I see this was posted before, but no answer given
Sometimes a caller will call just before closing, then if staff is unable to answer the call before closing time, the caller will be stuck waiting until max hold time.
I am not seeing anything in the queue menu to address this.
never mind: found it
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The simplest way is for developers to simply bundle their signed keys into their package. It doesn’t require any lookup, and is provably secure, including offline. This is poorly documented, unfortunately, which is my fault.
The other thing is that FreePBX shouldn’t be using the SKS keyservers at all - simply change to use the new keyserver (the name of which I have temporarily forgotten, blame all the booze), which requires email authentication before allowing updates. That immediately, and trivially, solves the problem.
The FreePBX primary key should be read only ANYWAY. There is nothing that can prove it is more original than is already - or the new one - as it’s hard-coded in numerous places.
Yeah, I have no idea what could be causing it because it’s a lot of things. It could be anything from the local network, the itsp, or a configuration issue. So, where I would start is to try and eliminate the network as a possibility, since 90% of VoIP issues always seem to be network related.
First thing I would do is look at the log files in /var/log/asterisk/ and see if that gives you any clue as to what is going on.
Next, I would look at the traffic to and from the FreePBX system and the phones, itsp, etc. and try to reproduce the issue. You can do this with something like tcpdump app, or some other packet capture application. There is a SIP packet call flow, and those packets must lineup or there is a possibility one side will hangup the call.
Link to tcpdump instructions:
Thanks for the pointers. I’ll try and capture a call with tcpdump… the trouble is that it may occur with two calls and then be fine for a period and then it will happen again. Typically if a user calls right back it will answer no problem.
The logs in my second post where from the /var/log/asterisk file just filtered down to the specific calls.
ISP is AT&T fiber so I don’t expect too many issues on that front (but I won’t put anything past them). If it is a configuration issues, where could I begin to look to troubleshoot that?
As far as configuration goes? I have no idea, there are many settings and combinations of settings that could be causing it. But it’s unlikely settings.
Hey James,
Yeah, everybody who has a key signed with the old key should go through the process of getting their keys signed with the new key.
Best wishes,
Matt
As mentioned by others, it’s legitimate. There have been a lot of complaints caused the old freepbx module signing key being poisoned on public keyservers so at the end of the year we’re adding support for new signing keys that this framework update covers. We pushed it out as security released so that it would get to as many systems as possible.
Matt
FYI: This new key can be just as easily poisoned as the old key since the poisoning takes place on the GPG Ring of Trust. This is why the server should be set to only hkps://keys.openpgp.org. The longer FreePBX keeps referencing SKS (which is broken) the sooner you’ll have someone poisoning the new key again. In fact the old key is perfectly fine on hkps://keys.openpgp.org since it was scrubbed and cleaned of all poisoning. (and on that note I don’t think issuing a new key was even needed since it can be poisoned just as easily as the old key)
The new key. Or any key. Will always been vulnerable to poisoning on “bad” networks. Like SKS (and actually ALL networks freepbx currently uses)
Yeah, I think we’re going to update that too - just trying to change one thing at a time since security stuff is tricky and didn’t want to throw a whole bunch of changes out at the same time.
Thanks for your help and good advice on that side of things too!
Matt
Thanks… I’ve asked our Portal team to investigate.
Login to your system via ssh, do an ‘asterisk -rc’ to connect to it, and at the console type ‘pri set debug on span x’ (where x is the span number you’re debugging). The console should output pri specific protocol debugging at that point to give some more clues as to what’s going on.
Matt
Hello,
I have a Sangoma S500 and was wondering. I setup a transfer to voicemail button however it is a three step process to get to the extensions list. The button is setup as an XML API, Rest Voicemail transfer button.
First screen press it changes my screen to show voicemail at the top and then extension on the screen which is blank. The only thing I can do is hit the check mark.
Second screen shows only extension and allows me to input someones extension manually.
If I put no extension in and just hit the check mark the third screen shows what I want. A list of all my extensions. Once I get here I can easily transfer to who I need to.
My question is. Can I bypass the first two steps so that when I press the XML API button to take me straight to my list of extensions?
for those who find this i believe the OP found the ‘leave empty’ , its on the capacity options tab ~
it’s described as follows:
Determines if callers should be exited prematurely from the queue in situations where it appears no one is currently available to take the call. The options include:
* Yes Callers will exit if all agents are paused, show an invalid state for their device or have penalty values less than QUEUE_MAX_PENALTY (not currently set in FreePBX dialplan).
