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One way audio one or twice a day, Intermittent, all other calls are working fine, callback will work fine too

April 18, 2018, 3:51 pm
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@nickmcs wrote:

Hello

I am having an issue with our IP phone system where it is getting one way audio, the client can’t hear us when we call, if the call is placed right away it will work just fine.

the setup
Freepbx server > VNP > site A
Freepbx server > VNP > site B
Freepbx server > VNP > site C
Freepbx server > VNP > site D
router: Edgerouter POE5
FreePBX 13.0.192.16

I am at a lost, I have contacted our SIP provider, but they can’t help right away because they need to check with their media peer.

The issue has been happening for a few months as far as I can recall, at first I didn’t think it was our system, but once I started doing recording of the calls it was clear that the issue is with our system and not with bad cell call or something like that on the clients end.

I have provided our SIP provider with tcpdump and wav files already, just waiting now on them to find out what’s going on, been a couple of weeks already.

Anyone experienced this, wonder if this is an issue with my setup or SIP provider? What could be causing one way audio, always happends on outgoing call and client can’t hear.
Please help. thank you

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Intel Nuc models

April 19, 2018, 12:35 am
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@Mynorx wrote:

Just wondering if anyone has successfully installed FreePBX 14 in an intel nuc (model #)? Im looking to build a small system (less than 10 users) for an electrical contractors office. Thank you.

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Time Groups/Conditions

April 19, 2018, 2:10 am
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@risaw1981 wrote:

Hello you beautiful people!!

I have been playing with my FreePBX configuration for some time now. I had a basic setup with Mon - Fri 09:00 - 17:00 time conditions working nicely. Now I am trying to apply Bank Holidays and I have hit a brick wall. I’m in the UK and cannot see where I am going wrong with this, here’s my Bank Holiday Time Group that isn’t working. Basically if the time group does match, calls go to ring group, if it doesn’t match it goes to IVR. All calls go to IVR although I am on a working day.

Screenshot-2018-4-19%20AsteriskNow%20Admin
Screenshot-2018-4-19 AsteriskNow Admin.png687x2350 122 KB

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Cron email to asterisk@mydomain

April 19, 2018, 2:21 am
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@steve_pbuk wrote:

I reveid the below email last night after performing some updates on the server. I’m not worried about the message because everything is working fine. My question is where do I set the email address the these emails get sent to? At the moment is it sending to: asterisk@mydomain.co.uk (It actually sends to my domain but changed it for this post…)

I’m 99% sure I have never set that email in the system. Below is a screenshot of my notifications page:

2018-04-19%2009_35_46-FreePBX%20Administration

Here is a copy of the email that I received.

Ticket recall
Cron  [ -x /var/lib/asterisk/bin/schedtc.php ] && /var/lib/asterisk/bin/schedtc.php
From: (Cron Daemon) 
Sent: 18 April 2018 21:42
To: asterisk@mydomain.co.uk _(Tweaked by me for this post)_
Subject: Cron [ -x /var/lib/asterisk/bin/schedtc.php ] && /var/lib/asterisk/bin/schedtc.php

Exception: Asterisk is not connected in file /var/www/html/admin/libraries/php-asmanager.php on line 242 Stack trace:
 1. Exception->() /var/www/html/admin/libraries/php-asmanager.php:242
 2. AGI_AsteriskManager->send_request() /var/www/html/admin/libraries/php-asmanager.php:591
 3. AGI_AsteriskManager->Command() /var/www/html/admin/libraries/php-asmanager.php:1544
 4. AGI_AsteriskManager->parseAsteriskDatabase() /var/www/html/admin/libraries/php-asmanager.php:1513
 5. AGI_AsteriskManager->database_show() /var/www/html/admin/libraries/php-asmanager.php:210
 6. AGI_AsteriskManager->LoadAstDB() /var/www/html/admin/libraries/php-asmanager.php:1599
 7. AGI_AsteriskManager->database_get() /var/www/html/admin/modules/timeconditions/bin/schedtc.php:28

Can I change the To email anywhere?

