@jerryriggin wrote:

We upgraded from freePBX12 to freePBX 13.0.54 and it seems "Voicemail" is no longer an option "no answer" destination for a ring group or an option for an IVR selection? Is this a bug? If not, what is the intended work-around?

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@lhoward96 wrote:

I'm trying to install the FreePBX iso on a KVM using a kickstart file to make it go from command to running PBX quickly. I'm using libvirt to create the new virtual machine, however I'm finding it hard to figure out how to actually do this.
I keep getting "▒" after I press install without the kick start file.

Would it be better to do CentOS 6.5 then just run the FreePBX install afterwards? And if so, does anyone have an example of a kickstart file for installing FreePBX on CentOS?

Thanks

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@gdisarro wrote:

Good morning,
yesterday I installed the last FreeBPX distro on my Raspberry and configured everything to work with WebRTC. I have two problems:

1. If the Raspberry is connected to Internet, WebRTC call from UCP panel works fine (the phone icon is green and browsers can connect to websocket server on the Raspberry). If the Raspberry is not connected to Internet, WebRTC call from UCP doesn't work (browsers can't connect to websocket server and the phone icon is yellow). In both cases the raspberry has a fixed IP address. Is there something I can check in the configuration?

2. When raspberry is connected to Internet, the only webrtc calls that work are those made directly from UCP panel. All others attempts (sipML, SipJS, JSsjp and so on) don't work. I need to embed webrtc call inside my existing website. Is there a working example not linked to the UCP?

Hope someone can help me!

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@ronin064 wrote:

Hi, I installed FreePBX 13.0.54, and now I'm trying to install free fax (full fax may be later..)
I tried to download Freefax from Sangoma site, then the license number... but I cannot figure aout ho to install in Freepbx and obtain a fax extension...
Is there some step-by-step howto I can follow?
How can I resolve.
Thanks for You attenzion.

Giulio

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@avayax wrote:

FreePBX 10.13.66-1.
How do I start prosody if it is not running?

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@bwaynef wrote:

So, as I mentioned in this post I'd previously followed the instructions I provided a link to (in that thread) and got a working version of freepbx12. The second time I tried I ended up posting here and you can see where that led (a fresh install of freepbx 13). That said, I used the PBX Upgrader to upgrade the previous install(12) to 13.x. It seemed to complete, but now asterisk isn't being started. Any tips?

TLDR: I upgraded a working Freepbx 12 installation to 13 and asterisk doesn't start.

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@shauberg wrote:

When using pjsip for a sip trunk I am having a issue with a pjsip trunk constantly showing offline due to the provider filtering on the "From" header in the options packets. All option packets are getting dropped in this case. Registration, incoming and outgoing calls work fine as long is not set to failover to another trunk, just the trunk constantly shows offline. A packet capture reveals the pjsip options packets always has:
From: <asterisk@[IP Address of transport]>
which is an internal IP address. The Contact header always has
Contact: <sip:asterisk@[external IP:port]>
which also appears to not be generated from the pjsip conf's. If I setup the trunk using chan_SIP the From header is:
From: <sip:[username]@[provider's Domain]>
and the Contact header shows
Contact: <sip:s@[sip:external IP:port]>
and I get 200 OK responses. Nothing in the pjsip conf's appear to be used to generate the value for the From or Contact header in the options packets, is there a was to set this value or disable qualify for PJSIP trunks to sort of mimic qualify=no behavior in chan_sip? Changing the qualify frequency to blank or 0 always reverts back to the default value.

Removing the qualify_frequency line for the trunk in pjsip.aor.conf disables sending option packets, but the trunk will always show offline and the config will be overwritten on reload. Of course this was just a test and would certainly cause nat timeout issues.

FPBX: 10.13.66-7
Asterisk: 13.7.0

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@mike_b wrote:

I have been using FreePBX for a while, and I am generally quite happy with it. One thing I would like to be able to is to highlight a number (for example I would like to just highlight the phone number I get with a voice message from Asterisk and be able to dial it from my PC). I looked through a lot of options by googling "asterisk click to dial", but most of these are for web browsers, and I am not sure if they would work with emails in Outlook, or if I could highlight any number and have it dialed. And I don't want to install and de-install dozens of add-ons into various browsers to test this.

Does anybody have an idea where I could find something like this, that works and is malware-free?

Thanks.

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@claloano wrote:

I have a trunk that carries the ported number ...

I wish you could see the outgoing ported number and not the number of logins

you can do something by setting FreePBX gui?

