@drummerjoe wrote:

We have phones external to the network. They are connected via openvpn (epm and yealink). We have a blf button assigned to feature code *76 and their extension so that passing it turns the light red or green depending on their dnd status. However, it randomly doesn’t work, and it happens on random phones. The light will just always stay green no matter what their dnd status.

Is there anything that needs to be done in the firewall since these are external phones to ensure presence always works? I assumed the VPN would prevent things like this from happening vs doing direct sip connections from the phones to freepbx on the public ip.

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@AmidouFlorian92 wrote:

Hi I have this message on my asterisk CLI. I just want to stop showing this how can I do?
[2019-08-22 13:23:35] NOTICE[11077]: chan_sip.c:28752 handle_request_register: Registration from ‘“1700” sip:1700@199.16.131.19’ failed for ‘77.247.110.201:6388’ - Wrong password

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@thiwanka2036 wrote:

I am trying to migrate settings from freepbx 12.0.76. to 14.0.11 as per the instruction first i have ran the script on new machine and copy the conversion id and ran the script on donor machine and enter the copied ID, donor machine started the conversion process but new machine keeps on waiting for donor? Tried manually copying /var/www/html/conversion_backup.crypt on the DONOR machine, to /tmp/conversion_backup.crypt new machine it doesn’t work as well

[root@Flatrix-PBX-CC asterisk]#
Checking that ‘curl’ exists … OK!
Validating sha256 integrity … OK!
Trying to download converter to /tmp/tmp.1kJVmIiP1C … Complete!
Validating download … OK!
Starting FreePBX Converter version release/20171122r1
Testing connectivity to Conversion server…Success!

FreePBX Conversion Wizard

The FreePBX Conversion Wizard needs to be run on two machines, firstly on the
NEW machine, which must be an ACTIVATED FreePBX Distro machine, and then it
must be run on the DONOR machine.
The DONOR machine is the machine that is currently processing calls, and is
the machine that will be migrated to the NEW machine. No changes will be made
to the DONOR machine, and this script will not stop or restart any services
that may cause an outage.
If this is the NEW machine, just push ‘Enter’ to prepare this machine

Enter Conversion ID (leave blank if this is NEW):
Testing FreePBX functionality … Success! (Version 14.0.11)
Getting Deployment … 85890368
Getting all Module versions (for conversion) … OK
Reserving a conversion slot … Reserved!

The Conversion process is now ready. Please run the script on the
DONOR node now, and when asked for a slot identifier, please enter
the following ID:

    4485c8e4-6d54-4e6a-b265-67e6f9918e54

[/] Waiting for Donor…

Donor machine

[root@Ikman html]#
Checking that ‘curl’ exists … OK!
Validating sha256 integrity … OK!
Trying to download converter to /tmp/tmp.DdWaT5xuXp … Complete!
Validating download … OK!
Starting FreePBX Converter version release/20171122r1
Testing connectivity to Conversion server…Success!

FreePBX Conversion Wizard

The FreePBX Conversion Wizard needs to be run on two machines, firstly on the
NEW machine, which must be an ACTIVATED FreePBX Distro machine, and then it
must be run on the DONOR machine.
The DONOR machine is the machine that is currently processing calls, and is
the machine that will be migrated to the NEW machine. No changes will be made
to the DONOR machine, and this script will not stop or restart any services
that may cause an outage.
If this is the NEW machine, just push ‘Enter’ to prepare this machine

Enter Conversion ID (leave blank if this is NEW): 4485c8e4-6d54-4e6a-b265-67e6f9918e54
Testing FreePBX functionality … Success!
Checking Slot ID … OK!
Getting modules to convert … Complete! 114 modules
Dumping astdb … Complete!

