A second will not pass while the ringtone is playing. The user should start counting seconds after picking up the phone. Currently, when I call the phone, the second starts immediately.
1 post - 1 participant
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How should I configure the following settings in Asterisk FreePBX?
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Allowlist module does not work (always passes) if the CID was entered in the privacy function
Hi,
I have this setup:
-
One inbound route for DID 12345678 (example) with CID “_X!”, so matching all numbers which are not anonymous (starting with a digit). This inbound route has allowlist active. Target for failed allowlist callers is a virtual extension with Call Screening enabled which forwards the call back to my main PBX via Follow me after the caller was identified / recoreded his name.
If the allowlist matches, the call is directly forwarded without any call screening. -
One inbound route for DID 12345678 with CID empty. This route catches anonymous calls. Privacy feature is enabled. Target is the same virtual extension as in the first route which does caller screening.
I’m fighting for some hours with this issue:
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Anoymous callers are running correctly into the privacy module asking for their number.
-
But after the number was entered, the configured destination in the “anonymous” route is completely ignored. Regardless what I set as the target, after privacy got the new CID the target of this inbound route is ignored.
-
I finally found that the anonymous caller after passing the privacy procedure straight goes into the second inbound route. It looks as privacy immediately triggers a new inbound route matching.
This behaviour in general is OK (although nowhere mentioned or configured), but now the strange part kicks in: This call wrongly always does an allowlist pass! Although the entered CID is nowhere in the allowlist nor stored anywhere else!
So, under the bottom line: Whenever a call has passed through privacy, getting it’s CID here, allowlist in a later stage will always pass this call and does not work as expected. Looks like a bug in allowlist?
1 post - 1 participant
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UCP not running, cannot connect to database
Today we had a nasty surprise. A freepbx 15 server that had been running for years was accidentally shut off and when we restarted it UCP refused to run. The error we can see in ucp_err.log is the following:
{ Error: Access denied for user ‘freepbxuser’@‘127.0.0.1’ (using password: YES) code: 1045 }
The stock install only makes user freepbxuser accesible from “localhost” so why would UCP want to use 127.0.0.1 for its host? We had to create another freepbxuser account with the same password and permissions but with host 127.0.0.1 so UCP could start. Why did this configuration change? Where is this configuration? Shouldn’t all freepbx apps use freepbx.conf in /etc to pull the database configuration?
Since UCP was not running we could not use Zulu so this prevented regular operations. I need to understand what happened in order to prevent something like this from happening in the future.
3 posts - 2 participants
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Bulk adding users to contact manager
I’m trying to find out how to add users to Contact Manager groups in bulk. I know how to edit individual users and add them to “Show In Contact Manager Groups” and “Viewable Contact Manager Groups” but I have about 200 users to change this for and would prefer not to do it one by one.
I thought that Bulk Handler > Contacts might do it but it doesn’t appear to have done anything. The import was successful but the names didn’t show up there.
1 post - 1 participant
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Viking Call Box Intercom
Hey all, Viking Technician here!
Hopefully this is the first thing that pops up on Google when you search for Viking E-30-IP (or another similar Viking Model) with FreePBX.
I figured I would organize a list here and hopefully be allowed to update as time continues.
In the Viking IP Programming Software, Press CTRL + SHIFT + V to open up the expanded menu. This shows the inbound and outbound log. Its a good resource if you are having registration issues
If getting “sip error code 0” check to see if units are banned or if the Free PBX intrusion is running. If the intrusion detection was set to not be running and the server Reboots the intrusion can cause our unit to become banned again.
Another thing I’ve noticed is the DTMF signal not being sent or received by the units. To fix this, change the DTMF Method to Inband. If that still gives issues , try cycling through the options until one works.
FreePBX uses Asterisk as a special key in their system causing door entry to be a little tricky since our default relay activation code is “**”. In the Viking IP Programming software, you may need to change the relay activation code to something else like “00”
A LOT of issues can be solved by enabling “In-Band Call Audio Progress” in the Viking IP Programming Software
As always if you run into any issues or if the above doesn’t work, you can call our product support line at 715-386-8666 or submit a ticket on our website at vikingelectronics .com
2 posts - 1 participant
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Unavailable Endpoint, Missing Contact Info
Hey there! I inherited responsibility for my company’s phone system. The person who set it up was very much so capital “I.T.” whereas I’m a bit more, “Tag, you’re ‘i.t.’.” Trying to learn this system, but not an expert by any means.
One of our Endpoints (2012) is unavailable. It still has the Aor: 2012 section but is missing the “Contact: 2012/sip:2012@x.x.x.x;5060;x-ast…” part. I’ve rebooted Asterisk and have checked things within reason, but I’m a bit out of my depth on this. All of our lines except for this one come back up.
A few things I know:
The network is whitelisted on the firewall.
The handset for our W60B Yealink base shows the correct local IP address (192.168.1.50) and the button on the base locates the phone.
I used the “Secret” for the Password in the Yealink set-up. Is that correct?
