Ticket is internal. I will update this thread when it’s done.
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Queue Agents Invalid when there is a single agent
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Responsive Firewall always blocks "good" external users
Ty, this doesn’t apply to firewall as it has an internal validation mechanism to verify signatures.
As @jerrm suggested, I kept hacking and bypassed the validation by commenting out line 55 in validator.php before compiling the phar.
I’m testing my changes but I must have a syntax problem… Will keep everyone posted.
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Cant add customer to send traffic/prefix stripping
OK. Slowly - I had a small stroke at the beginning of March, so I’m not a quick to follow as I was.
You’ve listed a whole bunch of destinations, but not really a lot of paths to getting to them. For example, adding a route is literally as simple as adding a route in the “route” module.
There’s not really such a thing as a ‘customer’ in FreePBX, since that implies a multi-tenant setup and FreePBX doesn’t really ‘do’ multi-tenant very well. All of that kind of management (usage stats, etc.) is normally handled outside of the system using the reports and databases that the system maintains.
One thing that might help is that inbound calls and outbound calls are different enough that it helps to think of them as distinct things, so if you want to manipulate an inbound call’s Caller ID, you can do it simply through one of several different mechanisms. Outbound CID is manipulated the same “basic” way, but using different tools.
Billing your customers is handled completely outside FreePBX. There are plenty of billing packages out there that can read the Asterisk database and generate invoices.
Hope that helps.
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DAHDI Trunk
This morning my DAHDI trunk has stopped and on checking everything seams ok.
Nothing has changed or been updated but the log is showing Alarm Cleared on channel 1 on a Sangoma A200 card.
Anybody seen this or any pointers.
10073[2021-04-06 14:57:17] NOTICE[2844] sig_analog.c: Alarm cleared on channel 1
10074[2021-04-06 14:57:27] VERBOSE[2844][C-00000007] sig_analog.c: Starting post polarity CID detection on channel 1
10075[2021-04-06 14:57:27] VERBOSE[7495][C-00000007] sig_analog.c: Starting simple switch on ‘DAHDI/1-1’
10076[2021-04-06 14:57:36] WARNING[7495][C-00000007] sig_analog.c: CID timed out waiting for ring. Exiting simple switch
10077[2021-04-06 14:57:36] VERBOSE[7495][C-00000007] sig_analog.c: Hanging up on ‘DAHDI/1-1’
10078[2021-04-06 14:57:36] VERBOSE[7495][C-00000007] chan_dahdi.c: Hungup ‘DAHDI/1-1’
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Fax Pro SipStation Outgoing Failing
Really frustrating we still havent seen the Asterisk update within FreePBX updates to fix this faxing issue so we can get back to sending faxes with Fax Pro module…I thought we would have seen the update already as its been almost a month since Asterisk fixed it.
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Problem with Zulu UC in Android
Yep, not in my settings either. Was hoping Sangoma would have responded on this already.
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Problem with Zulu UC in Android
There were three replies from Sangoma employees already in this thread. Have you opened a support ticket?
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Problem with Zulu UC in Android
Yea, and all 3 of their replies pointed to the Wiki and area to enable it which no longer exists as shown in the above posted screenshots . Yes, I have a ticket open today.
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Problem with Zulu UC in Android
I do not use the module but just a hunch, you might have a different version installed than what is demoed on the wiki. Check for updates.
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Problem with Zulu UC in Android
Wiki was stale. I have since updated it.
If you want to use TLS signaling in Sangoma Connect, ensure the module is up to date. You can either hard set the transport to TLS in the extension advanced settings, or you can leave that set to Auto and set the default transport on the SNG Connect Settings tab.
We are still actively working thru the various SRTP usage cases which corresponds to the media encryption and opportunistic settings on the extension advanced tab. Not all signaling/media encryption combinations are supported by the Connect client.
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How to disable "Mark Answered Elsewhere"
Have you tried other phones? It could well be that it doesn’t work with Grandstream (anymore). It could also be related to the firmware version of the phones.