— WARNING — WARNING — WARNING — WARNING —

Some directories are large, and may slow down the backup. You can copy these
directories across later, after the conversion is complete. This will speed up
the conversion. Answering 'Y’es will embed them in the backup that is sent to
the new machine

Directory /var/spool/asterisk/monitor? (304.05GB) [yN] N

Creating encrypted backup for new machine. This may take some time.
Backing up the following:
54M /var/lib/asterisk/moh
0 /var/spool/asterisk/asternic
22M /var/spool/asterisk/backup
4.0K /var/spool/asterisk/fax
4.0K /var/spool/asterisk/incron
4.0K /var/spool/asterisk/meetme
8.0K /var/spool/asterisk/outgoing
52K /var/spool/asterisk/sysadmin
4.0K /var/spool/asterisk/system
292K /var/spool/asterisk/tmp
12K /var/spool/asterisk/voicemail
224K /tftpboot
17M /var/lib/asterisk/sounds/custom
Complete!
Sending backup details to new machine … Complete!
Skipping module accountcodepreserve (Nothing to convert)
Skipping module amd (Nothing to convert)
Module announcement … announcement
Module areminder … areminder areminder_calls areminder_settings areminder_updates
Module arimanager … arimanager
Skipping module asterisk-cli (Nothing to convert)
Skipping module asteriskinfo (Nothing to convert)
Skipping module backup (Nothing to convert)
Skipping module blacklist (Nothing to convert)
Module broadcast … broadcast_campaigns broadcast_campaign_groups broadcast_settings broadcast_groups broadcast_callees broadcast_log
Skipping module builtin (Nothing to convert)
Skipping module bulkhandler (Nothing to convert)
Skipping module calendar (Nothing to convert)
Skipping module callaccounting (Nothing to convert)
Module callback … callback
Module callerid … callerid_entries
Skipping module callforward (Nothing to convert)
Module calllimit … calllimit calllimit_usage
Module callrecording … callrecording callrecording_module
Skipping module callwaiting (Nothing to convert)
Skipping module campon (Nothing to convert)
Skipping module cdr (Nothing to convert)
Skipping module cel (Nothing to convert)
Skipping module certman (Nothing to convert)
Module cidlookup … cidlookup cidlookup_incoming
Module conferences … meetme
Skipping module conferencespro (Nothing to convert)
Skipping module configedit (Nothing to convert)
Module contactmanager … contactmanager_groups contactmanager_group_entries contactmanager_entry_numbers contactmanager_entry_images contactmanager_entry_userman_images contactmanager_entry_xmpps contactmanager_entry_emails contactmanager_entry_websites
Module core … sip dahdi featurecodes incoming outbound_routes freepbx_settings trunks users devices dahdichandids globals language_incoming languages outbound_route_patterns outbound_route_sequence outbound_route_trunks outroutemsg trunk_dialpatterns trunks
Module cos … kvstore
Module customappsreg … custom_destinations custom_extensions
Module cxpanel … cxpanel_server cxpanel_voicemail_agent cxpanel_recording_agent cxpanel_email cxpanel_phone_number cxpanel_users cxpanel_queues cxpanel_conference_rooms cxpanel_managed_items
Skipping module dahdiconfig (Nothing to convert)
Skipping module dashboard (Nothing to convert)
Module daynight … daynight
Skipping module dictate (Nothing to convert)
Skipping module digiumaddoninstaller (Nothing to convert)
Skipping module digium_phones (Nothing to convert)
Module directory … directory_details directory_entries
Module disa … disa
Skipping module donotdisturb (Nothing to convert)
Skipping module dundicheck (Nothing to convert)
Skipping module endpoint (Nothing to convert)
Module extensionroutes … extensionroutes
Skipping module extensionsettings (Nothing to convert)
Module fax … fax_details fax_incoming fax_users
Module faxpro … fax_store faxpro_hook_core