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Call Forwarding/Proxying

April 19, 2018, 3:21 am
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@echo501 wrote:

Hello,

Is it possible to set up an extension at my office that can be dial and from which can then provide a number to forward the call? That way if a person makes a call for work from their cell phone or another off site phone, etc, they could call the office, notionally dial the ext. NNN, and then dial the actual number to call. This would make the call appear to the person on the other line that the call came from the office, instead of their cell phone? Essentially, I want to know if we can route / proxy a call through the office for work related calls?

I would think this this is possible but I’m not a telecon engineer so I don’t know all the pieces necessary to do this.

Any help would be most appreciated.

Thanks

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Change Background Color of FreePBX webpage

April 19, 2018, 9:49 am
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@Christopheric1 wrote:

I am looking to differentiate the color of the background page for the freepbx web setup screens. I have multiple clients and sometimes I can be in more than one looking at settings and make a change to the wrong one. I would love to be able to have a color background so that I know which one I am changing.

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Queue - To Continue to Hold

April 19, 2018, 11:06 am
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@rshah_exo wrote:

Hi,

We have a queue, we are breaking out of it to an IVR at a predetermined interval and we’d like it to say “To Continue to hold, press 1 - to leave a message, press 2”.

How can we do this and maintain the callers place in line? We know how to break out of the queue into an IVR – and I assume we can just have option 1 go back to the queue, but would that keep their place in line?

RS

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Phone long numbers

April 19, 2018, 11:11 am
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Freepbx crashing network, 128gb of upload but only 10.1mb of download

April 19, 2018, 3:43 pm
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@JGAN wrote:

I have a fresh install of Freepbx 14.0.1.24 and Asterisk 13.19.1. For some reason it has a ton of upload bandwidth usage and seems to essentially DDOS my network – in the last 24 hours it uploaded 128gb of data but only downloaded 10.1mb.

I use Twilio as my SIP trunk, which is on port 5060. The server log shows a lot of random bots trying to brute force an extension registration, but I have anonymous registration turned off so they are being denied. It doesn’t always seem to correlate with when Freepbx uploads large amounts of data however. Is it still perhaps related or is something else the issue?

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Possible Call Recording Bug?

April 19, 2018, 11:37 pm
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@bigjess007 wrote:

I think I may have found a bug in call recording. Or I don’t have something setup properly :tired_face:

I want to record every call (in and out) on a particular extension. In Extensions settings > Advanced tab, I have “Forced” set for all 4 calls types. If I call this extension from another extension or use this extension to call another extension, recording works. If I use this extension to make a call out, recording works.

However, if a call comes in from a ring group to this extension (either via trunk or calling the ring group directly), the call is not recorded. The inbound route that feeds this ring group has settings > Other tab > Call Recordings set to Don’t Care. The ring group that sends the call to this extension has it’s setting > Call Recording set to Don’t Care.

When I look at the logs, I see the call comes in, and I see the recording check against the trunk & ring group, and the value is dontcare; which matches the ring group’s settings.

But there is no recording check entry when the call is answered by the extension that has it’s recording settings set to force.

So why when a call is delivered by a ring group is there not a recording check being done at the extension level when the extension answers?

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Zulu does not install after uninstallation

April 19, 2018, 11:39 pm
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@CMPTRWRpier wrote:

After un-installing Zulu UC I can not install it anymore, it gives me the following error:

> Updating tables zulu_interactions_interaction_states, zulu_softphones, zulu_tokens, zulu_interactions_contacts, zulu_interactions_interactions, zulu_interactions_members, zulu_interactions_streams, zulu_interactions_stream_bodies, zulu_interactions_stream_links, zulu_login_tokens...Done
> PJSIP Transports for WS and WSS must be enabled in Asterisk SIP Settings under the Chan PJSIP Settings tab. Then restart Asterisk
> Error(s) installing zulu:
> Failed to run installation scripts
> Updating Hooks...Done

