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Locked Out of FreePBX 15

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@side1track1 wrote:

I’m currently locked out of FreePBX 15 via SSH and the GUI - I’ve tried multiple computers on the network accessing the IP using a browser and SSH via Putty (connection timed out) and I believe it is because fail2ban and the firewall were both enabled. Phones are still currently registered and incoming/outgoing calls work; however, I’m at a loss for how to now access. Does the fail2ban delete after a certain amount of time? I can still ping the IP via a command prompt. I’m going to try accessing via a VM; however, does anyone else have any ideas on how to access? Thank you!

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GUI stops responding if left on dashboard - any browser

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@Chuckak wrote:

Current PBX Version:14.0.13.4
Current System Version:12.7.6-1904-1.sng7

If I leave the browser, Chrome, Firefox or Edge on the FreePBX dashboard for more than a few hours the Network widget quits updating. Then awhile later I cannot navigate to any tabs. I cannot refresh the browser.
Closing and reopening sometimes works but usually not. Can’t find webpage. If I change browsers it will work for awhile again. If I use the IP address instead of FQDN it will work until it times out again.

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Asterisk: a targeted VOIPspionage campaign - update PBX to patch the vulnerability

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@moussa854 wrote:

FYI, I would like to share these two articles hoping to encourage the community in keeping their system up to date.

In summary, attackers could use a vulnerability to access the FreePBX, steal data and install crypto mining scripts. It seems that the security hole has been patched, so it is recommended not to put off updates for long.

Source: https://www.virusbulletin.com/conference/vb2019/abstracts/asterisk-targeted-voipspionage-campaign/

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Registration to SIP Trunk fail after network device restart

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@va2xjm wrote:

Hi everyone,

For a while, from time to time, the PBX was not able to re-register to external trunks until the PBX was rebooted.

Today I have been able to make tests and find out that when the router between Internet and the PBX is restarted, the PBX is not able to re-register. Whatever is tried, even restarting Asterisk do not work. The PBX must be rebooted to have it re-register.

The extensions works from both LAN and WAN.

Re-registration goes on every 60seconds without success until the PBX is rebooted. The same has been tested with 2 providers.

[2019-10-05 21:27:32] WARNING[1105]: res_pjsip_outbound_registration.c:796 schedule_retry: No response received from ‘sip:montreal7.voip.ms:5060’ on registration attempt to ‘sip:@montreal7.voip.ms:5060’, retrying in ‘60’
[2019-10-05 21:27:34] WARNING[1105]: res_pjsip_outbound_registration.c:796 schedule_retry: No response received from ‘sip:montreal7.voip.ms:5060’ on registration attempt to 'sip:
@montreal7.voip.ms:5060’, retrying in ‘60’
[2019-10-05 21:27:37] WARNING[1105]: res_pjsip_outbound_registration.c:796 schedule_retry: No response received from ‘sip:montreal7.voip.ms:5060’ on registration attempt to 'sip:***@montreal7.voip.ms:5060’, retrying in ‘60’
pbx
CLI> pjsip show registrations

<Registration/ServerURI…> <Auth…> <Status…>

voipms_***/sip:montreal7.voip.ms:5060 voipms_*** Rejected
voipms_***/sip:montreal7.voip.ms:5060 voipms_*** Rejected
voipms_***/sip:montreal7.voip.ms:5060 voipms_*** Rejected

Objects found: 3

Anyone have a solution that could fix this ? This is really a PITA… I actually have Zabbix monitoring “pjsip show registration” output and send alerts when trunks are down, but rebooting PBX should not be a solution…

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SQL Table recover

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@xentity wrote:

hi there,

may it be possible that someone please dumps his definition of the sql tables:

  • ‘asterisk.incoming’
  • ‘asterisk.ivr_details’
  • ‘asterisk.kvstore_FreePBX_modules_Conferences’

Had some file system issues and the backup was also effected (transferring the backup to another system was an open point… shame on me).
However, I was able to fetch all config files (asterisk is working) and most of the DB tables. But the tables mentioned above were not able to recover.

The following commands

mysqldump --no-data -u root asterisk incoming
mysqldump --no-data -u root asterisk ivr_details
mysqldump --no-data -u root asterisk kvstore_FreePBX_modules_Conferences

should do the trick on a most recent system.

The content of incoming should be restorable by the content of extensions_additional.conf, is that right? I didn’t use IVR nor Conferences, therefore the content of those should be avoidable, right? But the tables are needed for the module updates.

Thx a lot.