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@claloano wrote:

I wish based on the outbound routes I use my Display Name changed

Maybe I can do this with a macro?

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@florianahm wrote:

I want to create a phonebook with list of extenstion and then users on their hard phones to have that phonebook list in their phones. Is that possible need to do it manualy ?

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@GeekBoy wrote:

In just a base Asterisk setup one can originate a call by simply entering the follow, per example:

channel originate SIP/*number to dial*@+outbound context+ application Playback hello-world

In FreePBX this is not the case and returns an error & warning in the +outbound context+ part.

In fact the system is looking for a host name.

For example
channel originate SIP/19165551234@vonage-out application Playback hello-world

would return

WARNING[1110][C-00000012]: chan_sip.c:6058 create_addr: No such host: vonage-out

Thanks for any tips & hints!

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@cynjut wrote:

Lost a server to overheat (and the magic smoke got out). Replaced it in December with a new server and installed FreePBX 13.1. Everything works well with one exception. Wait, let me give you details first:

Current Asterisk Version: 13.1
DAHDI Version: 2.9.2 Echo Canceller: MG2 (this is the version that works with the FXS card).

I have 18 POTS phones connected to the server which is used to call out based on a database that we use to schedule calls. It's a call center, and we have two other servers (Version 1.6) running. In addition to the 18 POTS phones, we have six SIP Phones designated for outbound dialing.

So, we are getting one-way audio on the DAHDI phones. Before anyone jumps to any conclusions - No, it isn't a firewall problem or misconfigured router. I've been trying to get this working for a month and I spent an hour with my VOIP provider checking, and all of the traffic is making it from the server to them and back again. I originally suspected a firewall problem since I was having one of those on another server, but this server is connecting to the remote end cleanly.

I've already tried every possible combination of RTP settings - some will make the system not work at all, but none will help the incoming DAHDI phones hear the other end.

Now, here's the strange part. The one-way audio is only on inbound connections to the DAHDI phones. Inbound connections to the SIP phones work fine (hence my assertion that the firewall is probably not the culprit). The other reason I'm pretty sure it isn't the firewall is that the problem exists with local calls. I added the localhost and RTP ports to the firewall specifically (just in case it was something in there).

Example time:
- Extension 131 dials 1-800-ARMYNOW and gets through. Call gets recorded, life is good.
- Extension 131 dials 125 (DAHDI to SIP) and gets through. Life is good.
- Extension uses "click to dial" feature of system (using the custom "call file" script I posted here a few years ago). Call goes through and remote end (outside the firewall) hears the caller, but the local end is dead air.
- Extension 125 dials 131 (SIP to DAHDI). Call goes through and 131 cannot hear 125. Destination 125 hears everything 131 says.
- "Wake Up Call" is left for 131. Extension 131 rings and hears nothing.
- Extension 132 dials 131 (DAHDI to DAHDI). Extenion 132 hears everything, 131 is dead air.
- Extension 131 dials 132 (still DAHDI to DAHDI). Extension 131 hears everything, 132 is dead air.

So, to summarize - if a call is placed TO a DAHDI extension, the person answering that phone will not hear the other end, regardless of the source. A call placed FROM a DAHDI extension works perfectly every time.

The other two servers work flawlessly using the same basic configuration, network, firewall, etc. SIP phones work flawlessly.

I thought for a minute that it might be a context issue, but there's nothing that corroborates that. I've considered it might be a CODEC problem - I've got everything running using ULAW, which the VOIP provider likes. I haven't seen anything in the logs, even with all the debug output I can turn on and 15 "-v"s on the command line. The dump of the traffic at the Ethernet port shows nothing suspicious. The VOIP provider verified that the traffic looks the same (when calling to an external number) regardless of the source or destination.

I'm open to suggestions.

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@MashX wrote:

I'm starting a project to remotely manage users on a FreePBX system (hopefully). I need to be able to run code from a web server to add/edit/delete users on a remote pbx. Is this even possible? If so, where do I start? I have experience with JS and PHP. Thanks.

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@jasonmel wrote:

Hello,

I notice when on Reports -> Asterisk Info -> Peers, the page does not refresh like it used to.

On previous versions if I click on Peers again the page would reload, but in this new GUI it does not. There also was a refresh button on the bottom left of the Peers page which is not available on the newer interface. Currently to see if the user has registered or dropped I must reload the browser which takes me to the Asterisk Info -> Summary and then I click on Peers again.