Skipping module featurecodeadmin (Nothing to convert)
Module findmefollow … findmefollow
Skipping module firewall (Nothing to convert)
Skipping module framework (Nothing to convert)
Skipping module freepbx_ha (Nothing to convert)
Skipping module fw_langpacks (Nothing to convert)
Module hotelwakeup … hotelwakeup hotelwakeup_calls
Module iaxsettings … iaxsettings
Skipping module infoservices (Nothing to convert)
Skipping module irc (Nothing to convert)
Module ivr … ivr_details ivr_entries
Skipping module languages (Nothing to convert)
Module logfiles … logfile_settings logfile_logfiles
Skipping module manager (Nothing to convert)
Module miscapps … miscapps
Module miscdests … miscdests
Module music … music
Module outroutemsg … outroutemsg
Module paging … paging_groups paging_config
Module pagingpro … pagingpro pagingpro_core_routing pagingpro_scheduler_events pagingpro_scheduler_range pagingpro_scheduler_exclusions pagingpro_scheduler_crons
Module parking … parkplus
Module parkpro … parkplus_device parkplus_announce
Skipping module pbdirectory (Nothing to convert)
Skipping module phonebook (Nothing to convert)
Skipping module phpinfo (Nothing to convert)
Module pinsets … pinsets pinset_usage
Skipping module pinsetspro (Nothing to convert)
Skipping module pm2 (Nothing to convert)
Skipping module pms (Nothing to convert)
Skipping module presencestate (Nothing to convert)
Skipping module printextensions (Nothing to convert)
Skipping module queuemetrics (Nothing to convert)
Module queueprio … queueprio
Module queues … queues_details queues_config
Skipping module queuestats (Nothing to convert)
Module qxact_reports … qxact_calls qxact_agent_calls qxact_agent_actions qxact_system_events qxact_reports
Module recordings … recordings
Module recording_report … kvstore recording_report
Module restapi … restapi_general restapi_log_event_details restapi_log_events restapi_token_details restapi_tokens restapi_token_user_mapping
Module restapps … restapps_settings restapps_stats
Module ringgroups … ringgroups
Module sangomacrm … kvstore sangomarcrm_suitecrm_users
Module setcid … setcid
Module sipsettings … sipsettings
Module sipstation … module_xml
Module sms … sms_messages sms_routing sms_media
Skipping module soundlang (Nothing to convert)
Skipping module speeddial (Nothing to convert)
Module superfecta … oauth_log oauth_consumer_registry oauth_consumer_token oauth_server_registry oauth_server_nonce oauth_server_token superfectaconfig superfectacache superfecta_to_incoming superfecta_mf superfecta_mf_child
Module sysadmin … sysadmin_options
Module timeconditions … timeconditions timegroups_groups timegroups_details
Skipping module tts (Nothing to convert)
Skipping module ttsengines (Nothing to convert)
Skipping module ucp (Nothing to convert)
Module userman … userman_users userman_users_settings userman_groups userman_groups_settings
Skipping module vega (Nothing to convert)
Module vmblast … vmblast vmblast_groups
Module vmnotify … vmnotify vmnotify_notifications vmnotify_events
Module voicemail … voicemail_admin
Skipping module voicemail_report (Nothing to convert)
Module vqplus … virtual_queue_config vqplus_queue_config vqplus_qrule_config vqplus_qrule_detail vqplus_callback_config vqplus_callback_calls vqplus_callback_log
Skipping module weakpasswords (Nothing to convert)
Module webcallback … webcallback
Skipping module webrtc (Nothing to convert)
Module xmpp … prosody xmpp_users xmpp_options
Skipping module zulu (Nothing to convert)
Attempting to discover users email addresses … (130 users, 0 with email) Complete!
Export complete!
This backup is currently being processed by the conversion servers, and will
be automatically downloaded and installed on the NEW machine when completed.