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SRV for domain name - cannot connect remote devices

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@maxyca wrote:

Hello all!
I use PJSIP and I have created 2 SRV for my domain name like this:
_sip._tls.my-domain.com. priority = 0; priority = 0; weight = 5; port = 7770
_sip._tcp.my-domain.com. priority = 0; weight = 5; port = 443

However, unfortunately, remote devices doesn’t connect to Asterisk. I also use a non-standard port for PJSIP.

What am I doing wrong?

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Edge Modules?

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@xrobau wrote:

I just noticed that there’s some bugs with the current ‘stable’ track in 14, and ran a fwconsole ma --edge upgradeall - why not live dangerously, right?!?

But there appears to be some modules that have stalled in edge for a LONG time - for example, firewall hasn’t been released with the changes from 6 months ago, and core…

Updating Hooks...Done
Upgrading module 'core' from 14.0.25.4 to 14.0.28.19
Downloading module 'core'
Processing core

Historically, this was meant to happen the first day of the week, but it looks like it’s slipped through the cracks. Any news on when they’re going to be published?

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Full HDD FreePBX

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@Pletev wrote:

Hello, The system reports to me:

Storage space is getting high on the following drives of your system:
/dev/sda2 is 76% full

Unfortunately I don’t have access to SSH consoles. Can I delete old backups from the web interface?
Thank you for the advice.

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Voicemail to Email stopped working

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@bazzacad wrote:

For some reason our Voicemail to Email stopped working. I’m not sure if it’s a PBX or network issue. Any suggestions on how to trouble shot/debug the issue?
Current PBX Version: 14.0.13.4
Current System Version: 12.7.6-1904-1.sng7

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Paging calls don't hangup

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@averythomas wrote:

I am using the Paging and Intercom module 14.0.12 and occasionally when a call is made to the paging group the channels are still active and I have to kill the calls manually. That is a problem because, any emergency pages do not go through.

Is there a way to make it so new paging calls are automatically prioritized even if a channel is stuck or maybe a channel timeout that will kill the channels?

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Integrating Freepbx 13 with Samsung Officesrv 7200

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@joebates wrote:

Hi, I am new to the forum. This is my first post. I have been installing and maintaining Freepbx systems for the last 4 years. I have been asked to integrate Freepbx with a Samsung Officeserv 7200. I have limited knowledge of the Samsung Officeserv. Is there anyone out there who has managed to integrate both systems successfully. Ideally I need to set up a trunk between the two systems to make outbound SIP calls and allow the Samsung extensions to call the Freepbx extensions (and vice versa). The idea is to phase out the Samsung Officeserv over the next year or so (the system is only 5 years old, so a difficult sell to dump it and start again with a new Freepbx system). Any advice much appreciated.

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Outbound call - cannot head subscriber b for several seconds

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@dux wrote:

Hello, everyone.
When I make calls from my softphone, the called party cannot hear me for a couple of seconds. What I have found in logs is that I hear an extra ring after channels join each other in a bridge, and that confuses the caller. How can I correct this?

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Server refused to allocate pty

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@jtvdw wrote:

Hi,

I’m running asterisk on a centos6.6 in a proxmox virtual inviroment, I moved the container from 1 node to anotehr and now I’m getting this error: “server refused to allocate pty”, tried a couple of things but wasn’t successful.

Also, I’ve yum updated to 6.10 and now when I login if I click on a menu item, I get logged out.

Any help would be appreciated.

Thanks.

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How to speed up playback of system prompts in voicemail

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@ccartwright wrote:

Hello
Our users are reporting that listening to voicemail messages is an extremely slow process for them.
We have had the following voicemail settings enabled: Say CID, Say Duration, Envelope Playback.

Is there a way to speed up the playback of the system prompts to our users. We can turn off these settings but for those who wish to still have them turned on, can we speed them up?
Thank you,

Carol

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+ sign strip in freepbx outbound routes

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@promise2k wrote:

When a user dials from a Zoiper Softphone on their mobile when they dial a number ‘+971aaaaaaaaa’ i am getting below error.

‘+971aaaaaaaaa’ rejected because extension not found in context ‘from-internal’.

is there a way in freepbx i can strip the + sign and replace that with 00

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Call Transfer On Hold Music

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@v70ff wrote:

When we place someone on hold, the on hold music plays fine but we then transfer the call and the on hold music stops.

Is it possible to have on hold music to continue playing when a call is transferred until someone picks up the call?

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Is IAX2 still best trunk type for Internal Calling between FreePBX Systems? Specifically related to Encryption

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@jgiebler wrote:

Asking this question hoping for some good thoughts back.

Recently we went through an initiative to upgrade all of our FreePBX boxes to the latest supported versions and transfer our trunks to Encrypted PJSIP. We then also made sure that each extension was making encrypted connections to each server.

