@claloano wrote:
Inviting someone to the conference if the voice mailbox
something unpleasant happens ...
in practice remains hooked the call forever!
how you could be remedied because not happen again?
Posts: 2
Participants: 2
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Inviting someone to the conference if the voice mailbox
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MagicJack and FreePBX?
@josephchrz wrote:
Hello we have 3 local analong lines all work on the FreePBX using the samgoma A200 card with FXO modules Finally got it setup to where it is working. However my boss the Ower wanted to setup a forth line But Verizon wanted to have a forth line just for his phone Ext. However Verizon wanted to much money. So he has a Magic Jack he brought from home I try to set it up and Call into it and i get a error message saying the Number you have dialed is a non working number. Which is strange. So i unplugged it and just hooked up a analog phone straight t the magic jack and it work i was able to call to the magic jack from my cellphone and call from the magic jack to my cellphone with no problem. I don't see a mention of this kind of error before. Really need help can someone please help me? Thank you.
Oh i forgot we are using the forth analog port on the card also setup a new trunk for it as well as inbound and outbound for it.
Joseph
Posts: 10
Participants: 5
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Action: Redirect AND Action: ConfbridgeList BUG
@claloano wrote:
I think I found a bug ...
13.0.120 FreePBX and Asterisk 13.7.0
When it becomes a call in place in the conference ... so:
Action: Redirect
Channel: **** ...... .......
ExtraChannel: **** ...... .......
Context: from-internal
Exten: *****
ExtraExten: *****
Priority: 1Everything works like a beauty but ......
Here's what you see by monitoring the conference:
Action: ConfbridgeList
Conference: 800Response: Success
EventList: start
Message: Confbridge user list will followEvent: ConfbridgeList
Conference: 800
CallerIDNum: 11
CallerIDName: Caludio-TH
Channel: SIP / 10-0000000d
Admin: No
MarkedUser: No
WaitMarked: No
EndMarked: No
Waiting: No
Muted: No
AnsweredTime: 0Event: ConfbridgeList
Conference: 800
CallerIDNum: 11
CallerIDName: Caludio-TH
Channel: SIP / 11-0000000e
Admin: No
MarkedUser: No
WaitMarked: No
EndMarked: No
Waiting: No
Muted: No
AnsweredTime: 844Event: ConfbridgeList
Conference: 800
CallerIDNum: 3349472209
CallerIDName:
Channel: Local / 800 @ from-internal-00000005; 2
Admin: No
MarkedUser: No
WaitMarked: No
EndMarked: No
Waiting: No
Muted: No
AnsweredTime: 22Event: ConfbridgeListComplete
EventList: Complete
ListItems: 3As we see clearly the channel 10 is indicated by ID 11 that gives me problems later in monitoring software that it implements in the course ...
For now I've made up, but it would be solved
"Or maybe you do not use something well ???"
Posts: 1
Participants: 1
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Need help with Landline analog hunt group
@josephchrz wrote:
Hello i was wondering something. In my work we use a RCA system that has a rca box with 8 analog lines Port but only 4 are being used. The analog lines are from Verizon. I'm not sure how to the hunt group works or not. But from what i can tell is that. If the first line is busy it rings to the next and so forth. So I'm trying to repeat that process if one line coming in from the analog trunk it will ring to the next one and so forth. Right now i have it setup one of the lines it rings to one group coming in and then they transfer it to a second group and so forth. I have it setup where there is 3 groups. But not sure how to set it up for the hunt part of the incoming.
I'm using the Sangoma A200 card with 1 module 2 lines. first line is only plugged in port 1.
Posts: 6
Participants: 4
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Asteriskuser missing/needed for CDR Database?
@Ecano wrote:
I installed a new Raspberry FreePBX image.
Then I copied the settings (only the mysql config, not the CDR Database) from a BeagleboneBlack system to the Raspberry system via Backup/Restore. (With the Beaglebone system I started with Freepbx 12, both sytems now have FreePBX 13 and Asterisk 11) .
After starting the Raspberry, the system sent mails "Asterisk Run Dir [/var/run/asterisk] is missing or not writable! Is Asterisk running?"
A fwconsole restart brought the following:
root@pbx2:/etc# fwconsole restart
Asterisk not currently running
Running FreePBX shutdown...Running FreePBX startup...
Taking too long? Customize the chown command, See http://wiki.freepbx.org/display/FOP/FreePBX+Chown+Conf
Setting Permissions...[Exception]
Unable to connect to CDR Database using string:mysql:host=localhost;port=3306;dbname=asteriskcdrdb,asteriskuser,rasp_amprestart [-i|--immediate] [args1] ... [argsN]
The Raspberry database has only a "freepbxuser", but no "asteriskuser"
I removed the CDR Moduels from the system (via Module Admin), same problem.My Questions:
-Who/which module needs the asteriskuser,when the CDR module is not installed any more?
-Where can I change the database user (from asteriskuser to freepbxuser) and the password for the CDR?I fixed the issue while adding the asteriskuser to the database with the rasp_amp password.
But I think that is not the best way.Michael
Posts: 4
Participants: 2
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System Recording and applying to IVR for users
@jmakevich wrote:
Is there any way to enable a not-very-technical user to record a system recording and apply it to an IVR, without giving them full access? Ideally something in the UCP, or a plug-in that would be simpler for them. If not, then how would I limit their permissions in the admin portal? Thanks!