Please Note! All SIP and IAX trunks on the NEW machine will be set to DISABLED,
and you will need to manually re-enable them. This is to avoid accidental
outages if the NEW machine registers to a peer unexpectedly.
Cleaning up…Done!

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@wibble wrote:

I have two freepbx installations, one as a DR failover for the primary… When it comes to updating the secondary there’s zero problems, as it’s normally passive anyway, it’s just a case of updating. However, I cannot seem to successfully block the channel to the primary, while keeping it connected to the Internet to be able to update it. Sure, I can take it off the network completely, but then I can’t update it!.
I’ve tried blocking channels in the firewall, but it still seems to respond to 5060 UDP. How can I take down the service so that the supplier will failover to the secondary, while keeping the access to the Internet for updates? How do other people do this?
TIA, Steve

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@mikeaph wrote:

I have multiple FreePBX’s running. i randomly received an email overnight from one of them with the subject “Call recordings backed up” and the body of the email reading:

We have just performed a monthly backup of your call recordings.

If you wish to save your call recordings, please log in and download your call recording archives.

My problem is, these machines have been running for a while and this is the first time I’ve seen this message. I checked my backups and all seem normal.

Is there a separate backup location for call recordings? Where can I find where this email and backup are setup so I can see what it is doing and how to change it?

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@mcisar wrote:

Trying to de-ugly something for a client. Essentially they have a gate the status of which can be polled for open/closed status with a PHP script. Somewhere along the line someone morphed that into a page on their webserver that they can keep refreshing in a browser to see if the gate is open or closed. I’m sure it seemed like a good idea at the time but in practice not optimal because either the browser gets closed, buried, etc.

What I’d like to do is dump that status into a custom devstate on their FreePBX box so we can just set up a BLF button to monitor it on a couple of phones.

What is the easiest way to update a custom devstate on FreePBX by an external script (or commandline). Obviously PHP would be cleanest end result because I can just merge that into the polling script they already have… but I can easily pass off to another script if PHP isn’t the optimal choice for talking to Asterisk. I just don’t want to churn out 100 lines of code to do something that could be handled with 2 lines of shell script and a cron job.

Thanks!

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@ChrisMaverley wrote:

Hello Folks,

Our freePBX server was hacked this weekend and used to run up a big bill of international calls. I was alerted by our VoIP provider. I am puzzled because the box was protected by a hardware firewall, a Juniper J series router. It wouldn’t have been possible to SSH to the box or register an extension. However, port 80 was open to the outside world, I didn’t have default username and password set for the GUI.

There is no record of the calls in the GUI but I can see them in /var/log/asterisk/full

We nipped this in the bud thanks to precautions our VoIP provider have in place but as we are a small enough company this could have done a lot of financial damage.

Can this type of attack take place on Port80 and if so how?
I have changed the default ports and the GUI is no longer reachable from the public internet.

I would appreciate any tips or links to relevant articles.

Regards,
Chris.

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@PitzKey wrote:

Hi guys,

FreePBX 14 Asterisk 13.22.0

The GUI shows System Last Rebooted:
2 months, 3 weeks, 16 hours, 44 minutes, 31 seconds, ago

[root@freepbx ~]# asterisk -rx"core show uptime"
System uptime: 1 hour, 12 minutes, 41 seconds
Last reload: 1 hour, 12 minutes, 41 seconds

I see in /var/log/asterisk/full

[2019-09-03 09:10:09] Asterisk 13.22.0 built by mockbuild @ jenkins7 on a x86_64 running Linux on 2018-07-25 22:30:39 UTC
[2019-09-03 09:10:12] VERBOSE[40768] asterisk.c: Asterisk Ready.

I see in /var/log/asterisk/freepbx.log

[2019-Sep-03 09:10:10] [INFO] (functions.inc/callback-daemon.php:3) - Queue callback daemon started

But when running

[root@freepbx ~]# fwconsole pm2 --list
+--------------+-------+--------+----------+-------------------+-----+---------+
| Process Name | PID   | Status | Restarts | Uptime            | CPU | Mem     |
+--------------+-------+--------+----------+-------------------+-----+---------+
| ucp          | 40798 | online | 1144     | 2 months, 21 days | 0%  | 63.61MB |
| qcallback    | 40845 | online | 124      | 2 months, 21 days | 0%  | 33.64MB |

So seems like Asterisk was restarted and from the log it seems like Queue callback daemon also stopped, but from pm2 it does not seem so.

ls -l /tmp | grep core.

Does not return anything.

What else can I look for to find what caused Asterisk to be restarted?