I’m happy to share logs and screenshots if they are helpful in diagnosing this. The biggest question being, how do I re-establish a Contact for an Endpoint?
Thanks!
1 post - 1 participant
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Blind transfer to outbound route doesn't work because of dial patterns
Hello community, sorry if it’s a duplicate but I couldn’t find this exact problem.
Setup is the following - I have a small office with extensions and only one of them (1100) can dial external number.
I have an outbound route with Dial Pattern ./1100.
It works fine except the following scenario:
A calls 1100 and asks to go to external number.
1100 does a blind transfer (via button on a phone).
Call is dropped for A.
I’ve investigated it and it seems that during Blind transfer A is checked against Dial Pattern and surely it fails.
As a workaround I’ve added extensions_custom.conf and it works:
[from-internal-custom]
exten => _X.,1,ExecIf(“${CONTEXT}”=“from-internal-xfer”?Set(__TRANSFER_CONTEXT=custom-test_transfer))
[custom-test_transfer]
exten => _X.,1,Set(CALLERID(num)=1100)
But for me it looks insecure - A can call B and ask Blind transfer to external and it will be allowed.
It’s a quite simple setup and I can’t get why it doesn’t work for me.
My system is FreePBX 16.0.40.7.
1 post - 1 participant
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DNS entry in Google Domain / Secure https
Hi ,
I am trying to get a static dns entry for sangoma free pbx in Google domain , the client has his existing domain set in Google FYI… willing to configure secure https access, to have phone App works and phones provisionning… BTW i will go with LEts Encrypt for the ssl certificate .
Has anyone already experienced such manner ?
Hope to find a well defined steps !!!
2 posts - 2 participants
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SSL Certificate
Friends, I’m using a subdomain to access Web Call, custom extension number, but not sure how to issue an SSL certificate. I don’t believe this can be done, in this case, through cPanel. Is it possible to have freepbx issue a certificate and avoid all the security messages that come up in the user browser?
1 post - 1 participant
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Stop Audio with ARI
Hi!!!
I tried to play audio(tt-monkeys) with AGI(pyst2)
During tt-monkeys playing, i tried to start ARI (EXEC Stasis) and play audio(hello-world) using REST API // POST /channels/{channelId}/play/{playbackId}
How can I stop tt-monkeys when hello-world start???
1 post - 1 participant
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What happened to the Cisco phone 3PCC documentation? I'm posting it here BTW
Sangoma had a very nice article - not written by them but written by a community member - that detailed how to add a Cisco 3PCC phone into FreePBX on the Sangoma Help (digium.com) website. But that article is now missing - even though many other articles are still present. In fact, searching that site for the keyword “3PCC” brings up many Cisco phone related articles, but none involved with 3PCC.
I have included screengrabs of the article here.
1 post - 1 participant
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Show inc. calls on several extensions
Hello,
i got a question.
I have several Groups of extensions.
For this example lets say i have my Marketing Group containing 5 extensions.
If one of those 5 extensions gets called, i want to display the incomming caller id and the extension called on the other 4 extensions. But only display, no ringing.
Is this somethis i can configure on FreePBX or do i have to configure this on the PhoneSoftware directly?
Thanks in advance!
1 post - 1 participant
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Online Blacklist / Reputation check using Tellows API
Hi,
I was wondering if anyone has ever done a reputation check of callers against the Tellows database. They have an API for checking, and there is a docker project for Asterisk which requires a “Makro”. Not sure if and how this would work for FreePBX as well.
Marco
1 post - 1 participant
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Apply Config must be pressed for changes to take effect
“As long as you don’t hit Apply Config then those changes have no impact on the current running config”
Quoting from a similar old post as I too always thought this was the case. However, today we noticed changes being fully (and also somewhat) applied to FM/FM without Apply Config being pressed.
This can be replicated with the following:
Follow Me 16.0.23
Asterisk 20.5.2
FreePBX 16.0.40.7
Also replicated in PBXact.
Created EXT:001 with FM/FM to 07123456789#.
Pressed Apply Config…
Rang extension internally - It rang the mobile number OK.
Changed the FM/FM number to 07987654321#, pressed Submit but did not Apply Config.
Rang the extension internally again, which rang the new number input into the FM/FM list.
Same happens when calling from outside to the inbound route.
Thanks.
4 posts - 3 participants
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Possible hack?
Hello!
We have a few boxes that have been communicating with 177.74.233.157 on port 8090 and we’re trying to figure out why and what it is. I’ve run fwconsole validate on a couple and this is what I see:
And here is what it appears to be transferring:
Does anyone know if this could be malware? Or are these valid? It appears to be transferring video files.
Thanks!
3 posts - 3 participants
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Freepbx server cant call inbound and outbound calls, Our trunk is offline how to troublesoot this?
Freepbx server cant call inbound and outbound calls, Our trunk is offline how to troublesoot this?
