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Freepbx SNG7 - proper ugrade from 32bits version using conversion tool - would need a little help

February 4, 2018, 8:49 pm
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@probegtze wrote:

I’ve upgraded my system from 10.13.66 to SNG7 using the following method:

10.13.66 32bit to 10.13.66 64bit using conversion tool
Upgraded 10.13.66 using
yum -y install http://package1.sangoma.net/distro-upgrade-1712-6.sng7.noarch.rpm
and
distro-upgrade.

This seemed to work well, but once I was done configuring the trunks, I discovered that all my announcements had not transferred, and since my phone system has a time condition that will direct the call to an announcement if the customer is calling outside of business hours, my phone system failed to come up. Since the conversion tools brings everything, it would seem useful to brings announcements too.

I then realized that some modules from a previous era had followed from 10.13.66 32bit. - the one that struck me was the Digiums ones; I have a ISO install of SNG7 and confirmed that some modules that were present in my original 10.13.66 32bit were now appearing in my updated PBX.

Later, I noticed that the following:
±-----------------------------------------------------------+
| Your system is currently up to date! |
| Your PBX is up to date. |
| Also 4 Uninstalled modules. |
±-----------------------------------------------------------+

  • I checked what was “Not installed” and “sipstation” was one of them. Trying to install the module got me this screenshot:
    image
    image.png1917x1076 177 KB

I also noticed that Queue Pro want not installed and that was causing issues in the fail2ban log: something was looking for the presence of audio files to work with the queues, so I reinstalled Queue Pro and the messages in the log are no longer present (after renaming the log to get cleaner data) - but there are quite a bit of “error loading module” left in the log. I tried pasting the Fail2ban log here, but that made the post exceed the limit of 32000 characters.

So far, I’m having a blast with Freepbx, but unfortunately, I run into many bugs that may or may not have consequences… Right now, SIPSTATION really is the one that bugs me, as I’d like to get Sangoma phones for my team.

(additional info: I tried executing from CLI fwconsole ma downloadinstall sipstation
results:

[root@FreepbxV3 ~]# fwconsole ma downloadinstall sipstation --force
No repos specified, using: [standard,extended] from last GUI settings

Downloading module 'sipstation’
Processing sipstation
Downloading…
617497/617497 [============================] 100%
Finished downloading
Extracting…Done
Download completed in 7 seconds

In utility.functions.php line 207:

trying to set keyword [SS_API_URL] to [] on uninitialized setting::

ma [-f|–force] [-d|–debug] [–edge] [–color] [–skipchown] [-e|–autoenable] [–skipdisabled] [–snapshot SNAPSHOT] [–format FORMAT] [-R|–repo REPO] [-t|–tag TAG] [–onlystdout] [–willupdate] [–securityonly] [–sendemails] [–] []…

Thanks for those who will take the time to look at this post!

*** Update: I just reinstalled everything from scratch, used the conversion tool again to go from 32bit to 64bit. Right after the install, here’s what I got:
exit: 1
Whoops\Exception\ErrorException: Invalid argument supplied for foreach() in file /var/www/html/admin/modules/parking/functions.inc/registries.php on line 16
Stack trace:

  1. Whoops\Exception\ErrorException->() /var/www/html/admin/modules/parking/functions.inc/registries.php:16
  2. Whoops\Run->handleError() /var/www/html/admin/modules/parking/functions.inc/registries.php:16
  3. parking_check_extensions() /var/www/html/admin/libraries/usage_registry.functions.php:33
  4. framework_check_extension_usage() /var/www/html/admin/libraries/usage_registry.functions.php:420
  5. framework_list_extension_conflicts() /var/lib/asterisk/bin/retrieve_conf:789

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Check Firewall is OK?

February 4, 2018, 11:03 pm
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@abcym15 wrote:

I have a newly installed Distro on Vultr VPS (PBX Firmware: 12.7.4-1712-2.sng7, Asterisk 13, FreePBX 14.0.1.36). It works great and I’m using the FreePBX firewall as usual.

