@MDTechTeam wrote:

I have a cron tab function and have an idea of how to do this if i was writing std c, but need some advice how to incorporate using code for crontab

What i want is the following

date = getDate();
switch(date){
    case apr 27:
        break;
    case jun 14:
        break;
    default:
        cp /var/spool/asterisk/tmp/playfile.call /tmp && mv /tmp/playfile.call /var/spool/asterisk.outgoing/
        break;
}

There are like 9 times in the year when i do not want the music to play at a specific time as it will interrupt the activities on those days.

Posts: 1

Participants: 1

Read full topic

@Gracien wrote:

Hi Team
We will like to push all Calls events , Call status events ,log in and log out extensions events in an specific URL.

Is it possible with SANGOMA UC60 ?

Kind Regards

Posts: 1

Participants: 1

Read full topic

@sorvani wrote:

So how did my system get into this state?

image.png797×61 7.12 KB

image.png810×313 26.3 KB

Posts: 1

Participants: 1

Read full topic

@joelones wrote:

Hello,

I’m kind of lost here and going a little mad. I just downloaded the recent version of Rapbx (for the RPi3) and upgraded the modules via the following command:

fwconsole ma downloadinstall framework --tag 15.0.16.38
fwconsole ma upgrade framework

I am now having a tough time with trying to have any one of my sip phones connect to ports 5060/5160. After trying a little I decided to test with telnet and am getting connection refused:

telnet: connect to address 192.168.1.75: Connection refused
telnet: Unable to connect to remote host

Meanwhile netstat reveals:

udp 0 0 0.0.0.0:5060 0.0.0.0:* 1159/asterisk
udp 0 0 0.0.0.0:5160 0.0.0.0:* 1159/asterisk

fail2ban is stopped:
service fail2ban stop

root@raspbx:~# iptables -L -v
Chain INPUT (policy ACCEPT 0 packets, 0 bytes)
 pkts bytes target     prot opt in     out     source               destination

Chain FORWARD (policy ACCEPT 0 packets, 0 bytes)
 pkts bytes target     prot opt in     out     source               destination

Chain OUTPUT (policy ACCEPT 0 packets, 0 bytes)
 pkts bytes target     prot opt in     out     source               destination

Weirdly I am able to connect to the UI and SSH with no problems, what in gods name am I missing here?

Please any suggestions would be greatly appreciated.

Posts: 3

Participants: 2

Read full topic

@odwar wrote:

Hello, first i would like to thank for this lovely community and mainly for the FreePBX project which is a perfect solution for a problem I’m trying to solve. I’m facing a simple task, I live in an Appartment building and the doorbells are an intercom from a company called 2N, in my flat i have a Grandstream phone and it works really simple, the intercom makes a direct sip call to the IP address of the Grandstram phone and if the phone enters some keypad combination while on the call, it unlocks the front door.

So as a first thing I tried to used a soft phone instead of the Grandstream phone, gave it the same IP addres and I succeeded with finishing the call. So the idea is, to have a server on the IP address the intercom is trying to call and then route the call eg. to my cellphone (with a sip client installed) or respond with a message.

I Enabled Anonymous Calls in SIP settings, then i created an extension “1”, set up Inbound route and routed all to the extension 1 - I logged on my cellphone’s sip client, as a sip server address i specified 192.168.1.XXX (where freepbx is running) and on my laptop i tried to call anything@192.168.1.XXX and it succesfully called my mobile. Felt great. But when i try to calling without the prefix before the @, so just sip:192.168.1.XXX (as if the intercom was calling), it only gets rejected with 484 address incomplete and the call is not even estabilished (not even the asterisk log tells about the call).

Is there a way to enable redirecting to extension if the SIP call is only being made to the IP address?
Note: I cannot change any settings in the intercom since I don’t have the credentials for it. I’m also not trying to call from outside of the network, so not trunks i needed i suppose?

Thank you 1000 times for your patience.

