@tgross wrote:
I have been asked to research the feasibility of providing certain features of the CER within the FreePBX/Asterisk framework.
Is anyone else familiar with those features and how they might be accomplished using FreePBX?
I know, its a broad question.
Posts: 1
Participants: 1
@AmidouFlorian92 wrote:
Hi all
I’m searching a way to configure an SMS gateway on my freepbx server.
I want to be able to send and receive automatic SMS on my server.
Is there a solution to do It using the CLI or what solution can I install to have an SMS module working with freepbx ?
Thanks
Posts: 4
Participants: 2
@tonyg wrote:
Hi,
Taking into account Kari and Baum, will i need to assign a DID to my lobby phone and set a dispatchable location information? I ask because we specifically do not want people on this phone calling out. it was meant to be a way for visitors to get an employee, we really did not want anyone waking up to the phone from outside and dialing out…but I am guessing that is what we will have to do…Anyone have any insights on this?
thanks
Posts: 3
Participants: 3
@treimers wrote:
Hi all-
I have what I hope is a pretty simple question here, and I’ll try to provide answers ahead of time to questions people may have:
I’m trying to configure a Yaelink T42G (firmware 29.73.0.50) against a FreePBX server (FreePBX 15.0.16.42), Asterisk 16.7.0
I’m getting no attempted registrations that I can detect.
Topology is:
Asterisk server has a public IP, no NAT, directly on eth0
(this is carefully firewalled).Phones are at my house behind NAT, with a single dynamic public IP
(Firewall at home is PFSense, with a public hostname known by dynamic DNS)
I’m a security/network engineer, and so I’m about 90% sure that I’ve got tcp/udp 5060/5061 and UDP 10000-20000 allowed in/out on both the public server IP and home network.
Only certain networks are allowed in to the server. I’m relatively sure that my home network by hostname AND current IP is allowed ,as is my workplace static IP range.I’ve built the extension(s) with 4-digit extensions and PJSIP.
pjsip appears to be runnig on the server:
[root@www ~]# lsof -i :5061
COMMAND PID USER FD TYPE DEVICE SIZE/OFF NODE NAME
asterisk 26388 asterisk 20u IPv4 10831715 0t0 UDP *:sip-tlsWhat do I need to put into the Yaelink though?
I see lots of notes on regular SIP (chan_sip I guess) on 5060But how do I configure that Yealink to be correct for PJSIP?
Do I need a STUN server configured, or does the public Asterisk server just figure it out?
How does NAT need to be configured on the phone and/or the Asterisk extension?Thanks Tim
Posts: 6
Participants: 4
@sebbor88 wrote:
I didn’t found this problem on forum so decided to write.
FreePBX 14.0.5.25
Asterisk 13.22.0One extension we have occured to not screen outboundcid when calling outside. I don’t know it was from beginning or from moment in time. Just got this incident now. Everything works fine for this user, calls, cdrs etc.