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Fax Pro SipStation Outgoing Failing
If you’re waiting for Asterisk versions 16.17.0 and 18.3.0, they were published less than 2 weeks ago. We have an internal ticket on this, and I expect to see rpms published to our testing repos shortly.
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Problem with Zulu UC in Android
Thank you for the reply.
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Fax Pro SipStation Outgoing Failing
Thank you for the update @lgaetz
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Sangoma Phone: Checking for firmware NOTE
The following note pops up whenever the resync value on EPM is reached:
We are running the latest firmware X.0.4.79 from Sangoma.
This only happens on Sangoma Phones. Is there a way to disable this popup??
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DAHDI Trunk
That shows battery appearing on the fxo interface at 2021-04-06 14:57:17, this clears the alarm but is seen as a wink that given your config causes a failed but spurious CID detection.
You need to find out why there was no battery prior to 2021-04-06 14:57:17 , further having battery does not guarantee that the trunk will provide dialtone (be working)
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How to disable "Mark Answered Elsewhere"
Yes, I have tried different phones and different firmwares on each as well. I believe that the code “200” (mark answered) is transfered from pbx to all extensions no matter what setting you have.
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Wrong time and DST in FreePBX 13
Hi,
we have a VM with SHMZ 6.6 and FreePBX 13.0.197.28, we noticed time is incorrect in Time Conditions, in Italy we have DST set to CEST right now, but if we set Europe/Rome we will end up with a CET hour, so our time conditions are messed up because our PBX is set to one hour back.
On the shell of the VM the command “date” shows the correct timezone and hour (CEST) and into php.ini we have:
date.timezone = "Europe/Rome" ;;;; Automatically updated by Sysadmin
We already tried changing Timezone into Sysadmin module to UTC and rebooting, then reverting back to Europe/Rome and rebooting without success.
Into Advanced Settings we have Europe/Rome as PHP Timezone set.
We don’t know how to fix this, the only workaround is to set to Athens timezone that will show the correct time until next DST change, but it’s a workaround, please could you help us?
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Sometimes caller can't hear me
It appears that you have a complex setup, a queue (1160) pointing to a ring group (600) with 6 extensions (4 SIP, 1 forwarded to a mobile, 1 IAX). Is that correct?
This results in a log more than 1000 lines. If we can’t find the trouble easily, can you temporarily route a DID directly to an extension, to see whether the problem is related to the trunk or to the complex call flow?
A few questions, based on a brief look at the log:
It appears that the queue is playing ‘aslog’ music, but the ring group is playing ‘ring’, which may override that. Which does a caller normally hear, while waiting for an agent to answer? On a failed call (where he didn’t hear the agent), did he still hear the music or ringing? If so, that would indicate that the trouble is related to the call flow (or possibly the extension), rather than a trunking or networking problem.
The username 0033429XXXXXX looks like a phone number. Is it a number on your system? The number called 0473XXXXXX is different; do you have multiple DIDs? If so, do you have a spare one that could be used for testing?
It appears that ext. 1010 is forwarded to a mobile 0603XXXXXX, but that call failed because no Outbound Route was matched. I’m guessing that the route with that match pattern has something in the CallerID field, which didn’t match the ‘anonymous’ incoming caller ID. Possibly, this interfered with extension 1009.
The INVITE to extension 1009 had to be sent 3 times before it got a reply. Is there something unusual in the network path that may have caused this (VPN, weak Wi-Fi, etc.)? Otherwise, this may indicate a problem with the Aastra or its configuration – do you have the trouble when answering from other devices?
A usual way to troubleshoot this sort of problem is to run tcpdump continuously (into a ring buffer of capture files). When the trouble occurs, you can look at the RTP to the trunk to see whether it contains voice and was sent to the proper IP address and port. However, I hesitate to do this on a Pi, for fear of wearing out the SD card. Does your Pi have other accessible storage (SSD, hard drive, network share)?
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Error updating via webui
Thanks for fixing my formatting 
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