Posts: 1

Participants: 1

Read full topic

@grahamkhan89 wrote:

Good Day All,

This is my first post here, I need some guidance on how to disable a macro that runs when ever an agent wants to make an external call,

what happens is that when ever an external call is placed by an extension they are required to enter a password. i would like to have this feature dissabled

below is a copy of the log file.

[2020-04-13 09:41:00] VERBOSE[9537][C-00000037] netsock2.c: == Using SIP RTP TOS bits 184
[2020-04-13 09:41:00] VERBOSE[9537][C-00000037] netsock2.c: == Using SIP RTP CoS mark 5
[2020-04-13 09:41:00] VERBOSE[8376][C-00000037] pbx.c: – Executing [0213714972@from-sip:1] Set(“SIP/2051-00000060”, “ORIG_EXTEN=0213714972”) in new stack
[2020-04-13 09:41:00] VERBOSE[8376][C-00000037] pbx.c: – Executing [0213714972@from-sip:2] Macro(“SIP/2051-00000060”, “extcall”) in new stack
[2020-04-13 09:41:00] VERBOSE[8376][C-00000037] pbx.c: – Executing [s@macro-extcall:1] Answer(“SIP/2051-00000060”, “”) in new stack
[2020-04-13 09:41:00] NOTICE[8376][C-00000037] res_rtp_asterisk.c: Unknown RTP codec 95 received from ‘192.168.0.21:45980’
[2020-04-13 09:41:00] VERBOSE[8376][C-00000037] pbx.c: – Executing [s@macro-extcall:2] GotoIf(“SIP/2051-00000060”, “0?s-BADAUTH,1”) in new stack
[2020-04-13 09:41:00] VERBOSE[8376][C-00000037] pbx.c: – Executing [s@macro-extcall:3] Wait(“SIP/2051-00000060”, “1”) in new stack
[2020-04-13 09:41:01] VERBOSE[8376][C-00000037] pbx.c: – Executing [s@macro-extcall:4] Set(“SIP/2051-00000060”, “TIMEOUT(digit)=1”) in new stack
[2020-04-13 09:41:01] VERBOSE[8376][C-00000037] func_timeout.c: – Digit timeout set to 1.000
[2020-04-13 09:41:01] VERBOSE[8376][C-00000037] pbx.c: – Executing [s@macro-extcall:5] Set(“SIP/2051-00000060”, “TIMEOUT(response)=3”) in new stack
[2020-04-13 09:41:01] VERBOSE[8376][C-00000037] func_timeout.c: – Response timeout set to 3.000
[2020-04-13 09:41:01] VERBOSE[8376][C-00000037] pbx.c: – Executing [s@macro-extcall:6] Set(“SIP/2051-00000060”, “Attempts=0”) in new stack
[2020-04-13 09:41:01] VERBOSE[8376][C-00000037] pbx.c: – Executing [s@macro-extcall:7] Set(“SIP/2051-00000060”, “Attempts=1”) in new stack
[2020-04-13 09:41:01] VERBOSE[8376][C-00000037] pbx.c: – Executing [s@macro-extcall:8] Read(“SIP/2051-00000060”, “Passread,vm-password,6,3”) in new stack
[2020-04-13 09:41:01] VERBOSE[8376][C-00000037] app_read.c: – Accepting a maximum of 6 digits.
[2020-04-13 09:41:01] VERBOSE[8376][C-00000037] file.c: – <SIP/2051-00000060> Playing ‘vm-password.ulaw’ (language ‘en’)
[2020-04-13 09:41:05] VERBOSE[8376][C-00000037] app_read.c: – User entered nothing, 2 chances left
[2020-04-13 09:41:05] VERBOSE[8376][C-00000037] file.c: – <SIP/2051-00000060> Playing ‘vm-password.ulaw’ (language ‘en’)
[2020-04-13 09:41:09] VERBOSE[8376][C-00000037] app_read.c: – User entered nothing, 1 chance left
[2020-04-13 09:41:09] VERBOSE[8376][C-00000037] file.c: – <SIP/2051-00000060> Playing ‘vm-password.ulaw’ (language ‘en’)
[2020-04-13 09:41:13] VERBOSE[8376][C-00000037] app_read.c: – User entered nothing.
[2020-04-13 09:41:13] VERBOSE[8376][C-00000037] pbx.c: – Executing [s@macro-extcall:9] GotoIf(“SIP/2051-00000060”, “1?s-BADAUTH,1”) in new stack
[2020-04-13 09:41:13] VERBOSE[8376][C-00000037] pbx.c: – Goto (macro-extcall,s-BADAUTH,1)
[2020-04-13 09:41:13] VERBOSE[8376][C-00000037] pbx.c: – Executing [s-BADAUTH@macro-extcall:1] Playback(“SIP/2051-00000060”, “vm-goodbye”) in new stack
[2020-04-13 09:41:13] VERBOSE[8376][C-00000037] file.c: – <SIP/2051-00000060> Playing ‘vm-goodbye.ulaw’ (language ‘en’)
[2020-04-13 09:41:14] VERBOSE[8376][C-00000037] pbx.c: – Executing [s-BADAUTH@macro-extcall:2] Hangup(“SIP/2051-00000060”, “”) in new stack
[2020-04-13 09:41:14] VERBOSE[8376][C-00000037] app_macro.c: == Spawn extension (macro-extcall, s-BADAUTH, 2) exited non-zero on ‘SIP/2051-00000060’ in macro ‘extcall’
[2020-04-13 09:41:14] VERBOSE[8376][C-00000037] pbx.c: == Spawn extension (from-sip, 0213714972, 2) exited non-zero on ‘SIP/2051-00000060’
[2020-04-13 09:42:55] NOTICE[9537] chan_sip.c: Peer ‘2051’ is now UNREACHABLE! Last qualify: 53

