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Creating custom extensions in mysql database

March 16, 2019, 11:33 am
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@Alkbert wrote:

Hi friends. We have an interconnection agreement with an PSTN operator, supplying us with a range of telephone numbers like part of the deal. The configuration of FreePBX for the availability of the calls is as follows:

exten => 97000092,1,Dial(local/1262@from-internal,20)
exten => _0197XXXXXX/1262,1,Set(CALLERID(num)=97000092)

So the extension 1262, when calls a PSTN number, changes its ID to 97000092, which is recognized in the telephone network as a PSTN number and can be routed properly. When you call 97000092, it is redirected the call to our Asterisk and the 1262 starts ringing. Everything related to the interconnection had been written in the custom_extensions.conf.

Adding to that, we have configured a trunk properly and an outbound route like this:

_0197XXXXXX/97XXXXXX

If you notice, 97XXXXXX is the range that the PSTN provider had supplied us. Any unauthorized access is prohibit if you are not “registered” in the .conf file (the configuration was stated previous).

Now the issue or problem is the increasing number of extensions we have that need to be accountable in amounts of telephones given. I’ve been searched that FreePBX has an mysql server running and I’ve noticed that has a custom_extension table there. So: is it possible to write all the configurations in the extensions_custom.conf file in this table?

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Telekom SIP Trunk

March 18, 2019, 12:38 am
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@maeherrasenhaus wrote:

Hi there,
has anybody a working chan_sip or chan_pjsip config he could share with me?
My problem is, that I can receive calls from the outside via the Telekom SIP trunk, but I can not place any calls probably due to a failing registration…
Cheers Michael

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The person at extension XXX is unavailable

March 18, 2019, 8:20 am
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@nelsongg94 wrote:

Hi,

I have a FreePBX server (14.0.5.25) with ~30 working extensions. One of them (403) is responding with “the person at the extension … is unavailable” message. I’ve tried rebooting the phone (GXP1610) and deleting/re-creating the extension and nothing worked. After I added the PJSIP extension again it doesn’t play the same message, I only hear the busy tone.

Here are the log messages when I call the extension:

[2019-03-18 15:18:49] WARNING[29459][C-00000345]: chan_sip.c:22959 func_header_read: This function can only be used on SIP channels.
[2019-03-18 15:18:49] WARNING[29459][C-00000345]: chan_sip.c:22959 func_header_read: This function can only be used on SIP channels.
[2019-03-18 15:18:49] WARNING[29459][C-00000345]: chan_sip.c:22959 func_header_read: This function can only be used on SIP channels.
[2019-03-18 15:18:49] WARNING[29459][C-00000345]: chan_sip.c:22959 func_header_read: This function can only be used on SIP channels.
[2019-03-18 15:18:49] WARNING[29459][C-00000345]: chan_sip.c:22959 func_header_read: This function can only be used on SIP channels.
[2019-03-18 15:18:49] WARNING[29459][C-00000345]: chan_sip.c:22959 func_header_read: This function can only be used on SIP channels.
[2019-03-18 15:18:49] WARNING[29459][C-00000345]: chan_sip.c:22959 func_header_read: This function can only be used on SIP channels.
[2019-03-18 15:18:49] WARNING[29459][C-00000345]: chan_sip.c:22959 func_header_read: This function can only be used on SIP channels.
[2019-03-18 15:18:49] WARNING[29459][C-00000345]: chan_sip.c:22959 func_header_read: This function can only be used on SIP channels.
[2019-03-18 15:18:49] WARNING[29459][C-00000345]: chan_sip.c:22959 func_header_read: This function can only be used on SIP channels.
[2019-03-18 15:18:49] WARNING[29459][C-00000345]: chan_sip.c:22959 func_header_read: This function can only be used on SIP channels.
[2019-03-18 15:18:49] WARNING[29459][C-00000345]: chan_sip.c:22959 func_header_read: This function can only be used on SIP channels.
[2019-03-18 15:18:49] NOTICE[29459][C-00000345]: app_dial.c:1000 do_forward: Not accepting call completion offers from call-forward recipient Local/1@from-internal-0000003d;1
[2019-03-18 15:18:49] NOTICE[29459][C-00000345]: core_local.c:756 local_call: No such extension/context 1@from-internal while calling Local channel
[2019-03-18 15:18:49] NOTICE[29459][C-00000345]: app_dial.c:1106 do_forward: Forwarding failed to dial ‘Local/1@from-internal’

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Wanted to block calls from each phone but to talk to each other if the call is transfered by a third phone

March 18, 2019, 8:23 am
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@2btested123 wrote:

I have a weird request. I want to block specific extensions from calling each other (’‘x’’ extension to call ‘‘y’’ extension) but if they need to call each other ‘‘x’’ extension needs to call a ‘‘z’’ extension which has communication with both to transfer the call to ‘‘y’’ extension.