I tried to delete the Zulu directories, re-download and install zulu but the result does not change

"[root@computerware ~]# fwconsole ma downloadinstall zulu
No repos specified, using: [standard] from last GUI settings

Downloading module 'zulu'
Processing zulu
Verifying local module download...Verified
Extracting...Done
Download completed in 2 seconds
Updating tables zulu_interactions_interaction_states, zulu_softphones, zulu_tokens, zulu_interactions_contacts, zulu_interactions_interactions, zulu_interactions_members, zulu_interactions_streams, zulu_interactions_stream_bodies, zulu_interactions_stream_links, zulu_login_tokens...Done
PJSIP Transports for WS and WSS must be enabled in Asterisk SIP Settings under the Chan PJSIP Settings tab. Then restart Asterisk
Unable to install module zulu:
 - Impossibile eseguire gli script di installazione
Updating Hooks...Done"

Someone knows how to help me?

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Migrating OBi110 and Linksys SPA3102 to current (V14) distro: Firewall issue?

April 20, 2018, 1:47 am
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@LesD wrote:

I’m just migrating from 2.11 PIAF system to a reasonably current distro (PBX Firmware:12.7.3-1708-1.sng7). All modules are up to date.

I have two PSTN lines connected via two adapters (OBi110 and SPA3102) and trying to switch them over to the new system.

Copied over trunks and incoming and outgoing routes without making any changes.

I have logged into the two adapters and changed the IP of the server (one reference in each case as far as I can see) but they do not work (tested incoming calls only).

The only relevant log entry I can find is in the fail2ban log:

[2018-04-20 11:36:22] WARNING[2518] res_pjsip_registrar.c: Endpoint 'anonymous' has no configured AORs
[2018-04-20 11:36:22] SECURITY[2920] res_security_log.c: SecurityEvent="FailedACL",EventTV="2018-04-20T11:36:22.653+0300",Severity="Error",Service="PJSIP",EventVersion="1",AccountID="anonymous",SessionID="dcede6ee@10.27.27.192",LocalAddress="IPV4/UDP/10.27.27.247/5060",RemoteAddress="IPV4/UDP/10.27.27.192/5061",ACLName="registrar_attempt_without_configured_aors"

The IP is that of the OBi110 adapter I am testing.

Can anyone interpret the meaning of the messages?

Is the reference to ‘pjsip’ an issue as I believe the trunk is defined as ‘sip’?

Chan_Sip Peers shows:

Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description
1-pstn/1-pstn             (Unspecified)                            D  No         No             0        UNKNOWN

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Freepbx 14 - Superfecta - CallerIDService module bug

April 20, 2018, 6:49 am
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@sf1965 wrote:

Superfecta module for CallerIDService module has a bug at line 59, with the code shown below.

53: // CallerID service returns CNAM with the following prefixes if there are errors
54: $st_error = strstr($sname, "CNAM ");
55: $st_unknown = strstr($sname, “UNKNOWN”);
56: $st_unavail = strstr($sname, “UNAVAILABLE”);
57:
58: // give up if any errors
59: if($st_error || $st_unkown || $st_unavail) {
60: $this->DebugPrint(_(“Error in Lookup.”));
61: return;
62: }

The variable $st_unknown is misspelled at line 59. In the module the variable is spelled as $st_unkown, by changing the variable to the correct spelling the module works without any problems.