It recently dawned on me that IAX2 which is used to communicate between our servers (call one system from another) likely doesn’t have the same encryption level that we implemented every where else via PJSIP

What is your experience? Is IAX2 still the best way to connect internal FreePBX boxes together or should we figure out how to move these connections to PJSIP?

Thank you for your thoughts

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Help understanding where a call went to

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@davids1 wrote:

Hi,
I have built a simple dialplan to be able to do some ring back features with short and long BLF presses. It’s working for most people but there seems there are a small number of calls where, during the BLF transfer, the inbound call goes somewhere and sits on hold for minutes and my client can’t get the callback. I don’t think it’s my dial plan, but it is my first real dial plan so I am sure it could be enhanced to help these types of things happening in future. I am finding it difficult to find out what’s happening by the logs.

Can anyone confirm my notes below or help me understand where the call might be going, and how, perhaps, the same users are having the same issues.

Background - 150 user PBXact in the cloud with 7 sites (about 20 phones (S500’s and S505’s) per site)
More than one site is having the issue, but its only on very few calls. Could it be a user issue, a faulty phone, a broadband issue at the time of the BLF transfer?

BLF buttons are setup to do Attended Transfer
Short Press BLF injects SP before the BLF Value
My dial plan looks for calls with SP in the first two characters, then sends the call to the Extension (minus the first 2 characters) with a ring back time that is less than the extension voicemail timeout. I also inject some additional text so the Ring back shows the user name of who was unavailable.

First my dial plan
[from-internal-custom]
exten => _SPX.,1,Set(origext=${CALLERID(num)})
same => n,Set(origextname=${DB(AMPUSER/${EXTEN:2}/cidname)})
same => n,Set(timeoutd=20) ; set timeout in seconds
same => n,Dial(Local/${EXTEN:2}@from-internal,${timeoutd},m)
same => n,Set(CALLERID(name)=RB:${origextname}-${CALLERID(name)})
same => n,Dial(Local/${EXTEN:2}@from-internal&Local/${origext}@from-internal,90,m); this calls both phones back but finishes at voicemail of destination
same => n,Hangup()`

Here is a list of the logs showing an inbound call yesterday - the full logs will be uploaded after this first post.
Call in to this number 021111111111
[2019-10-09 12:00:29] VERBOSE[11278][C-00002df2] pbx.c: Executing [02111111111@from-trunk:1] Set("IAX2/sbc3-3141", "__DIRECTION=INBOUND") in new stack

from this number 02020202020
[2019-10-09 12:00:29] VERBOSE[11278][C-00002df2] pbx.c: Executing [in@sub-record-check:3] ExecIf("IAX2/sbc3-3141", "11?Set(FROMEXTEN=02020202020)") in new stack

picked up by extension 805 after 5 seconds
[2019-10-09 12:00:34] VERBOSE[11324][C-00002df2] app_dial.c: SIP/805-00009163 answered Local/805@from-queue-00038a83;2

The recorded audio of this call reveals that they speak for about 20 seconds and then 805 puts the call on hold and the call sits listening to hold music for about 5 minutes before they hang up.

It looks like 805 presses the transfer button which puts the call on hold.
[2019-10-09 12:00:57] VERBOSE[11324][C-00002df2] res_musiconhold.c: Started music on hold, class ‘clientname’, on channel ‘Local/805@from-queue-00038a83;2’
[2019-10-09 12:00:57] VERBOSE[12601] chan_sip.c: Extension Changed 805[ext-local] new state Hold for Notify User

Then 805 presses the BLF of 812 a second later (this is where my dial plan code comes in - Short Press)
(I know they don’t need to press the Transfer button first but I am unsure if it makes a difference? or could this be the cause here? We have tested the code and pressing the transfer button before the short press option doesn’t make a difference - please correct me if I am wrong)
(one thing I can see might be an issue below is [SP812@from-internal:1]… should I try to get my code to also remove SP in it so it would look like [812@from-internal:1])

[2019-10-09 12:00:58] VERBOSE[11611][C-00002df4] pbx.c: Executing [SP812@from-internal:1] Set("SIP/805-00009166", "origext=805") in new stack
[2019-10-09 12:00:58] VERBOSE[11611][C-00002df4] pbx.c: Executing [SP812@from-internal:2] Set("SIP/805-00009166", "origextname=812 User Name - Dept") in new stack
[2019-10-09 12:00:58] VERBOSE[11611][C-00002df4] pbx.c: Executing [SP812@from-internal:3] Set("SIP/805-00009166", "timeoutd=20") in new stack
[2019-10-09 12:00:58] VERBOSE[11611][C-00002df4] pbx.c: Executing [SP812@from-internal:4] Dial("SIP/805-00009166", "Local/812@from-internal,20,m") in new stack

by the looks of the code - and this is where I need help…
It looks like 812 answered the call from 805 a few seconds later
[2019-10-09 12:01:00] VERBOSE[11612][C-00002df4] app_dial.c: SIP/812-00009167 answered Local/812@from-internal-00038a8e;2
[2019-10-09 12:01:00] VERBOSE[11611][C-00002df4] app_dial.c: Local/812@from-internal-00038a8e;1 answered SIP/805-00009166
[2019-10-09 12:01:00] VERBOSE[12601] chan_sip.c: Extension Changed 812[ext-local] new state InUse for Notify User