The firewall is correctly stopping anyone browsing to port 80 (unless they’re on the whitelist) but for some reason I am getting notices from Fail2Ban about IPs being blocked. I’ve got responsive firewall on for CHAN_SIP and SIP is in the local zone only.

I have this identical setup on a number of machines and I never hear from Fail2Ban, presumably because the firewall prevents unidentified IPs being able to attempt to register more than 9 times (which is my current Fail2Ban trigger level). So my question is, is there a way of checking that the FreePBX firewall is actually OK and is handling my SIP traffic as it should be? There are no clues on the GUI to suggest anything is wrong. The NIC is in the Internet zone.

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Configuration that would ring a manager's phone for listening-in opportunity on new inbound call

February 5, 2018, 11:18 am
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@klong wrote:

Hello,

I am trying to find a configuration that would do the following:

When extension 1234 receives a call and answers it, Asterisk would immediately place a call to extension 6789, and if 6789 picks up, he would be able to hear the call in progress, but be muted himself.

It would also work fine if when the call came in it started ringing both parties, but if 6789 picked up first it would not be acceptable for the caller to be notified the call had been answered until 1234 picked up, or to hear 6789.

Is this possible?

Regards,

Kevin Long

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Diverting a SIP engin trunk

February 6, 2018, 4:39 am
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@mixhali wrote:

Hi guys I have customer that i have deployed freepbx to in Australia. We use a SIP trunk from Engin. t the end of each day the customer wants to divert the engin trunk to mobile. I have tried dialing the standard divert number which is *21XXXXXXXXXX# however no success. I have edited the dial plan to accommodate this number pattern but still no good. how can i achieve this?

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Call Recording Retention

February 6, 2018, 1:41 pm
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Security Warning PBXAct

February 6, 2018, 9:29 pm
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Freepbx asking for a password on outgoing calls

February 7, 2018, 9:33 am
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@pwilson330 wrote:

Hi,

I’m in the process of building a new freepbx system. I build my trunk as pjsip (since that’s what matches my extensions) and I’m putting in the correct username and password for the outgoing routes, but when I try to make an outbound phone call, it asks for a password. Any suggestions?

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System Down: GUI Errors out - Internal address not defined in Global Settings. Please set internal address, or

February 7, 2018, 1:01 pm
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@VoIPTek wrote:

No matter what I do, I cant get back to any web gui interface.

Exception Error:
Internal address not defined in Global Settings. Please set internal address, or choose a different provision address in the template.

This will not let me change anything, only displays the error page.

This was basically a clean install, then added a Cisco Template and 1 extension.

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Enabled extension routes module with license, now all sorts of errors

February 8, 2018, 11:27 am
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@Riick370 wrote:

I’m trying to apply config and I get below. I have to disable it to get apply changes to work again.

exit: 255
HANDLED-ERROR: FreePBX\modules\Endpointman->doDialplanHook() isn’t there, but the module is saying it wants to hook. This is a bug in FreePBX\modules\Endpointman
Unable to continue. Undefined index: timezone in /var/www/html/admin/modules/extensionroutes/functions.inc/functions.inc.php on line 58
#0 /var/www/html/admin/modules/extensionroutes/functions.inc/functions.inc.php(58): Whoops\Run->handleError(8, ‘Undefined index…’, ‘/var/www/html/a…’, 58, Array)
#1 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(95): extensionroutes_hookGet_config(‘asterisk’)
#2 /var/lib/asterisk/bin/retrieve_conf(864): FreePBX\DialplanHooks->processHooks(‘asterisk’, Array)
#3 {main}

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Asterisk Trunk Dial Options not working after latest upgrade of Freepbx

February 9, 2018, 1:34 am
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@TimmiORG wrote:

Hi all,

I’m not sure how to write it but after the latest upgrade of Freepbx modules my Asterisk Trunk Dial Options stopped to work.