Corrected code shown below:

53: // CallerID service returns CNAM with the following prefixes if there are errors
54: $st_error = strstr($sname, "CNAM ");
55: $st_unknown = strstr($sname, “UNKNOWN”);
56: $st_unavail = strstr($sname, “UNAVAILABLE”);
57:
58: // give up if any errors
59: if($st_error || $st_unknown || $st_unavail) {
60: $this->DebugPrint(_(“Error in Lookup.”));
61: return;
62: }

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Noob question: ports to forward if I have only local extensions

April 20, 2018, 7:43 am
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@wassy83 wrote:

Sorry for this noob question, but I just want to be sure about this:

  • freepbx is behind my router’s NAT
  • firewall on my router is not blocking any outbound/inbound traffic
  • all my extensions are local, in the same subnet of my freepbx appliance.
  • external calls are coming from several pjsip trunks.
  • I don’t need access to my pbx from the outside world.
    is necessary to forward any port from the outside to my freepbx?
    many thanks

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Freepbx configuration - different Signalling and RTP ips

April 20, 2018, 8:04 am
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@daseagle wrote:

Hello,

Because of various circumstances, I “inherited” an Asterix/Freepbx server into my network. Pretty simple setup, one trunk with 8 outgoing concurrent calls.

Now, my ISP is upgrading some equipment on their end so it provided me with new ip addresses, to change them in our equipment.

  • signalling 99.xx.yy.10, port 5060/udp
  • RTP/media: 99.xx.yy.41

And here is my problem, since my extensive experience in old-style analogue PBX’s does little to help me.

I assume that the signalling ip is the one found in Connectivity -> Trunks -> pjsip Settings -> General tab, SIP Server field.

But the RTP/Media ip address? Where do I change that? I am reading through the documentation now, but anything to point me in the right direction would be most useful as I’m running out of time.

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No more add extension button!

April 20, 2018, 8:07 am
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@digiteltlc wrote:

I’ve just realized that two machines running 10.13.66.22 from distro , have lost the “add extension” and “quick create extension” buttons in their GUI…
The only red “delete” button is present.

Any idea ???

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Recommended Nic card for home setup?

April 20, 2018, 8:45 am
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@aquitain3 wrote:

I took the advice from my last post and am putting together a free pbx box with a 3GHz Athon 2x cpu.

Wondering what kind of NIC card I should get? I intend on running the main line through an ATA and into my analog phone network. Can I get by with a single nic card and the motherboard ethernet, or should I spring for a nice 2x NIC?

As a total aside…
I would also like to run a very lightweight IDS, so please advise if you think I can run both on the same box.

Cheers!

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LetsEncrypt update error

April 20, 2018, 9:40 am
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@Bradbpw wrote:

My LetsEncrypt cert tried to update last night and sent me this error

SECURITY NOTICE:

Some Certificates are expiring or have expired:
 There was an error updating certificate "pbx.xxxxxx.com": Error
'Requested
'http://pbx.xxxxxx.com//.freepbx-known/7251bf30b7ecfe7885a1c9259c760a74'
- Failed connect to pbx.xxxxxx.com:80; Connection timed out' when
requesting
http://pbx.xxxxxx.com//.freepbx-known/7251bf30b7ecfe7885a1c9259c760a74</br>

I am using System Admin version 14.0.12.2, which includes the LetsEncrypt port 80 setting. I also have port 80 forwarded in my router.

I attempted to disable the LetsEncrypt port in System Admin and change my UCP and my admin port to 80 (at separate times) but when the page reloaded no changes were made.

I have my firewall correctly set up as well, according to the certificate management page.

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SIP URI for incoming numbers

April 20, 2018, 9:48 am
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@knobby wrote:

Hi folks,
I´ve just set up a copy of FreePBX on a VPS, and am going through the configuration at the moment.
I have a few DID´s with a couple of compnaies which I want to route to my FreePBX. The SIP URI needs to be something@myserver - but what is the something part in FreePBX? Is it a particular extension number, e.g. 101@myserver ?
TIA

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Maximum concurrent lines for SIP extension

April 20, 2018, 10:07 am
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@mani9876 wrote:

Hello,

how many maximum concurrent lines are available for a SIP extension?
Is it somehow possible to use e.g. 4 lines for one SIP extension at the same time? If so, how can this be achieved, only if I place three other on hold, or is it somehow possible to put out the stream on different devices, e.g. if I have a suitable SIP Client on my PC?

Thank you
Regards

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