Thank you

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@ccartwright wrote:

Hello
I cloned our production system. While working on the system we decided to make this a second live system and the MAC address of the vm was modified so it was no longer the same MAC. It is currently on a separate subnet and isolated from the production system. I would like for this system to become a separate production system, however, it has the same deployment ID of the system from which it was cloned. How to I get a new deployment ID for this second system without impacting / disabling the first production system?
Thank you for your assistance.
Kind regards,
Carol

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@jtomelevage wrote:

What, if anything, is being worked on to accommodate the American Stop Robocalls Act? Specifically, what can FreePBX/Asterisk users do to prevent their legitimate phone numbers from being marked as SPAM / Robocalls?

John

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@ajs wrote:

This is a problem I am having with the non-pro parking module. I don’t know if this was caused by upgrading to FreePBX 15. Parking a call works normally, but I can’t pick up a parked call. This appears to be because the macro parked-call is passed the extension the call is parked at, but not the parking lot. As a result, /var/lib/asterisk/agi-bin/parkfetch.agi is called as

/var/lib/asterisk/agi-bin/parkfetch.agi <parking lot number>

and not as

/var/lib/asterisk/agi-bin/parkfetch.agi <parking lot number> <parking lot>

It executes "parking show ", which does not retrieve the parked call and tells you there is no call parked at the extension. If it executes “parking show default”, then the call is unparked as expected. I have verified this by altering /var/lib/asterisk/agi-bin/parkfetch.agi

[root@freepbx agi-bin]# diff parkfetch.agi.orig parkfetch.agi
16c16,18
< $r = $astman->send_request(‘Command’,array(‘Command’=>"parking show ".$argv[2]));

---

> //$r = $astman->send_request(‘Command’,array(‘Command’=>"parking show ".$argv[2]));
> $r = $astman->send_request(‘Command’,array(‘Command’=>“parking show default”));

Any idea why the parking lot’s name isn’t being passed? ${CHANNEL(parkinglot)} is the second argument of the parked-call macro but I guess it doesn’t contain anything.

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@milolethbridge wrote:

Hi all,

Running into a strange issue with one of our sites.

Their time condition does not want to stay in a ‘no override’ state.

The time condition is set to match from 0800 - 1745 Mon-Fri and divert to ring group, however this morning, calls did not come through and the time condition shows ‘temporary unmatched’.

When dialling the feature code to override, the time condition will then show ‘no override’ however as soon as a call hits, it is diverted to the unmatched destination and switches back to ‘temporary unmatched’.

Not sure this will show in CDR’s, from where can I pull logs to provide more info?

Conversely, if I’m doing anything stupid, please let me know.

Thanks in advance.

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@JessicaRabbit wrote:

Is the below still correct for PJSIP?

• Created by [Tony Lewis], last modified on [08 Feb , 2016]

PJSIP is the emerging SIP technology in Asterisk. It provides additional functionality and features not present in the legacy chan_sip and over time it will become the predominant SIP technology. However, because of it’s youth and more extensive feature set, customers are likely to encounter more bugs and issues. We encourage customer who either need these features or want to help progress and mature the technology to run with PJSIP, even in production, with the acknowledgement that there will be more issues. The Asterisk and FreePBX (Sangoma) Development teams are fully behind PJSIP and will try to address all bugs and issues that arise from it. However, chan_sip still remains the mature SIP channel that should be used where stability is the most critical factor and tolerance for early adoption of new technologies can’t be tolerated.

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@mikeaph wrote:

Experiencing a strange issue that is completely random.

I have FreePBX running on a PC, on premise. I have no audio issues when I call out or when people call in. However, on a regular basis, when I, or any of our users, call out to someone’s cell phone and get their voicemail, the voicemail does not recognize that anyone is speaking. One of two things happens depending on the number we are calling that we are calling (Verizon, T-mobile, Spint, etc.); the call hangs up after 30 seconds or so because it thinks no one is there, or, the voicemail system will cut you off while you are leaving a message telling you that no message has been recoded because we are not speaking. I have also noticed that when this happens, if I hit # or any other key, the voicemail system does not recognize the key press. So either the call is disconnected or we hang-up and then call the number right back and this time it will work with no problems. The voicemail will answer and I am able to leave a message.