Posts: 1

Participants: 1

Read full topic

@bitbanger wrote:

My system has FreePBX Framework 14.0.13.26.
I’m pretty sure that a few days ago, Module Admin was showing an update to 14.0.13.28 and a total of 26 modules available for update.
Today Module Admin says my 14.0.13.26 is newer than the repo version (14.0.13.23) and a total of 25 modules available for update.

Has Framework been rolled back to 14.0.13.23? Or has the repo been restored from a backup that wasn’t up to date? Or ???

Posts: 2

Participants: 2

Read full topic

@tahmidul99 wrote:

Have anyone ever used CID Superfecta to sync Google contacts with FreePBX? I didn’t any proper guide or tutorial for setting it up. I’m desperately seeking for this solution.
“Debug/Test Run” also did not provide enough information to set it up! Can anyone please post the complete procedure for setting it up?
It’ll help a lot of peoples, i think.

Thanks.

Posts: 1

Participants: 1

Read full topic

@danielmagerusan wrote:

Hello i use FreePBX 12.0.76.6, I want to redirect all calls to my mobile phone.
what should i do in this regard?
Thanks!

Posts: 1

Participants: 1

Read full topic

@zonatura wrote:

Hello, I explain my situation to you and ask you for help as far as this is concerned. In the office I have a subscription FTTH 1 Gbit fiber Work Solution Red + VAT and for obvious limits of the Vodafone Station I have replaced it for several months with a proprietary firewall (Sonicwall TZ400, Fortinet Fortigate 60E, Mikrotik Hex S). No configuration problems with any of the firewalls. It was enough to set Vlan 1036 on the PPoE connection as indicated by Vodafone.Public IP so without NAT of the Vodafone Station and every VPN (Ipsec site to site, L2TP / Ipsec) or Port forwarding functioning perfectly and without problems. So I went to see the configuration of Voip, with relative calm, given the unnecessary nature of my business.And there it was not roses and flowers as for the data part.I first tried to configure the sip trunk with Freepbx by finding various ideas on the network but nothing.If I remember correctly I was able to register and make calls but not to receive them.Reading here and there, I saw that someone had successfully configured the Grandstream HT801 and therefore bought it.After days of tests, I managed to get to a stable configuration that allows me to make and receive calls without any problem: the line is always present, the selection of numbers immediately (too much) and the quality of the excellent call.Now during the quarantine, I started looking at the configuration on Freepbx again but there is nothing to do.Despite the support of the configuration parameters of the HT801, there was nothing to do. I can’t even register the number anymore (it will be that in the meantime the version of asterisk has changed …).Now, in addition to the obvious method of using two HT801 (one connected to Vodafone Voip and one connected to Asterisk) that works, I ask you: is there anyone who has been able to configure directly as a Sip trunk (sip or pjsip) on asterisk the Vodafone Voip line?Thanks so much for your help.Obviously no problem sharing the HT801’s configuration if someone needed it.George