7 posts - 3 participants
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Registed pjsip trunks change status to rejected after two weeks, reload resolves problem
Dear FreePBX community,
Over the past years, I have a quite annoying issue with my FreePBX setup related to SIP registrations to SIP providers. My FreePBX makes use of Asterisk 16.3.0. While the FreePBX as such runs stable at least over months, after about two weeks SIP registrations to SIP providers change their state from Registered to Rejected, as obtained from:
asterisk -rvvvv
pjsip show registrations
This happens regularly about each two weeks. However, when I reload the configuration on the FreePBX shell by executing “fwconsole r”, the FreePBX registers again and succeeds. I do not doubt that there were indeed (short) network outages when the registration expired and no renewal was possible. However, I wonder whether FreePBX / Asterisk can be configured in a way that there are further retries instead of simply giving up.
I got the impression that the default behavior is to try registrations a few times and then to give up forever. Only manual interaction resolves the issue. Is there a way to perform follow-up retries on a regular basis, e.g. once an hour, if the standard time-out causes giving up retries on a shorter repetition rate?
Do you have any suggestions as how to improve the setup? Thanks a lot for your comments.
Best regards,
Peter
1 post - 1 participant
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Allowlist module - wrong logic or wrong processing order?
Hi,
I’m currently implementing an Antispam system with FreePBX which basically should do (in more or less this order):
- Well-known numbers in a list should bypass all checks and directly be forwarded to the final destination
- Anonymous callers shall enter their number (Privacy feature)
- Number at this point is checked online for Spam with Superfecta → Special route in case of Spamscore
- Call will be routed to a virtual extension doing caller screening
- Screened call will be forwarded to the final destination
All this independently for three different incoming DIDs.
I’ve learned a lot about the capabilities of FreePBX in the last days and everything is working great - with the exeption of the bypass of well-known numbers. I’m really completely desperated with the allowlist module as in my opinion it has a wrong logic (or at least a wrong position in the call processing).
The allowlist processes well-known “friendly” numbers. In my understanding this should work by “extracting” these numbers in the module from further processing which means they can be immediately forwarded to the next destination (Dynamic routes, queue, ring group, extension etc.). There is no need that a well-known number does any further processing by the privacy feature or superfecta - the number is already approved by the fact it’s in the allowlist.
Unfortunately Allowlist exactly behaves the opposite. It extracts all unknown callers from the inbound processing, completely killing the ability to be further checked by privacy and superfecta. While well-known and approved numbers continue being processed by Privacy and Superfecta.
I’m sure there is a reason for this inverted logic, because @mitchmitchell must have had a reason to implement it like this … so I guess it’s me who does not understand why it was done this way. However, I don’t see a proper way to complete my project with this behavior.
Even the inverse logic would be something that can be dealt with - if the allowlist “order” would not be this extremely high in the call processing. I’m not sure if this can be influenced in FreePBX in any way, but allowlist unfortunately catches all the interesting “unknown” callers quite at the beginning of the process. It “steals” Privacy and Superfecta the ability to get their hands on the call.
Finally I still would really love a solution for this dilemma. I did not look that deep how modules are linked in FreePBX of if they are in a language which allows modification. Thinking about the ability of a “dirty hack” to work around this issue like:
- Maybe finding how to invert the logic in a way that allowed numbers take the exception route instead of non-allowed
- Maybe shifting the point where allowlist is processed in a route to a point behind the Privacy and Superfecta processing
Both would solve the issue, with the first one being be much more clear and elegant in my opinion.
Any other ideas how to make allowlist work with Privacy and Superfecta are highly welcome
PS: Privacy can be worked around by catching anonymous callers with a second inbound route and send them to an IVR with following voicemail box. Not that elegant, but at least a solution. But for Superfecta I don’t have a solution right now.
4 posts - 2 participants
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GraphQL API
Good part of the day!
I need to get calls from freepbx, and I do this:
$query = “{
fetchAllCdrs(first: 10000, after: 0, orderby: date, startDate: "$date", endDate: "$date") {
cdrs {
id
cnum
did
disposition
duration
calldate
outbound_cnum
}
}
}”;
$response = Http::withToken($this->getAccessToken())
->post(config('free_pbx.url') . '/gql', [
'query' => $query
])
->json();
The problem is that I can’t pass startDate and endDate in format (‘Y-m-d H:i:s’), and also i can’t filter by ‘outbound_cnum’ field in the query.
Maybe there is other way to get calls filtered by date (‘Y-m-d H:i:s’) and outbound_cnum?
1 post - 1 participant
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Office 365 calendar recurring events issues
Stange issue with reoccuring events in office 365 linked calendar
problem is as follows, the customer has a on call calendar set up where engineers are oncall every 2 weeks, in office 365 the calendar shows it correctly with them alternation weekly
But in the Calendar in FPX they are sown as oncall every week even though the event when clicked on in FBX shows it as recurring event as ‘weekly’ and recurring every 2 weeks there seems to be some issue with the data being passed
Does the FPX calendar support recurring events every 2 weeks or just weekly recurring events ?
1 post - 1 participant
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