@nmarques wrote:
Hello,
I have Automatic System and Module Updates set to Enabled, but updates never happen automatically. I get e-mail notifications that there are updates, but they aren’t performed. Running FreePBX 14.0.3.17.
Posts: 1
Participants: 1
@duyle86 wrote:
I have a FreePBX system with a Grandstream GXW4104 using 2 analogue lines (connect from traditional PBX with extension 8495 and 8496). The Grandstream is registered and it works perfectly on outgoing calls.
On incoming calls it receives 1 call from FXO1 and transfers it to Ring group but it doesn’t accept any more calls on the other lines( FXO2,FXO3 and FXO4). The caller gets a ringing tone but no incoming call show on the sip phone. I can make outbound call normally in both 2 line.Chan_Sip PEERS
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
901 (Unspecified) D No No A 0 UNKNOWN
902 (Unspecified) D No No A 0 UNKNOWN
906 (Unspecified) D No No A 0 UNKNOWN
GXWT1 10.18.30.194 Yes Yes 5060 OK (2 ms)
GXWT11 10.18.30.194 Yes Yes 5060 OK (2 ms)
gxw4104 10.18.30.194 Yes Yes 5060 Unmonitored
6 sip peers [Monitored: 2 online, 3 offline Unmonitored: 1 online, 0 offline]
Posts: 2
Participants: 1
@jode wrote:
Hi ALL,
This is my first topic, and English is not my primary language.
- Situation:
I Have 2 Cisco 8861 with 36 lines extensions BEKEM (extensions 100 and 101) and them are used as central phone for the reception.
All the others phones are Cisco SPA 504 and SPA 508 from Cisco with an functional BLF.
The reception needs the BLF on their phones to monitor Extensions before or not forwarding a call to an extension.
The protocol is PJSIP for every phone. Extensions are already created from freePBX (extensions-> pjsip extensions)
I patch my freePBX with the usecallManager patch, and follow information from : wiki freepbx org/ display/ FOP/ Cisco
So I think my Asterisk is patched.
- Problem
I Really does not know how to make the BLF work. Explanation from usecallmanager .nz /sip-conf. html are for Asterisk Only, but I know is not recommended to edit SIP.conf manually on a freePBX Distribution. On other hand, explanation from wiki. freepbx. org/ display /FOP/ Cisco doesn’t work…
I think I have modification to do on extension_custom_post and other files but I’m not so sure of what exactly.
In addition I use PJSIP and not CHAN_SIP so I pretty sure that the configuration is different.
Anyone can help me?
Posts: 1
Participants: 1
@daansk44 wrote:
Hey everyone,
Is it possible to call someone via Freepbx and to play a recorded message. And when the recorded message is finished to be able to speak via the VoIP application of your phone / computer?
Posts: 1
Participants: 1
@antonis77 wrote:
Hi,
I want to have an IVR where the user will press 1,2,3,4,5 e.t.c and will be able to listen timetables for buses, trains, ships that i 've stored in a database.
What will i need to do this (text-to-speech or something similar) , any ideas ?
Thanks
Posts: 1
Participants: 1
@cherodj wrote:
Saludos
tengo una instalación de reciente de freepbx 13.0.195.18
y tengo una tarjeta DAHDI con 3 puertos FXO y 1 FXS.
tengo dos lineas conectadas en el puerto 1 y 2
entran llamadas suena el grupo de llamadas sin problemas puedo realizar las llamadas hacia el mundo sin problemas.estoy probando una troncal SIP de la compañia Netlip
logro que entre la llamada sin problemas pero suenan todos los teléfonos
y me interesa que ese troncal SIP suene solo en una extencion u otro grupo aparte
solamente.soy un poco novato pero e logrado hacer varias cosas. e leído en varios foros pero no logro que funcione como quiero.
les agradezco de ante mano.
Gracias.
Posts: 1
Participants: 1
@dannyprecise wrote:
What is the formula used for determing the wait time relayed to the caller on hold? It is it simply the average wait time for the last 24 hours or is it a more advanced calculation?