I have reproduced this on Firefox, Chrome and Edge but I'm unsure if it's a bug, or my Windows 10. Anyone else see this?

Thank you

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@reraikes wrote:

Re: http://issues.freepbx.org/browse/FREEPBX-11503

Works fine on the distro. This is a user misconfiguration. Please bring this up in the forums.

Did you actually try to build a FreePBX 13 + Asterisk 13 system on Debian?

I've made no changes to my install scripts/procedure in many weeks, but a recent update to FreePBX 13 or Debian has resulted in FreePBX 13 reporting "Can Not Connect to Asterisk". During FreePBX 13 installation, the following is reported, even though Asterisk was successfully installed and is running:

STARTING ASTERISK
Asterisk Started
Assuming you are Database Root
Checking if SELinux is enabled...Its not (good)!
No /etc/asterisk/asterisk.conf file detected. Installing...Writing /etc/asterisk/asterisk.conf...Done
Checking if Asterisk is running and we can talk to it as the 'asterisk' user...Done
Preliminary checks done. Starting FreePBX Installation
Checking if this is a new install...Yes (No /etc/amportal.conf file detected)
Database Root installation checking credentials and permissions..Connected!
Empty asterisk Database going to populate it
Empty asteriskcdrdb Database going to populate it
Initializing FreePBX Settings
.
.
.
Installing framework...
No directory /var/www/html/admin/modules/framework/amp_conf/htdocs, install script not needed
Generating CSS...Done
Module framework successfully installed
Updating Hooks...Done
Done
Generating default configurations...
Exception: Unable to connect to Asterisk Manager from /var/lib/asterisk/bin/retrieve_conf, aborting in file /var/lib/asterisk/bin/retrieve_conf on line 13
Stack trace:
1. Exception->() /var/lib/asterisk/bin/retrieve_conf:13
Finished generating default configurations
Trusting FreePBX...Trusted
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions...
.
.
.
Module voicemail successfully installed
Updating Hooks...Done
Reloading FreePBX
Error(s) have occured, the following is the retrieve_conf output:
exit: 1
Exception: Unable to connect to Asterisk Manager from /var/lib/asterisk/bin/retrieve_conf, aborting in file /var/lib/asterisk/bin/retrieve_conf on line 13
Stack trace:
1. Exception->() /var/lib/asterisk/bin/retrieve_conf:13
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions...

No other errors are reported during the FreePBX 13 / Asterisk 13 installation.

Asterisk is definitely running:

root@freepbx:~# asterisk -rvvvvv
Asterisk 13.7.2, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer markster@digium.com
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.7.2 currently running on freepbx (pid = 1602)
freepbx*CLI> core show sysinfo

System Statistics

System Uptime: 0 hours
Total RAM: 948140 KiB
Free RAM: 669576 KiB
Buffer RAM: 23844 KiB
Total Swap Space: 102396 KiB
Free Swap Space: 102396 KiB

Number of Processes: 159

and

root@freepbx:~# fwconsole restart
Running FreePBX shutdown...

Checking Asterisk Status...
Run Pre-Asterisk Shutdown Hooks

Shutting down Asterisk Gracefully...
Press C to Cancel
Press N to shut down NOW
Stopping Asterisk...
120/120 [============================] 100%
Asterisk Stopped Successfuly

Running Post-Asterisk Stop Scripts
Running FreePBX startup...
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions...
18319/18319 [============================] 100%
Finished setting permissions

Checking Asterisk Status...
Run Pre-Asterisk Hooks

Starting Asterisk...
100/100 [============================] 100%
Asterisk Started on 1798

Running Post-Asterisk Scripts

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@Sebbo wrote:

Hi,

is there a way to change the destination for all extensions from the Voicemail to a specific ring group?

It would be nice, if it's possible via CLI, because the GUI does not work yet, as I've already reported here: http://issues.freepbx.org/browse/FREEPBX-11475

I need to change this as soon as possible.

Hope, somebody can help me.

Thanks,
Sebastian

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@dvanaken wrote:

It has been a while since I built a new server and before I take a wrong step I wanted to check out the procedure. This is what I would do unless corrected:

1. Bring old server up to fpbx 13 in module admin -
   --- do I need to update all modules?
2. take a backup
   --- what level of backup? I just want the asterisk database and I ran into problems in the past moving too much.
3. restore backup to new server..?

I also have custom stuff in extensions_custom.conf. Can I drop that in place and expect it to work? Or do I need to be aware of changes from Asterisk 1.8 to 11?

Thanks for the help.