Problem is in the database. When i connect to sqlite database and select from astdb for AMPUSER/60886 i see all the variables:/AMPUSER/60886/rvolume|
/AMPUSER/60886/password|1111
/AMPUSER/60886/ringtimer|0
/AMPUSER/60886/cfringtimer|0
/AMPUSER/60886/concurrency_limit|3
/AMPUSER/60886/noanswer|
/AMPUSER/60886/recording|
/AMPUSER/60886/outboundcid|xxxxxxxxxx (changed fo Xes to hide)
/AMPUSER/60886/cidname|User 60886
/AMPUSER/60886/cidnum|60886
/AMPUSER/60886/voicemail|novm
/AMPUSER/60886/intercom|enabled
/AMPUSER/60886/recording/in/external|dontcare
/AMPUSER/60886/recording/out/external|dontcare
/AMPUSER/60886/recording/in/internal|dontcare
/AMPUSER/60886/recording/out/internal|dontcare
/AMPUSER/60886/recording/ondemand|disabled
/AMPUSER/60886/recording/priority|10
/AMPUSER/60886/dictate/enabled|disabled
/AMPUSER/60886/dictate/format|ogg
/AMPUSER/60886/dictate/email|
/AMPUSER/60886/dictate/from|ZGljdGF0ZUBmcmVlcGJ4Lm9yZw==
/AMPUSER/60886/language|
/AMPUSER/60886/queues/qnostate|usestate
/AMPUSER/60886/answermode|disabled
/AMPUSER/60886/device|60886
/AMPUSER/60886/answermode|disabled
/AMPUSER/60886/answermode|disabled
/AMPUSER/60886/answermode|disabled
/AMPUSER/60886/answermode|disabled
/AMPUSER/60886/hint|SIP/60886&Custom:DND60886,CustomPresence:60886
/AMPUSER/60886/accountcode|60050but when i type in Asterisk CLI> database show AMPUSER/60886
I see only one line:/AMPUSER/60886/accountcode : 60050
1 results found.For all other extension I see full configuration after typing database show AMPUSER/XXXXX. Only this one has a problem. Deleting and Creating this number again doesn’t help. The same situation. So I suppose Asterisk can’t see /AMPUSER/60886/outboundcid|xxxxxxxxxx and does not present this number outside. Call logs suggest the same.
Only difference I see in configuration for this extension is that “Internal Automatic Answer” (I translate from polish) isn’t set neither ‘Off’ nor ‘Intercom’. In all others I have ‘Off’. Here I can’t change this. After Submit and Reload configuration it stays the same.
Anyone can help? Any suggestions?
Posts: 1
Participants: 1
@josephchrz wrote:
Hello i was curious if freepbx voicemail to text. I saw some places online saying there there was a voice mail to text email. But i was curious if that is still available, Or if there was a voicemail to text of somekind?
Joseph
Posts: 1
Participants: 1
@josephchrz wrote:
Hello i have two types of Extentions from 700 to 710 and from 715 to 720. I’m searching online but cominng u with nothing on howto stop 700 to 710 contacting 715 to 720. Is there a way to do this?
Joseph
Posts: 1
Participants: 1
@mrjoe wrote:
I’m using 15.0.16.42 with Asterisk 16.6.2
I get the above error when trying to “Apply Config”
When I run that I get the following error: [Exception (404)]
Unable to locate the FreePBX BMO Class 'Pjsip’A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma install pjsip 2) fwconsole ma enable pjsipWhen I run the first one I get the following error: Unable to install module pjsip:
- Cannot find module
When trying to run the second I get: The following error(s) occured:
- Module pjsip cannot be enabled
Does anyone know what could be the problem?
It started after I updated the Modules.
Posts: 5
Participants: 2
@tensi0n519 wrote:
I’ve just set up SIPSTATION on my freepbx system and I am able to send and receive calls. I don’t have any prefixes setup and I am able to just dial the number which ultimately connects whoever I’m calling; in this case my cell phone. The issue I am having is that after I dial my outgoing number (my cell phone), I just hear silence. There is no tone as the phone connects to the outside, just silence. Hopefully what I am saying makes sense. Thank you.
Posts: 24
Participants: 4
@mvogel4949 wrote:
Is everyone using pjsip for extensions now? I’m thinking of making the switch but wondering if there are any pitfalls I’ll run into. SIP so far has been pretty plug and play but if it isn’t being supported than I suppose it’s best to switch. Not sure if it matters but I’m using FreePBX14 with Asterisk 13 on my systems.
Posts: 3
Participants: 3
@Philipp971 wrote:
Hi there!
A colleague (Softphone) told me that he was on a call with an external colleague and his cell phone.