Is there a way to play a specific announcement after a caller has been on hold for a while.
Example we have an announcement to play every 1:15 min but if they are on hold for more than lets say 5 min. we want to play a different message apologizing for the long hold.
Posts: 1
Participants: 1
@sentinelace wrote:
I am trying to setup transfer to voicemail. I type in ##* using DTMF then the extension and I get “I am sorry that is not a valid extension”. Then it says enter a valid extension which I do at 110. Then the logs show:
This extension does not exist or no password is set
I am dialing for example: ##*110#
The logs show “Invalid extension 0 entered”
Is this a bug? I don’t see any feature codes causing the issue. Fresh install as well
Posts: 2
Participants: 2
@nabberuk wrote:
After updating modules i now get an error, this error occurs when going to the contact manager, also when trying to save any extension. The error i get is;
Whoops \ Exception \ ErrorException (E_ERROR)
Zend OPcache class loading error, class FreePBX\modules\Contactmanager, function install
I’m a little unsure on how to fix this issue.
Posts: 1
Participants: 1
@Tommaso wrote:
Hello everyone!
There is someone out there that could help me about configuring a SIP trunk in FreePBX in order to connect my FreePBX to a gateway FXO (Yeastar TA410 4FXO ports) ?my current situations is that i can make inbound calls using only one POTS/FXO port and i’m only able to get INbound calls, if i try to do an outbound i get the “line is busy” message
any advices for the “sip settings” of the trunk?
i would say thathost= the IP of my gateway?
username= usr name that i use to acces to my gateway?
secret=psw that i use to acces to my gateway?
type= ??? by default its “peer” but i cant understand what i should put here! i found in the wiki that i could change it to “user” but it’s not working…any advices/ideas?
Posts: 1
Participants: 1
@mooseling wrote:
Hey all,
I’m completely new to voip, but I’m currently the only tech guy at my company, so when anything goes wrong it falls to me.
We have a voip server running FreePBX 14.0.3.2 here in the office, and 10+ phones here on the local network. Mostly Polycom SoundPoint 331s. We also have three employees that work from home, and they each have a Polycom SoundPoint 331 which connect to our server and take part in the whole thing.
Our office changed internet provider a couple days ago, and immediately those remote phones lost connection. I got them back online by having them change the ip addresses of the SIP server and the outbound proxy to point to our new ip address. That makes sense.
However, after the initial jubilation, it became apparent that all calls over these phones are dropping out after about 30 seconds, and now one of them says she isn’t receiving any calls at all. I’ve been reading around, and perhaps the call dropping is due to failing to receive call acknowledgement? But why would the one phone not be receiving calls at all any more?
I’m at my wits end here, especially because I didn’t set this system up and I have a lot on my plate at the moment anyway. So any direction or advice would be welcome!
Many thanks.
edit: I’d love to look at the asterisk log and learn something about this. If I could filter for something in particular, that would be brilliant, but I don’t know what I’m looking for. Any ideas?
Posts: 1
Participants: 1
@msmcknight wrote:
Hi guys,
I’m using FreePBX 13.0.195.4 and am trying to change the sip host for one of my trunks…
I can make the change under TRUNKS --> SIP SETTINGS, and it is reflected there correctly, even after a reload, but no matter what I try, it will not update the /etc/asterisk/sip_registrations.conf file.
The sip_additional.conf file is updated correctly with the new information.
The file is being re-written everytime I reload freepbx, but it’s written with the old hostname, as in “register=username:secret@old-host-name”
I can’t find anywhere else in the GUI where the hostname of this trunk is used, except in the SIP SETTINGS for the trunk, and yet, it’s reflected correctly there.
I’ve even tried deleting the sip_registrations.conf file in the hope it was corrupted, but no, the same file gets recreated with the wrong information.
I’m at my wits end, so if anyone has any suggestions, they would be greatly appreciated.