For the moment i block communication between each other by the code below located in extensions_custom.conf

;disallow calls from extensions 1XX to 2XX
exten => _2XX/_1XX,1,Hangup

Is there any way to make this work?

Thanks in advance!

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Follow Me Doesnt Work when Trunk goes directly to extension

March 18, 2019, 8:49 am
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@markcrobinson wrote:

The phone on my desk can be reached internally as ext 151 and by dialing a number whose inbound route is directed to ext 151.
I have a follow-me set up to my cell on ext 151.
When I dial my main number, choose 151, the follow me works fine.
When I dial the number connected to the inbound route that goes directly to 151, the follow me does not work.
Any clues as to why?

13.0.195.26

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Call Quality Issues

March 18, 2019, 10:38 am
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@dvsatech wrote:

I have several FreePBX systems configured through FreePBX Hosting (CyberLynk). I have been having call quality issues almost form the start (over a year now). I have had issues with calls to POTS lines, fax machines, and even extension to extension (i.e. not using my SIP provider Telnyx). Cyberlynk keeps telling me that they see no issues but I have had audio issues even listening to system recordings on my extension. I have upgraded to bigger servers and recently build the largest virtual server they offer and with only 1 extension configured still had issues???

Anyone else using FreePBX hosting at Cyberlynk experiencing these issues? I have 0 packet loss between me and Cyberlink (my server) and my network is 75Mbps fiber Internet (Verizon FiOS) and <4ms response time to my servers. I have had issues with Grandstream GXW4008, Polycom VVX410, and Grandstram GXP2170 and I experience issues on multiple servers?

I am on FreePBX FreePBX Distro 14.0.5.25 ‘VoIP Server’, and all my servers have the most recent patches applied to both the OS and the PBX. The issue appears to have really gone bad in the last month where call quality is terrible, almost unusable. I have about 100 customers on 5 servers with most customers (60) on 1 of the 5 servers and NONE of the servers ever have more than 2-3 simultaneous calls at any given time (I think the most was 3 ever)

.I have looked at SAR, and the freepbx statistics and even when call quality is bad there is at least 20% idle processor and plenty of RAM. I have noticed that while running TOP that when the call quality issues occur that there was always a PHP process running that pops to the top and looks lie its performing some calendar or other scheduling updates.

Anyone else running Cyberlynk? I was thinking of switching to amazon, anyone have freepbx amazon virtual services running? Am I wrong to expect crystal clear calls with 100+ Mbps fiber internet connections that have 0% packet loss to my servers IP?

My SIP provider (Telnyx) built a special MPLS network to amazon for VoIP services, I cant get cyberlynk to look into this even though they clam to have 40,000 freepbx users.Seems to me like that could be a good pairing for them. Suggestions?

DJ

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Weekly update are failing. Error importing repomd.xml for sng-base

March 19, 2019, 8:27 am
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@brucestclair wrote:

Did a yum clear all. Then ran yum update and received the following error.

failure: repodata/repomd.xml from sng-base: [Errno 256] No more mirrors to try.
://sng7.com/os/7.5/os/x86_64/repodata/repomd.xml: [Errno -1] Error importing repomd.xml for sng-base: Damaged repomd.xml file

Getting email daily about failing to check updates. Sorry linux noob.

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Creating BFL for Ring Groups on VVX410

March 19, 2019, 11:29 am
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@cdccomputers wrote:

i have about 15 x VVX410 and having an issue where I cant configure my BFLs to work.

On our old system (Grandstream PBX & Grandstream Phones) The Hands sets had 2 x BFLs, One for Sales Team and One for Service Team. So the staff could see whos calling for each department.