* Strict Same as Yes but more strict. Simply speaking, if no agent could answer the phone then have them leave the queue. If agents are inuse or ringing someone else, caller will still be held.
* Ultra Strict Same as Strict plus a queue member must be able to answer the phone ‘now’ to let them remain. Simply speaking, any ‘available’ agents that could answer but are currently on the phone or ringing on behalf of another caller will be considered unavailable.
* Loose Same as Yes except Callers will remain in the Queue if there are paused agents who could become available.
* No Never have a caller leave the Queue until the Max Wait Time has expired.
[2019-12-23 08:24:25] VERBOSE[32687][C-00000001] bridge_channel.c: Channel PJSIP/104-00000001 joined ‘simple_bridge’ basic-bridge <a81c4018-b0ae-42ba-a8eb-49e6a08252d5>
[2019-12-23 08:24:25] VERBOSE[32669][C-00000001] bridge_channel.c: Channel PJSIP/Skyetel_SE-00000000 joined ‘simple_bridge’ basic-bridge <a81c4018-b0ae-42ba-a8eb-49e6a08252d5>
[2019-12-23 08:24:25] VERBOSE[32669][C-00000001] bridge_channel.c: Channel PJSIP/Skyetel_SE-00000000 left ‘simple_bridge’ basic-bridge <a81c4018-b0ae-42ba-a8eb-49e6a08252d5>
So the trunk side dropped the call within the same second after answering. Possibly, a codec issue or a refused re-invite. With luck, a SIP trace may be all we need to diagnose. At the Asterisk command prompt, type:
pjsip set logger on
and call in until you get a failure, then post the log. Please don’t use Google Drive or any other service that requires the reader to log in or provide any information. We recommend
pastebin.freepbx.org
I believe that you can post links now, but if not, just replace the last dot with %2E for example
pastebin.freepbx%2Eorg
which can be pasted into a browser without editing.
Also, check
https://docs.netgate.com/pfsense/en/latest/nat/configuring-nat-for-voip-phones.html
Moving hosts, and I need to know the best way to move an active machine, with expired 1 year license for Endpoint Manager. The easiest way possible would be best. Can either do backup then restore or do migration. I can go on to deactivate on the portal but a, can you do that on an expired license and still move it over to another machine? I have the deployment ID. just one to simply move machines, ip addresses. from one hosted cloud service to another. Thanks!
I fixed it…
I had to uninstall UCP and all dependencies. Zulu and all dependencies.
Reinstalled everything except Zulu. Working now.
Thanks
Hi. I am trying to install freepbx-15 latest over an old install originally under DEbian 9, which I copied to another computer and upgraded the computer to Debian buster. Here is what I get when I start install in the freepbx directory, after answering all the questions:
Checking if SELinux is enabled…Its not (good)!
Reading /etc/asterisk/asterisk.conf…Done
Checking if Asterisk is running and we can talk to it as the ‘asterisk’ user…Yes. Determined Asterisk version to be: 16.5.1
Checking if NodeJS is installed and we can get a version from it…Yes. Determined NodeJS version to be: 10.15.2
Preliminary checks done. Starting FreePBX Installation
Checking if this is a new install…Partial
Database installation checking credentials and permissions…Connected!
Initializing FreePBX Settings
PHP Notice: Undefined property: FreePBX::$Config in /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/Cache.class.php on line 233
PHP Notice: Undefined property: FreePBX::$Config in /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/Hooks.class.php on line 82
PHP Fatal error: Uncaught Error: Call to a member function get_conf_setting() on null in /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/Hooks.class.php:82
Stack trace:
#0 /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/Hooks.class.php(27): FreePBX\Hooks->updateBMOHooks()
#1 /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/Hooks.class.php(254): FreePBX\Hooks->getAllHooks()
#2 /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/Hooks.class.php(349): FreePBX\Hooks->returnHooks(2)
#3 /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/Config.class.php(1118): FreePBX\Hooks->processHooks(‘AST_FUNC_EXTENS…’)
#4 /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/Config.class.php(835): FreePBX\Config->removeSetting(‘AST_FUNC_EXTENS…’)
#5 /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/Config.class.php(263): FreePBX\Config->remove_conf_settings(Array)
#6 /usr/src/freepbx/amp_conf/htdocs/admin/libraries/BMO/Self_Helper.class.php(124): FreePBX\Config->__construct(Object(FreePBX))
How can I get around this so I can continue this install?
Thanks in advance for any suggestions.