Thank you,
-Michael
Posts: 1
Participants: 1
@mvogel4949 wrote:
Is there a way to set a destination for a caller that is placed on hold? After 5 minutes ring the phone back instead of just dropping the call?
Posts: 1
Participants: 1
@domosute wrote:
Hi,
I would like to play a little tone to indicate the end of call when the other end hangup the phone.
Is there any easy way to achieve this?Many thanks in advance,
Posts: 1
Participants: 1
@TrinityConcord wrote:
I am running FreePBX 13.0.195.18. Flowroute has new regional servers that can send inbound IP authentication calls from one of 64 different IP addresses, in 4 groups of 16. Flowroute recommended a PJSIP trunk. I have set up my trunk per Flowroutes specifications, yet FreePBX rejects the incoming Invite with 401. Flowroute has reviewed all my settings and does not understand why it does not work. I looked at the Asterisk files generated by FreePBX for my trunk. I specified None for both Authentication and Registration, since it is not a register trunk. However, I was surprised to find these lines:
[Flowroute]
type=auth
auth_type=userpass
password=
username=Flowroutein the generated pjsip.auth_custom.conf file. It should not be an auth trunk. Might that be the problem? Is that a Freepbx bug? Any other ideas what the problem might be?
Posts: 2
Participants: 2
@Tommaso wrote:
Hello everyone! making a blacklist for incoming calls its pretty easy, but i really cant find out how to prevent my extension to be able to call some numbers
lets say that i want that no one can call 199 XXX XXX (in italy those numbers make you to pay sooo many money, this is why i want that no one inside my company can call this numbers)
well, i really cant figure out how to achieve this!
i guess i’ve to set it up in the “dial pattern” for my outbound route, but how? i google d a lot and i saw someone using “_” or “/” but i actually didnt understood if this is the correct way and how to properly use it!from what i’ve understood, another way to block outgoing call to a range of numbers is to create a “dead” trunk and add an outbound routes with dials patters like the above example “199 XXX XXX” so when someone call those numbers they will go on the dead trunk and wont be able to call??
any ideas?
Posts: 6
Participants: 2
@lmtmike wrote:
We have been using a small TalkSwitch on-premise PBX for many years. They were purchased by Fortinet and rebranded over the past few years. I am hoping to transition to the FreePBX solution and use the newer Fortinet FON-550i and FON-460i IP phones that I have invested in. I have gotten the basic connectivity configured by I am trouble with things like BLF and other soft key programming. Does anyone have experience with these phone sets? I may end up buying new phone sets but want to make sure it’s necessary first.
Does anyone have any experience with the Fortinet line of phones and FreePBX?
Posts: 3
Participants: 3
@faisalkhan wrote:
HI all,
Is there any way that we can enable any announcement for the callers who are queued in call waiting.
for example I have enabled a call waiting in my phone system and I want my callers to hear an announcement that the person you are trying to reach is on another call. It’s kind like a telco company.
Is that possible.
Thanks.
Regards,
Faisal
Posts: 5
Participants: 3
@daansk44 wrote:
How can you set a number to the Misc Destinations.
When I set it up for calling my mobile phone. I see ony a private number when I get called
Posts: 1
Participants: 1
@clyde277 wrote:
Hi folks,
Recently I’ve gotten into an annoying problem; under Dashboard -> FreePBX Statistics, I see there is an offline trunk. I used to have two trunks from two different providers previously, for testing purpose, and a while ago I deleted one of them from Connectivity -> Trunks.Although I deleted one of the trunks, FreePBX statistics it’s still showing that there is an offline trunk. I also see this deleted trunk under Reports -> Asterisk Info -> Channels -> Chan_Sip Channel(s).
Note that I can also see this deleted Trunk that is still under Database -> Tables -> Asterisk -> Trunks
Any clue what I should do in order to make sure I get rid of this unnecessary trunk? Maybe someone can tell me where is the trunk config file in the asterisk file structure?
Thanks!
Posts: 1
Participants: 1