Posts: 1

Participants: 1

Read full topic

@maxillosoft wrote:

I have Call forwarding to my cell with Confirm Calls turned on. When someone calls me and it forwards to my cell, if I do not answer, I get a voicemail every time that says “I’m sorry. The incoming call is no longer available or has been answered by someone else.” Anyway to prevent it from leaving that voicemail on my cell phone every time.

I am not sure if this is possible or if this has been answered already but I did not find anything in the discussions about this when searching.

Thanks
-Dimitry

Posts: 2

Participants: 2

Read full topic

@MC1 wrote:

Hello
I’m currently at version 14 and wanted to know what the advantages are to upgrading to version 15? Are there more features or any gothca’s I should know about? I’ve done a quick search on the FreePBX Wiki, but haven’t found anything.

Thanks in advance for your help.
Michael

Posts: 1

Participants: 1

Read full topic

@zfsag wrote:

I upgraded from 14 to 15 using the upgrade module.

Now I am receiving this error when trying to run Yum Update

[root@localhost ~]# yum update
Loaded plugins: fastestmirror, kmod
Loading mirror speeds from cached hostfile

One of the configured repositories failed (Unknown),
and yum doesn’t have enough cached data to continue. At this point the only
safe thing yum can do is fail. There are a few ways to work “fix” this:

 1. Contact the upstream for the repository and get them to fix the problem.

 2. Reconfigure the baseurl/etc. for the repository, to point to a working
    upstream. This is most often useful if you are using a newer
    distribution release than is supported by the repository (and the
    packages for the previous distribution release still work).

 3. Run the command with the repository temporarily disabled
        yum --disablerepo=<repoid> ...

 4. Disable the repository permanently, so yum won't use it by default. Yum
    will then just ignore the repository until you permanently enable it
    again or use --enablerepo for temporary usage:

        yum-config-manager --disable <repoid>
    or
        subscription-manager repos --disable=<repoid>

 5. Configure the failing repository to be skipped, if it is unavailable.
    Note that yum will try to contact the repo. when it runs most commands,
    so will have to try and fail each time (and thus. yum will be be much
    slower). If it is a very temporary problem though, this is often a nice
    compromise:

        yum-config-manager --save --setopt=<repoid>.skip_if_unavailable=true

Cannot find a valid baseurl for repo: schmooze-commercial/7/x86_64
You have mail in /var/spool/mail/root

Posts: 1

Participants: 1

Read full topic

@aoliva1217 wrote:

I have a archer ac4000 and I have tried using openVPN and PPTP and I can connect to the User Control Panel and Zulu but not the admin panel. I also got the system admin module thinking that would help but I try to VPN into a user and that doesn’t work either.

Is there any way for me to connect to the admin panel with either the router or the module?

Posts: 3

Participants: 2

Read full topic

@stephenl wrote:

Hello

System
Asterisk (Ver. 1.8.23.1)
FreePBX 2.11.0.37

I need to configure our PBX to allow remote workers (during this global pandemic) to make outgoing calls through our PBX.