Since moving to FreePBX and this VVX410 I have not been able to do this.

Any assistance would be appreciated.

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Strict RTP Learning - Help

March 19, 2019, 12:54 pm
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@jyanta wrote:

Hi Team, I noticed when dialing out lately, there has been a long delay prior to ringing. I pulled up Asterisk and noticed the following logs:

Spawn extension (from-pstn, 2721111, 1) exited non-zero on ‘PJSIP/Flowroute-1-00000007’
– PJSIP/Flowroute-1-00000007 Internal Gosub(func-apply-sipheaders,s,1(7)) complete GOSUB_RETVAL=
– Called PJSIP/94720581*17052721111@Flowroute-1
– PJSIP/Flowroute-1-00000007 is making progress passing it to PJSIP/201-00000006
> 0x7f86cc001f80 – Strict RTP learning after remote address set to: 23.29.31.91:17184 (I am guessing Flowroute?)
> 0x7f8694016af0 – Strict RTP learning after remote address set to: 10.10.10.90:2288 (Internal Phone Extension)
– PJSIP/Flowroute-1-00000007 is making progress passing it to PJSIP/201-00000006
> 0x7f86cc001f80 – Strict RTP switching to RTP target address 23.29.31.91:17184 as source
> 0x7f8694016af0 – Strict RTP switching to RTP target address 10.10.10.90:2288 as source
> 0x7f8694016af0 – Strict RTP learning complete - Locking on source address 10.10.10.90:2288
> 0x7f86cc001f80 – Strict RTP learning complete - Locking on source address 23.29.31.91:17184
– PJSIP/Flowroute-1-00000007 answered PJSIP/201-00000006

Any suggestions? I am thinking the 23.29.31.91 IP is the Flowroute sip but the 10.10.10.90 is my Phone IP. Any ideas would be great!

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Asterisk -> Cisco -> PSTN. Playback message when call realy answered

March 19, 2019, 10:10 pm
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@alexmyt wrote:

Hello!

I have FreePBX 14 (Asterisk 13) installation, connected to PSTN over SIP with Cisco2911 with FXO cards.
I need playback message to called person, and I try do it by adding to trunk dial option A(message).

The problem is that message begins to play when Asterisk connected to Cisco, not when called person pick up the phone.

I think this is becouse Asterisk receive SIP status 200 OK when he connect to gateway, not to called person.

Is it possible to do so Asterisk understand when real human pick up the phone? Or, may bee, configure a Cisco to send SIP status 180 Ringing until PSTN phone not answered?

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Setup Sip Trunk With UCOM Provider from Armenia

March 20, 2019, 1:43 am
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@armanbaghajyan wrote:

HI.

I decided write this post, because I spent whole week trying to setup trunk with Ucom Armenia SIP provider and couldn’t find any information about configuration with outbound proxy. Finally, I set up the trunk and it works great.

I had used chan_sip for trunk.

For general tap in trunk setup page.

Trunk Name - SomeName
Outbound CallerID - YourPhonNumber given from Ucom

Outgoing Sip Settings

username=Username/PhoneNumber from Ucom
fromuser=Username/PhoneNumber from Ucom
secret=SECRET
host=(outbound proxy domain name)
outboundproxy=(outbound proxy domain name)
fromdomain=(domain name)
port=5060
type=friend
context=from-trunk
insecure=port,invite
trustrpid=yes
sendrpid=yes
directmedia=no
qualify=yes
keepalive=45
nat=yes
dtmfmode=rfc2833
disallow=all
allow=ulaw

Incoming Sip Settings

User Context - Username/PhoneNumber from Ucom

username=username
type=friend
secret=SECRET
qualify=yes
port=5060
outboundproxy=outboundproxy
nat=yes
insecure=invite,port
host=outboundproxy
dtmfmode=RFC2833
context=from-trunk

Register String
Username@domainname:secret:username@outboundproxydomainname/username

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Dialplan setup AMD FreePBX

March 20, 2019, 4:15 am
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Calls fail to connect in one direction only across vlan

March 20, 2019, 9:05 am
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@SterlingPkg wrote:

Setup: Fanvil i31S Security Door Phone (door access control + video + sip phone) on vlan 71 (vlan tagged port for security), FreePBX(pbxact) box on untagged port. Ubiquiti Unifi router doing inter-vlan routing, no firewall rules to block traffic. Door phone registers, and is operable in the typical scenario (call outbound from door phone), no specific tuning on the Fanvil door phone, UDP transport.