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@GeekBoy wrote:

Had a security breach on an extension this morning.

Extension password is a twelve digit alpha-numeric code with a tested working fail2ban process in place.
Thus, brute force intrusion is unlikely.

This PBX has about eight extensions, with only one affected. Looking in the logs, they look clean with no usual activity, or access.

In fact the fail2ban log shows zero password failures in the entire log for that extension, except for today when I changed the password and the legitimate phone tried to authenticate.

So it appears the intruder was able to obtain the password someway.

On a note the extension is in private residence, connected to a router, with the voip phone with he default admin/admin credentials in place.

Any ideas?

Thanks

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@Sebbo wrote:

Hi,

we have a little issue with some phones, which is very annoying.

We are using following hardware/software:
- FreePBX Distro (Version: 13.0.54; Core Version: 13.0.42; Asterix Version: 11.20.0)
- snom 320 (Firmware: snom320-8.7.5.35-SIP-f.bin)
- snom 370 (Firmware: snom370-8.7.5.35-SIP-f.bin)

We have one external datacenter with IPfire and FreePBX and multiple office locations with DrayTek and FRITZ!Box as router.

Between all offices and the datacenter we don't have any IPsec/LAN-to-LAN connection. All phones uses the PBX domain as registrar, which points to an external IP address mapped on the IPfire. IPfire has NAT/Forwarding rules for following ports to the FreePBX server:
- 5060/UDP
- 10 000/UDP - 20 000/UDP

I've already tried the IP instead of the domain for the registrar, but this doesn't improved or fixed it.

Network information:
- Datacenter: 10.40.1.0/24
- Office A: 10.1.1.0/24
- Office B: 10.20.1.0/24
- No LAN-to-LAN / IPsec connection anywhere

Anyway, at all locations we have some phones with that problem, that they are sometimes not reachable until I click on "Re-Register" in the webinterface or just reboot the phone. Not all phones are affected by this problem.

In the log of one snom phone, I could find following very often:

Feb 9 11:38:13 [ERROR ] PHN: RTCP: invalid port RC4 for stream
Feb 9 11:38:13 [ERROR ] PHN: RTCP: invalid port RC4 for stream
Feb 9 11:38:13 [ERROR ] PHN: RTCP: invalid port RC4 for stream
Feb 9 11:38:13 [ERROR ] PHN: RTCP: invalid port RC4 for stream
Feb 9 11:38:18 [ERROR ] PHN: RTCP: invalid port RC4 for stream
Feb 9 11:38:18 [ERROR ] PHN: RTCP: invalid port RC4 for stream

This was also found in the logs:

Feb 9 12:04:51 [WARN ] SIP: process_registrar_packet: 401 needs 128 bit nonce
Feb 9 12:04:51 [NOTICE] SIP: process auth: Match challenge for user=024, realm=asterisk
Feb 9 12:04:52 [ERROR ] PHN: Wrong Dst values: .
Feb 9 12:08:03 [NOTICE] PHN: TPL: Socket 60 idle/connect timeout
Feb 9 12:08:08 [NOTICE] PHN: TPL: Socket 61 idle/connect timeout
Feb 9 12:10:07 [CRITIC] PHN: Warning: Ignore invalid parameter: REREGISTER
Feb 9 12:10:07 [WARN ] SIP: process_registrar_packet: 401 needs 128 bit nonce
Feb 9 12:10:07 [NOTICE] SIP: process auth: Match challenge for user=024, realm=asterisk
Feb 9 12:10:07 [WARN ] SIP: process_registrar_packet: 401 needs 128 bit nonce
Feb 9 12:10:07 [NOTICE] SIP: process auth: Match challenge for user=024, realm=asterisk
Feb 9 12:10:08.001 [NOTICE] SIP: Registration Metrics failed
Feb 9 12:10:08.001 [NOTICE] PHN: TPL: Socket 62 idle/connect timeout

I've already searched for "RTCP: invalid port RC4 for stream", "Wrong Dst values" and "process_registrar_packet: 401 needs 128 bit nonce", but I couldn't found anything (helpful) related to this.

Due I'm not able to call these failed pones internal by dialing the extension number, I think it isn't a trunk issue, why I won't post these settings yet. If you need them, I can post them, if needed.

Are there any special configurations for NAT needed? Do I need to change something on IPfire? What is/are the causes for this issue?

If you need any further information, don't hesitate to ask me for.

I hope, somebody can help me out. Thanks in advance!

Best Regards,
Sebastian

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