I was using something like that: Tb(p-preferred-identity^trunk-name^1)

exten => trunk-name,1,NoOp(outbound CID: ${CALLERID(name)})
exten => trunk-name,n,Set(OUTCID=${IF($["${CALLERID(name):0:4}"="CID:"]?${CALLERID(name):4}:${CALLERID(name)})})
exten => trunk-name,n,Set(PJSIP_HEADER(add,P-Preferred-Identity)=sip:${OUTCID}@domain.com)
exten => trunk-name,n,Return()

This was working in the past.

Now I can see the following:
[2018-02-09 08:40:29] VERBOSE[14824][C-00001140] pbx.c: Executing [s@macro-dialout-trunk:31] Dial(“PJSIP/41-000014c5”, “PJSIP/01757255009@trunk-name,300,Tb(p-preferred-identity^trunk-name^1)b(func-apply-sipheaders^s^1)”) in new stack
[2018-02-09 08:40:29] VERBOSE[14824][C-00001140] app_stack.c: PJSIP/trunk-name-000014c6 Internal Gosub(func-apply-sipheaders,s,1) start
[2018-02-09 08:40:29] VERBOSE[14824][C-00001140] pbx.c: Executing [s@func-apply-sipheaders:1] ExecIf(“PJSIP/trunk-name-000014c6”, “0?Set(CHANNEL(hangup_handler_push)=crm-hangup,s,1)”) in new stack
[2018-02-09 08:40:29] VERBOSE[14824][C-00001140] pbx.c: Executing [s@func-apply-sipheaders:2] NoOp(“PJSIP/trunk-name-000014c6”, “Applying SIP Headers to channel”) in new stack
[2018-02-09 08:40:29] VERBOSE[14824][C-00001140] pbx.c: Executing [s@func-apply-sipheaders:3] Set(“PJSIP/trunk-name-000014c6”, “SIPHEADERKEYS=”) in new stack
[2018-02-09 08:40:29] VERBOSE[14824][C-00001140] pbx.c: Executing [s@func-apply-sipheaders:4] While(“PJSIP/trunk-name-000014c6”, “0”) in new stack
[2018-02-09 08:40:29] VERBOSE[14824][C-00001140] app_while.c: Jumping to priority 8
[2018-02-09 08:40:29] VERBOSE[14824][C-00001140] pbx.c: Executing [s@func-apply-sipheaders:9] Return(“PJSIP/trunk-name-000014c6”, “”) in new stack

Would be great if someone is able to help.

Best regards
Timmi

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Invoke a function on Asterisk and get response through SIP requests

February 9, 2018, 4:07 am
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@Arunbalannv wrote:

I have an FreePBX server and a custom developed client for extension registration. And I would like to trigger some function on this Asterisk server by sending a SIP request.
And also I need to get back the result of the executed task to the client application.
In my situation, a direct HTTP request sending is not possible between Asterisk server and the client.
So I require a solution based on SIP itself. Is there any way to do this using SIP OPTIONS, SIP MESSAGE or by means of any other requests?

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FreePBX Will not save network settings at all

February 9, 2018, 12:02 pm
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@GamerSam wrote:

Hello again. I am relatively new to FreePBX. I had posted before asking for help and advice. My boss had me make us a FreePBX system for our Stores phone systems, using Dell Optiplexs. We got the one for our main store up and running, its working fine, and we have no issues. He then tasked me with making one for our second store. Again using a Dell Optiplex and a AEX410 card. Everything was working fine, until about a week ago. I turned it back one to work on it more, and oddly, it was not getting an ip address. I have looked up online how to set the ip via Linux console commands. But no matter what, it doens’t seem to save them. I cannot access the GUI sometimes, since it does not keep the IP address. And when I do, it does not let me save the Networking Settings in there either. I have made a few attempts, nothing, the settings just revert every time I restart. It definitely recognizes the Ethernet port. Now when I log into the GUI it says cannot connect to Asterisk, and half the things are unassailable. Does anyone have any good advice, or might I have to reinstall the entire thing?