After this call, the external colleague said that he received a message from a customer who was WITH IN this call as a third person but the other ones does not heard him and they didnt recognized him at all… The softphone does not showed up a third caller or showed up a initiated call to this customer…I absolutely dont know where to begin with and i really want to ask you if this situation is possilble at all?! When a soft- or hardphone DID NOT initiate a call and send it as request to the PBX, how the hell did a PBX automatically call a customer WITHIN the same call as they two are?!
I really think that the colleague with the cell phone started a conference or something… BECAUSE, according to the employee, it was THE SAME customer they spoke about… i mean there are coincidences in the world but THIS… is very strange.
Is this a possible case at all for a PBX that it start calls without any info?!
Thanks for help!
Philipp
Posts: 2
Participants: 2
@tippetts wrote:
Good day,
We have quite a few of the polycom vvx311 phones. I recently noticed there is a much newer firmware for these phone ver 6.2. We are currently using version 5.5 on the vvx 311. We currently have version 5.4 in slot 1 and ver 5.5 in slot two with fimware in slot 2 being applied.
What is the process for making the 6.2 version available in the Endpoint Manager so that I can drag the new version in either slot 1 or 2? Do I just upload the polycom ld files into the specific tftpboot\polycom\1 or 2 folder?
Here is what I currently see in the EPM.
Available Firmwares
- 0.00
- 0.01
- 1.11
- 1.07
- 1.08
Firmware Slot 1
Firmware successfully installed.
- 1.09
Non-Lync/Lync Compatible (VVX) Version
IP301 3.1.8
IP501 3.1.8
IP601 3.1.8
IP4000 3.1.8
IP320 3.3.5
IP330 3.3.5
IP321 4.0.9
IP331 4.0.9
IP335 4.0.9
IP430 3.2.7
IP450 4.0.9
IP550 4.0.9
IP560 4.0.9
IP650 4.0.9
IP670 4.0.9
IP5000 4.0.9
IP6000 4.0.9
IP7000 4.0.9
VV300 5.4.0.5841
VVX310 5.4.0.5841
VVX400 5.4.0.5841
VVX410 5.4.0.5841
VVX500 5.4.0.5841
VVX600 5.4.0.5841
VVX1500 5.4.0.5841Firmware Slot 2
Firmware successfully installed.
- 1.10
IP301 3.1.8
IP501 3.1.8
IP601 3.1.8
IP4000 3.1.8
IP320 3.3.5
IP330 3.3.5
IP321 4.1.1AA
IP331 4.1.1AA
IP335 4.1.1AA
IP430 3.2.7
IP450 4.1.1AA
IP550 4.1.1AA
IP560 4.1.1AA
IP650 4.1.1AA
IP670 4.0.11
IP5000 4.1.1AA
IP6000 4.0.11
IP7000 4.0.11
VV300 5.5.0
VVX310 5.5.0
VVX400 5.5.0
VVX410 5.5.0
VVX500 5.5.0
VVX600 5.5.0
VVX1500 5.5.0
Posts: 4
Participants: 3
@jcadman wrote:
I have setup paging so that I can page all devices however when I call the page on my Mitel 5330, the grand stream IP phones I have ring normal and do not page.
Posts: 1
Participants: 1
@samdland wrote:
Hi, I was tasked with installing freepbx and configuring it to work with voxbeam. The ISO made short work of the install, but I have no ide about the other stuff. Can someone please help me? What steps are required to configure the Voxbeam stuff?
Posts: 2
Participants: 2
@assos40 wrote:
Hi,
Using FreePBX 14.0.13.23
I am trying to add a new system recording.
The filename is ‘+2105897854’ but a get an error on submiting
“Undefined index: playback
File:/var/www/html/admin/modules/recordings/Recordings.class.php:330”
If i remove the ‘+’ sign then everything is working properly.
I need the ‘+’ sign in front of the filename.
Any idea how to fix this.
Thank you
Posts: 11
Participants: 3
@cal wrote:
Hi all,
I am running asterisk 13.29.2 and Current PBX Version:14.0.13.24.
