg. Contact comes from the Contact Header which contains the contact information about the user. In other words, where this Contact exists and requests/replies should be sent to. Since all these endpoints are sitting on networks behind NAT, the Contact Header is going to contain the Contact User (generally the SIP user) and the local host of the device (its IP).
When the PBX is on the same network or Layer2 then having the RFC1918 (private) IPs is not a big deal because everything is using RFC1918 IPs. However, when the devices are “remote” and require the device to communicate over the Internet then those IPs can’t be used. Many devices have NAT settings where you can tell it what the external IP/Ports are to be used. Some devices are “NAT Aware” and realize they are behind NAT and will attempt to use the external information. Finally, some routers will “repair” the packet and replace the RFC1918 IPs with the external information.
So in your case all the devices should be registering to the PBX and their Addr->IP should be their WAN/external IP of their network. The Reg. Contact field will either be their local IP or if the device/router was smart enough, it would have the same external IP as the Addr->IP. If you are seeing your PBX’s IP as their Addr->IP then you have a configuration issue some where or perhaps their router/network is doing something. Again, this is a NAT/Network based issue. Where that issue is could be on either side or both.
What do SIP debugs or traces show when these devices are REGISTERing to the PBX? What do you see in the Contact Header, the Via headers, etc, etc. That is going to give you a big clue as to what is going on.
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Framework Install Issue
Asterisk 16 Freepbx 14
Received a notification about a bunch of module upgrades including framework
Went in to do the update and got the following error for Framework (which is now disabled)
fwconsole ma install framework
Updating tables admin, ampusers, cronmanager, featurecodes, freepbx_log, freepbx_settings, globals, module_xml, modules, notifications…Done
installing files to /var/www/html…done
installing files to /var/lib/asterisk/bin…done
installing files to /var/lib/asterisk/agi-bin…done
Checking for upgrades…
2 found
Upgrading to 14.0.5.9…
-> Running PHP script /var/www/html/admin/modules/framework/upgrades//14.0.5.9/upgrade.php
In upgrade.php line 3:
syntax error, unexpected ‘(’ in /etc/asterisk/asterisk.conf on line 2
ma [-f|–force] [-d|–debug] [–edge] [–stable] [–color] [–skipchown] [-e|–autoenable] [–skipdisabled] [–snapshot SNAPSHOT] [–format FORMAT] [-R|–repo REPO] [-t|–tag TAG] [–sendemail] [–] []…
Current framework version is 14.0.5.5 and the latest is 14.0.5.20
Can you please help me fix this
thanks
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Alert-Info
The server is encoding your URL to avoid cross-site scripting. This is a good thing and (I assume) on purpose.
Since cross-site scripting is one of the easy ways to lose control over a website, this is a good thing and I suspect that there aren’t actually many ways to avoid this. Things you could try include “escaping” the < with a back-slash (which may or may not work) or copying the ring-tone to your phones and using a local alias for the wav file.
If neither of those work, you could submit a feature request and see if Sangoma will fix it. In the past, they’ve been reticent to do much with this field, since they use it as a market differentiator for their phones, but they may be able to get you going on this.
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Cisco 8961 missed calls
Hi guys,
I’m configuring a Cisco 8961, all is good (I think)…
I have a little problem, I can’t open missed calls, and I can’t get rid of them…
There isn’t any softkeys when there is a missed call…
Thanks for the help
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Happiness With Sonicwalls - It can happen!
You may want to go into your NAT Policies for your PBX.
At the very least on the outbound NAT policy, edit and go into Advanced and check the “Disable Source Port Remap” option. This tells the sonicwall to not rewrite the outbound source port, which may confuse some voip providers on which port to send inbound SIP traffic to. Combining this with increasing your UDP timeout should keep the SIP ports open and on 5060
You MAY need to do this on your inbound NAT policy for the PBX as well, but usually just outbound will work fine.
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Any physical phones handle more than 50 BLFs?
I’m not sure that Tony was endorsing FOP2. ISymphony is also a workable solution and is a Sangoma partner. I’m not sure 120 phones in either is going to be particularly useful (unless you are throwing the display to a Jumbotron). 
We’ve seen many cases historically where systems with more than a couple of active lines will fail quickly with more than 50 BLFs at the same time. The problem is that the event queue can’t handle a lot of events coming up at the same time, so if you have more than one phone listening to all of those events (and in this case, the other 119 phones didn’t have any BLF events) you will see hints start to drop off and cause confusion. With 120 phones listening to 120 BLFs, the likelihood that this would work for more than a few seconds is pretty low.
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This issue appeared when upgrade framework
That translates to “guess again!” and isn’t likely to garner any useful responses. The best answer didn’t work, but you didn’t help us understand what else is happening so that we can suggest other alternatives. If our answers don’t work, we need more information, not criticism.
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Any physical phones handle more than 50 BLFs?
@SilkBC Are the phones remote to the PBX or on the same network as the PBX?
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EPM - polycom template missing values
Hi,
I have noticed some strange issues with setting BLFs using the VVX sidecar devices.
Under the basefile editor, select MAC-Features.cfg
There are some missing values under the attendant portion.
missing 54 label and type
missing 55 label
missing 56 type
missing 57 address and label
missing 58 label
missing 59 type
missing 60 address
missing 65 label
missing 66 address
I manually added them via the basefile editor and this fixes the BLF issues on the sidecars.
features
attendant
resourceList.54.label
__b54label__
features
attendant
resourceList.54.type
__b54type__
and so on…
Hopefully this makes sense and you guys can adjust the official template
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Any physical phones handle more than 50 BLFs?
Assuming that the PC is reasonably recent (Gen 6 or better), it should be capable of driving a 4k monitor at full 3840x2160 resolution. Replace the two monitors with a small (39 to 43 inch) 4k TV, giving a display area equivalent to four normal HD monitors. There should be plenty of room for FOP2, in addition to their normal work.
4k TVs are very inexpensive now, but don’t go too cheap – you need one that doesn’t do chroma compression; see https://www.rtings.com/tv/learn/chroma-subsampling . Also, one with a ‘game’ or ‘direct’ mode for low latency. Take a laptop to a local store and confirm that the set performs well as a monitor. We have nine of these sets and they are a pleasure to use, compared to multiple HD monitors.
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EPM - polycom template missing values
The Community Forum is not a place to report bugs/feature updates. Please use https://issues.freepbx.org for these things. Otherwise it will never be considered or looked at to be put in the work flow.
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EPM - polycom template missing values
It makes perfect sense, but we’re all just users. We can’t adjust any of the commercial modules.
You might get lucky and one of the devs might create an “issues” ticket for you, but don’t hold your breath especially on EPM. Polycom isn’t a partner anymore, so getting updates into the EPM templates requires “someone” to take ownership of the udpates. That won’t be Digium/Sangoma (since Polycom is a competitor) but if one of the users is willing to do the legwork and make is a simple update, it’s more likely to happen from a ticket.
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Any physical phones handle more than 50 BLFs?
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Any physical phones handle more than 50 BLFs?
Really? That’s exactly what our front desk does… and they absolutely love it!
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Any physical phones handle more than 50 BLFs?
Look at InputDirector. Software KVM… We used it a LOT at 911 call centers, where there could be 3, 4, even 5 PCs across anywhere from 4 to 7 monitors. One keyboard and mouse to rule them all.
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