I have been able to duplicate this issue on two other systems. Both of those systems are FreePBX running in the cloud on a Vultr. And the exact same scenario occurs.

I can literally call a phone number, get the persons voicemail and leave a message with no problem, or call the same number and experience the issue that I described above. Then, call the number again and it will work with no issue.

I have no clue if this is a FreePBX issue, firewall issue, trunking issue (using Flowroute on all)…

Considering that they are different instances of FreePBX, different hardware and different firewalls, I’m not sure what the problem could be.

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@bajramia wrote:

Hi All,
I have a customer who i have setup a warm spare and backing up nightly we are having issue with ghost voicemails, if en extension receive a voicemail and haven’t read when back up runs that voicemail moves to the other server and phones receiving alerts from standby you have a new voicemail when customer checks it says you dont have no voicemail.

Thank you

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@NorColorNorName wrote:

Hello,

i was asking myself if it possible to call someone with my asterisk server without any softphone. Do a dialplan / commands or whatever allow me to do that ?

My goal is to execute many calls with the server and the first one who answer will be transfer to one extension and all other will be hangup.

Any advices will help !
Regards,
NCNN.

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@BlazeStudios wrote:

[split from unrelated thread - mod]

OK so now there is a tweet out there telling Foritgate users to check their stuff because of FreePBX servers getting pwned due to Fortigate exploits. However, so far the only publicly reported issue of this (this one) wasn’t due to a Fortigate exploit since the firewall had a rule allowing access to the GUI. Last time I check, putting a firewall rule in is not an exploit.

So in this thread we have one person saying “A few reported it to me” and in the tweet it’s “several” so can we quantify that? Is it more than 2 but less than 12? More than that? Has enough data been provided that shows this is a Fortigate exploit or is it cases like this one?They are using Fortigate but then they had a rule to allow access.

Since I can’t see how logically this can be a Foritgate exploit and only impact FreePBX systems, are there reports from Fortigate about any of these exploits? Firmware updates with a patch? Are there any suggested support steps users can take?

While it is totally possible others were pwned by Fortigate exploits, clearly the OP wasn’t because the firewall allowed the access and thus this particular instance is pointing more to exploits in Apache vs the firewall.

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@rnrstar wrote:

I’m running FreePBX 14 with Asterisk 16 and we use AMI to initiate calls. Essentially what we do is to ring the extension and when the user picks up we then dial the external number. The problem we are running into is if the user is on DND, the call goes directly to voicemail which looks like a pickup and so the outbound number is dialed. This results in a voicemail being created.

What we would like to do is to turn off DND on the extension before we send the call. I can’t find any documentation on how to turn off DND via the AMI API.

Any suggestions?

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@BGM wrote:

My polycom 601’s won’t auto-answer pages, so I’m playing with their config settings, but I have to constantly change the time on the scheduler and wait for it.

Is there a way to test the page-group without having to set the scheduler? If I just dial the page extension, it connects the calling extension and I don’t want to do that. I want the system to send the announcement to the destination.

I am trying to make bells play over the phone. Once I get it working then I can set the schedule.

It used to be working before (I have a thread here), but now it isn’t and I can’t figure out why. The phone always rings instead of answering.

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@dotcom wrote:

Hello,

I’m unable to block malicious SIP INVITES to my box.
Via sngrep, I see the following packet coming in:

INVITE sip:00441519470451@mypbxipaddress SIP/2.0
Via: SIP/2.0/UDP 77.247.110.65:62393;branch=z9hG4bK1998065274
Max-Forwards: 70
From: sip:16@mypbxipaddress;tag=432918223
To: sip:00441519470451@mypbxipaddress
Call-ID: 2058906050-1389826825-1470979558
CSeq: 1 INVITE
Contact: sip:16@77.247.110.65:62393
Content-Type: application/sdp
Content-Length: 206

I already tried adding the line below to my iptables script, but no luck:
$IPT -A INPUT -i eth0 -p udp -m udp --dport 5580 -m string --string “00441519470451” --algo bm --to 65535 -j DROP

Thanks for your urgent help!

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