Scenario 1: Initiate call from Fanvil door phone, call connect, two way audio available. Perfect

Scenario 2: Initiate call from Sangoma desk phone, call doesn’t connect, Fanvil door phone locks up and reboots

Scenario 3: Move Fanvil door phone to untagged port. Initiate call from Sangoma desk phone, call connects, two way audio, no issues.

I’m assuming I’m missing something obvious on the inter-vlan setup, tips or suggestions?

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Sometimes calls are not accepted

March 20, 2019, 2:23 pm
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@lkonings wrote:

Hello,

I’m using freepbx 14 with asterisk 13 and configured:

  1. Inbound route which goes to a group with extenentions in it.
  2. Outbound route which uses a sip trunk (point 3)
  3. SIP trunk

The system is not stable in terms of accepting calls. Without changing anything, several calls are accepted without problems and sometimes i’ve got the message “This call cannot be accepted at this moment, please try again later”. When this happens i don’t see any information in the asterisk console (asterisk -rvvvv) so it looks like this message is comming from the provider and not from asterisk/freepbx.

Can anyone help me out? I’m stuck…

Thanks a lot.

Kind Regards, Leo Konings.

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Voice Mail Caller ID on Voicemail Email

March 21, 2019, 2:53 am
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@Frog1 wrote:

Hi,

Could someone assist me with Changing the Voicemail I.D listed on the Voicemail email please?

What I mean is when a voicemail is left after no answer, an email is sent out that looks like the following.

From: Test

Length: 0:00 seconds

Date: Wednesday, March 20, 2019 at 04:29:26 PM.

I’d like to change the from name but I’ve searched high and low and cannot see any option to change this.

Any help would be greatly appreciated.

Thanks,

Wayne.

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Systemclt - Authorization not available

March 21, 2019, 7:19 am
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@jacovanwyk97 wrote:

Hello, I’m quite new to this.

I’m trying to install freepbx on centos 7 but whenever I’m using the systemctl command it gives me an authorization not avalable message.

For example:

[root@user-pbx-37 ~]# systemctl enable mariadb.service

Authorization not available. Check if polkit service is running or see debug message for more information.
Failed to execute operation: Connection timed out

Anyone know how I can fix this?

Thanks

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Inbound route - how can I have some delay?

March 21, 2019, 7:29 am
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@mst wrote:

Experts,

I need to have some delay on PBX answering inbound calls (1-3 seconds). How can I achieve that?

So the call is kind of delay from sip provider and FreePBX answer inbound call after 1-2 seconds delay.

Thank you so much.

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Upgrading to New System

March 21, 2019, 7:31 am
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@BGM wrote:

We are running FreePBX 12, and I want to upgrade to 14. I would follow this guide to migrate everything, but my problem is that since the new FreePBX is not activated, I have no access to System Administration in the new system. How can I make sure the new system is configured when I don’t have access to any of the commercial modules, nor to the system’s own configuration through the FreePBX interface?

I hate to think that I would have to buy a second license to activate the system just for the time needed to migrate everything. I need to have both systems running correctly in order to configure the new one before moving from my production server to the new server.

The docs for the Conversion Tool says: “The conversion tool requires that the new machine is Activated.” Do I really have to purchase a second license to do this?

Has anyone else had this problem? What do you do?

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Digium D50 fail2ban

March 21, 2019, 7:38 am
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@learjet3204 wrote:

the phone signs in… Gets its config…
works great for anywhere from 1min to 30mins
then i will get some for of the following error.
different attempt numbers.

The IP XX.XX.XX.XX has just been banned by Fail2Ban after

9 attempts against SIP on localhost.

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How to collect incoming caller ID in real time

March 21, 2019, 8:35 am
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@aaaa2209 wrote:

Hello. I want to collect the incoming caller ID in real time for a programmer. What he is trying to do is to put those caller number information in a system he developed. I don’t want him to access the freepbx server if it’s possible. How can I do that? Is it possible to ring a sip client like zoiper or microsip, then exporting the infomation from the client?

Thank you!

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