For some reason 812 didn’t get connected to the call, the call didn’t follow the ring back code and sat on hold until they hung up.

I can see 10 seconds later these logs showing 812 and 805 going idle and hangup. This is where I am stuck, I cant work out where the call sent to, and finally is there anything i can enhance my dialplan code with to stop this happening?

[2019-10-09 12:01:10] VERBOSE[12601] chan_sip.c: Extension Changed 812[ext-local] new state Idle for Notify User 811 
[2019-10-09 12:01:10] VERBOSE[11611][C-00002df4] bridge_channel.c: Channel SIP/805-00009166 left 'simple_bridge' basic-bridge <a9d6a425-6ab7-481c-8ba3-5486ec1b0510>
[2019-10-09 12:01:10] VERBOSE[12601] chan_sip.c: Extension Changed 812[ext-local] new state Idle for Notify User 831 
[2019-10-09 12:01:10] VERBOSE[12601] chan_sip.c: Extension Changed 812[ext-local] new state Idle for Notify User 830 
[2019-10-09 12:01:10] VERBOSE[11611][C-00002df4] pbx.c: Spawn extension (from-internal, SP812, 4) exited non-zero on 'SIP/805-00009166'
[2019-10-09 12:01:10] VERBOSE[11611][C-00002df4] pbx.c: Executing [h@from-internal:1] Macro("SIP/805-00009166", "hangupcall") in new stack

I shall paste the full call log in the next few minutes.
Thanks

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ITSP trunk sharing across active/active servers

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@michthom wrote:

Hi folks. This might fall into the “don’t be daft” category, but here goes.
A small enterprise has grown and has multiple incoming trunks segregating call types through a highly-resilient ITSP. But the FreePBX box is a single site/single server risk. The desire is this - configure a second server on another site, and have both register to all the ITSP trunks simultaneously, but have a preference for each trunk to one or other server?
That way if either server goes down the remaining system takes calls from the trunks that initially preferred the dead server.
E.g. Routing calls into the PBXs in a similar way to sequential forking calls out to client devices, but on the incoming side of the FreePBX?
I’ve done lots of googling but to no avail so either this is so easy nobody has written it up or so difficult/stupid that it’s not worth attempting?

The low tech way would be to disable half the trunks on one server and the other half on the other server, and manually bring them all online on the remaining server when the other failed. But that’s slow to respond and runs a risk when the failed server comes back up that we’re back to two competing registrations, rather than cooperating nicely.

There are commercial solutions out there, but those I’ve seen concentrate on local clustering, so leave the single site risk in place, or cost more than the small budget available.

What am I missing?

Cheers
Mike Thomson

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One way audio using wireguard VPN (tried advice from other posts without luck PLZ READ)

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@Tylersprice wrote:

I have freepbx 14 set up and working internally on my LAN, I dont have pbx internet facing so i VPN into my network to make an internal call. Everything works fine as long as im on my LAN and no VPN, The moment i VPN into my LAN I cant get audio to my softphone. I have some logs of ‘asterisk -rvvvv’ of a succesful LAN call and the one way audio VPN call and it seems to me that RTP is setting the wrong IP.
My wireguard host has ip of 10.0.0.201, the wireguard interface/server is 10.100.100.1 and my phone is 10.100.100.2. When i make a call thru VPN first RTP says 10.100.100.2:xxxx than a few lines down it changes it to 10.0.0.201. I think this is where the problem is and IDK how to fix it. any insight would be helpful.

Ive added my external IP to network settings and added 10.0.0.0/255.0.0.0 under local networks (and rebooted), Intrusion detection and firewall are off for now until i resolve this issue, all other nat settings are stock. I changed chan_sip to 5060 but other than that its fairly stock. Id like to be able to vpn ini instead of having it initernet facing. Ive seen alot of instructions regarding openvpn or other protocols that work on level 2 but wireguard is level 3, could that be causing the issue? Or is there an iptables rule i need to implement on my wireguard host?
Thanks!

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