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System Firewall FreePBX

February 10, 2018, 7:26 am
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@igor_stojanoski wrote:

Missing Requirements:
The File “/usr/lib/sysadmin/includes.php” must exist.

In custom installation on Centos 7 i have this error. Is there any way i can fix this problem ?
I really like to enable the system firewall because using
systemctl start firewalld is not working and i have added my public ip address on whitelist using this command: firewall-cmd --permanent --zone=trusted --add-source=1.2.3.4

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Help with Polycom VVX BLF Call Pickup

February 12, 2018, 8:43 am
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@xptpa2020 wrote:

We are a Yealink shop. We recently won a Polycom customer who kept their Polycom VVX600 phones. We are using FreePBX v13 with EPM (shout out to the EPM guys - it is very very good). The only issue that we can’t seem to tackle is how to setup call pickup when a monitored BLF is flashing. Yealink just works (it acknowledges the ** pickup code). With the vvx600, a monitored extension’s BLF is flashing, indicating an incoming on that extension. The users then touches that BLF on his phone, but it calls that ringing extension, rather than picking up the inbound call that was coming to that extension.

I am hoping a Polycom shop can give some direction to us.

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Yealink Version 82 Firmware in Commercial Endpoint Manager

February 12, 2018, 8:55 am
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What is this error/warning?

February 12, 2018, 11:44 am
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@jcroy wrote:

Can someone explain to me what this means?
Seeing a lot of this in our full logs. Havent seen this in the past until we made some networking changes.

[2018-02-09 05:22:30] WARNING[18329] res_pjsip_registrar.c: Endpoint ‘anonymous’ has no configured AORs
[2018-02-09 05:22:30] WARNING[9081] res_pjsip_registrar.c: Endpoint ‘anonymous’ has no configured AORs

Looking it up has not revealed the answer.

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Could not get banned list

February 12, 2018, 12:49 pm
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@kyiu wrote:

I just discovered the server is unable to show the banned list in the intrusion detection. It’s possible this error is caused by the recent issue I was trying to fix (Broken Sysadmin module).

When I go to Sysadmin–>intrusion detection, I got the following error:
Exception
Could not get banned list
COPY
Click Here Report this to the PBX bug tracker
Stack frames (4)
3
Exception
/var/www/html/admin/modules/sysadmin/Sysadmin.class.php1719
2
FreePBX\modules\Sysadmin getFail2BanList
/var/www/html/admin/modules/sysadmin/functions.inc/intrusion.php58
1
sysadmin_get_banned
/var/www/html/admin/modules/sysadmin/page.sysadmin.php377
0
include
/var/www/html/admin/config.php560

No idea how to fix this!!
Please help…

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Seeking opinions on VoIP monitoring and analysis

February 12, 2018, 7:12 pm
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@billsimon wrote:

I’m in need of something that can do SIP capture and analysis as well as RTP capture and analysis. HOMER is ok for the cost (free) but doesn’t do RTP and has a bunch of other quirks. VoIP Monitor (voipmonitor.org) seems to have it all and I’m planning a trial. What else do people use?

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Is FreePBX is supported for WEBRTC?

February 13, 2018, 12:44 am
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@karthick wrote:

UCP phone shows, Phone Status only supported over https.

I am using FreePBX 14.0.1.20 version.

I want to use my web page as a SIP Client.

If FreePBX is support for WEBRTC, how to configure it ?

Is any other additional software needed for my requirement ?

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Client Question (spoofing an external caller ID)

February 13, 2018, 9:44 am
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@Red wrote:

For all you experts out there is it possible to spoof an incoming number, with the pbx system. The call comes in to an extension, gets transferred to another extension, but instead of the originating caller ID, they want to use the ID of the transferring extension. This hides the originating caller ID from the intended recipient.
I don’t know why you would want to do this but client used to have a Toshiba system, where it was possible. Now they are using Grandstream/PBX.
They are actually wanting to do this from external number to external number using the phone system as a switchboard, for clients that work from home.

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