I came across a possible soloution using DISA, however have been unable to find any notes on how to impliment

Ideally I would like the remote user to dial into our system, and dial an extension which would prompt for the DISA pin, followed by the number to be dialed

Any guidance would be appreciated

Thank you

Posts: 2

Participants: 2

Read full topic

@jin wrote:

Hi,
if someone is interested, I’m writing here to make it know that we are developing a (unofficial) docker image of FreePBX 15 with Asterisk 16 LTS.
Right now the release (0.9.1) is in advanced testing phase (using it in production since two weeks).
Other testers and positive feedback are welcome

This is the actual features list:

• 60 secs install from zero to a running full features PBX system.
• Really fast initial bootstrap to deploy a full stack Asterisk+FreePBX system
• Small image footprint
• CentOS 8 64bit powered
• Asterisk PBX Engine (compiled from scratch)
• Opus, G729, Motif codecs compiled
• FreePBX WEB Management GUI (with predownloaded modules for quicker initial deploy)
• First automatic installation managed when deploying the izpbx, subsequent updates managed by FreePBX Official Version Upgrade
• Persistent storage mode for configuration data (define APP_DATA variable to enable)
• Misc izpbx scripts (into container shell digit izpbx+TAB to discover)
• tcpdump and sngrep utility to debug VoIP packets
• fail2ban as security monitor to block SIP and HTTP brute force attacks
• supervisord as services management with monitoring and automatic restart
• postfix MTA daemon for sending mails (notifications, voicemails and FAXes)
• cron daemon
• Apache 2.4 and PHP 7.3 (mpm_prefork+mod_php configuration mode)
• Automatic Let’s Encrypt HTTPS Certificate management for exposed PBXs to internet
• Commercial SSL Certificates support
• Logrotating of services logs
• FOP2 Operator Panel (optional)
• Integrated Asterisk Zabbix agent for active health monitoring
• All Bootstrap configurations made via single central .env file
• Many customizable variables to use (look inside default.env file)
• Two containers setup: (antipattern docker design but needed by the FreePBX ecosystem to works)
  • izpbx-asterisk: Asterisk Engine + FreePBX Frontend + others services
  • mariadb: Database Backend

for more info, consult the official izPBX project docker hub page:

https://hub.docker.com/repository/docker/izdock/izpbx-asterisk

Because make successfully running Asterisk/FreePBX SIP protocol on a Cloud Native Environment is a challenge, based on user feedback, will follow FAQ and troubleshooting guide to make a little less pain the deployment (future work will be in a Kubernetes deploy via Helm Chart)

Kind regards

Posts: 1

Participants: 1

Read full topic

@edwinr wrote:

Hi, Got freepbx 13 with asterisk 13, i having a problem in getting the events from ami. Need help in which of the events that trigger the call was pickup from the otherend, call still ringing and call was hangup for Outbound calls?

Posts: 4

Participants: 2

Read full topic

@thelastdreamer wrote:

Hello to all, i need some help

i am trying to make a extensions_custom.conf custom file which when a number(external) call this number it will create a new simulated call with this specific number( as caller id) and call a specific extension
which will remain active until the end user terminates the call

Thanks to all!

Posts: 2

Participants: 2

Read full topic

@joelones wrote:

I’m trying to update a rather old version of freepbx and getting stuck at the phonebook setup.
I installed CID Superfecta and want it to pull from the phonebook (as it is setup by default). From what I understand the phonebook module no longer exists and is now the Contact Manager.

Can the Contact Manager be used the same way the phonebook module used to be? If so, how it is setup?

Basically can it be used with CID Superfecta, to pull out the “name” info?

Thanks.

Posts: 2

Participants: 2

Read full topic

@JoshuaCadman wrote:

I have a Cisco 8841 that is connected to my FreePBX Distro and thought about streaming my IP cameras with rtsp stream onto the display of my phone when I dial a number. After doing some digging I have seen many posts about app_rtsp and using that with asterisk however I have not found anywhere that explains how it works properly or found anyone that has successful got this to work. Has anyone had experience with RTSP streams with asterisk/freepbx or is this best to leave it alone?

Posts: 1

Participants: 1

Read full topic