I have it so when a call comes through it goes to a ring group which the clients would normally put on park.
I currently have it set so ring group MOH is set to ring and the parked calls should have a stream going. Ive double and triple checked the stream which is fine. i even set it to default and still no music.
i saw in another thread that parking uses the same channel as where the call came from i.e ring group. Is this my problem? Do i have to stream the music through the ring group first in order to get MOH in parking?
The clients want the ring groups to have ringing though not music which is a dilema.
No good a custom configurations so if anyone has a solution would be so greatful.
Im new to Freepbx and asteriskThankyou in advance
Posts: 6
Participants: 4
@Philipp971 wrote:
Hi there!
So - i’am looking for a good and simple Wallboard to show some details and infos from the PBX within the browser. Something like: https://www.fop2.com/index.php
So - there is no official installation guide for the FOP2 (or i dont found it yet) and i want to know what Info/Wallboards you are using? Did you got any experience with FOP2? Any recommended alternative?
Thanks for any help!
Philipp
Posts: 6
Participants: 3
@maxyca wrote:
Hi guys!
How to make the system turn off (poweroff) and not into system halt? I need this to turn off the server completely. And the UPS could turn it on again.
I use apcupsd packet on another server in my LAN. And in Asterisk with follow config:
## apcupsd.conf v1.1 ##
UPSCABLE ether
UPSTYPE net
LOCKFILE /var/lock
DEVICE 192.168.1.1:3551
UPSCLASS standalone
UPSMODE disable
POLLTIME 10
Posts: 1
Participants: 1
@microchipmatt wrote:
Hello everyone, I know the hot debate that goes on here about VLAN’s, Free PBX and QoS. I am more leaning towards, I don’t need more VLANS with my new VoIP setup but want to explain my scenario and get confirmation on my thought process with this.
We are a large K-12 environment that spans 3 large geographical areas. We have 30 sites, and staff per site ranges from about 5 - 60 per site. All staff will get a VoIP phone. We are blessed to have fibre on a government network between all of these schools that we operate our VLANS on. We have one flat named VLAN, with routing between all sites to flow traffic. On all the Edge devices, QoS is on and DSCP 46 and 26 traffic (SIP and RTP) are designated as Priority 3 and 5, accordingly, with DSCP 46 being Priority 5 Expedited Forwarding.
We Use Grandstream phones that are already set to mark RTP traffic as DSCP 46, and SIP traffic as DCSP 26, for layer 3 routing which all of our edge devices support as mentioned above. So all of our traffic in the current our current network is prioritized using DSCP, to ensure QoS. Since QoS doesn’t kick in until the shared interface for the edge devices is congested, and that interface and bandwidth is shared for all VLAN’s I don’t see the need to VLAN or QoS any more since A. another VLAN wouldn’t help, the traffic is already segmented via VLANs and Subnets, and B QoS is on…I don’t see any advantage at this point since QoS is already implemented and so are any needed VLAN’s. In my eyes, if the edge interface becomes congested for a site, a VLAN is not going to help, and QoS will prioritize the traffic, in the same way, no matter what VLAN is implemented, they are separate entities. QoS and CoS work the way they work with the DSCP protocol, and priority 1 - 7. In my eyes, my thinking is sane and clear on this. Do I seem sane, does my logic seem correct. I will also say to date our calls have been crystal clear, with this scheme. We are about to expand voip now to more sites.
Posts: 1
Participants: 1
@handleric wrote:
Hello,
I have Fail2Ban configured through the sysadmin module to block for 15 days and send me an email for each block event. Since doing so i’ve gotten a notification that it’s blocked the same IP about every 3.5 minutes since activating it.
The core observation is that when going back to Fail2Ban in the FreePBX GUI, it shows “No Banned IP’s” and obviously if I continue to receive these block events it tells me the brute force is somehow ongoing.
Posts: 1
Participants: 1