Posts: 3
Participants: 2
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Not seeing Active Directory as an auth option in User Manager
@tthomas292 wrote:
On Ubuntu 14.04.4 with FreePBX 13.0.123, I do not have an option to choose Microsoft Active Directory as an authentication method in User Manager > Settings > Authentication Settings. I do have options to use Asterisk voicemail and FreePBX Internal Directory. I also verified that the Msad.php file does exist in the module's functions.inc folder. Any ideas why this would be?
Posts: 3
Participants: 2
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Declined outbound calls
@rimbalza wrote:
Hi any, a n00b here
I installed the distro last week (Current Asterisk Version: 13.7.1 says the info page.) and configured 2 outgoing trunks.
I have each trunk with dial rule that says "0|X." for one trunk, "9|X." for the other so I can address the one or the other.
Calling between extensions works, outgoing (and incoming fwiw) calls go for the desired route but I have a problem with some numbers: if I call from any IP phone the number 9333xxxxxx I address the second trunk and call an italian mobile (all starts with "3"). It goes ok.
From the same phone I call 902xxxxxxx that is a land line (starts with "0") I get "Declined". Using "0" as prefix addresses the other trunk with same results, no calls to land lines.
I just want to be able to call any number from my extensions.
What should I debug? What am I missing? I followed other thread about outgoing routes but I think I miss something here...
Thanks for any hint
Posts: 4
Participants: 2
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Help with password
@davis132 wrote:
hello help as entering freepbx from CLI, they blocked all web access, where store the SIP password???
A phone is not registered and do not give me the password.
where i can enable the access to the web access only have the root password.. thanks
Posts: 3
Participants: 2
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The problem with the transfer of number to sip-gateway
@siv002 wrote:
Hello.
There are two server with FreePBX. At first when you call through the gateway, the gateway number comes in the form of a string "СÑ,ÐμпР° нÐμнко Ð~Ð³Ð¾Ñ € ÑŒ". On the second server to the gateway comes to an empty string. Both servers are installed recently. On both servers configured sip-trunks and outgoing routes. What could be the reason? And her decision.Sincerely.
Igor Stepanenko.
Posts: 1
Participants: 1
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No fpbxinterfaces in ipv6
@rexp wrote:
Hi,
New install and firewall keeps restarting. The firewall.log says the following:
PHP Notice: Undefined index: fpbxinterfaces in phar:///var/www/html/admin/modules/firewall/hooks/voipfirewalld/firewall.php on line 566
No fpbxinterfaces in ipv6
1464386704: Wall: 'Firewall Rules corrupted! Restarting in 5 seconds
More information available in /tmp/firewall.logip6tables section for fpbxinterfaces:
Chain fpbxinterfaces (1 references)
pkts bytes target prot opt in out source destinationChain fpbxknownreg (0 references)
pkts bytes target prot opt in out source destination
0 0 ACCEPT all -- * * 0.0.0.0/0 0.0.0.0/0 mark match 0x1/0x1
0 0 fpbxsvc-ucp all -- * * 0.0.0.0/0 0.0.0.0/0
0 0 fpbxsvc-zulu all -- * * 0.0.0.0/0 0.0.0.0/0Am I missing something here? Since I don't have an ipv6 interface, can I just disable that check? The output of iptables shows the same thing for that chain.
Here's the output of /tmp/firewall.log
http://sprunge.us/SPdX
Posts: 1
Participants: 1
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Permit and Deny calls from Extension
@shahnaseem wrote:
Hi All, Is there a way to assign multiple permit and deny by IP subnet addresses on a particular extension?
I have two IP sub net
172.16.1.0/255.255.255.0
192.168.1.0/255.255.255.0I need when extension on 172.16.1.x range can call any no. But when it will be on 192.168.1.x call block to any extension as well as outside.
I am using FreePBX 13.0.105
Please help.
Posts: 1
Participants: 1
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TLS for every extension
@konkar wrote:
Hello, it is nice to be here.
I recently installed the latest FreePBX image, to create an internal softphone-based communication system within my office. I use CsipSimple for Android and pjsip extensions on Asterisk by using the FreePBX UI, without any problems. Calls are made and everything work as should be.The problem is that i can't find a way to actually encrypt my communication by using password protected certificates, even after days of researching. The last thing i did, was to follow this link's instructions [ XXXX://iprouteth0.blogspot.com/2013/04/csipsimple-srtp-and-sip-tls-with.html ] , but when i fired up Cain&Abel to pentest my own environment, the conversations could still be recorded.
Please, do you guys could help me to achieve certification encryption to make sure my wifi guests can't hear my office's talkings..?
Thank you for your time.
Posts: 4
Participants: 2
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Mimicing Traditional "First call comes in line 1, second line 2, etc."
@stevensedory wrote:
Hey all, wanted to contribute to the community a bit. We often experience frustrated customers when they move to VoIP and can no longer yell across the office, "Hey Bob, call on line 2 for you!". Impractical, right? But a lot of offices have a certain flow, especially some high pace offices like logistic companies, and being able to have this functionality, instead of having to explicitly transfer a call to an extension, is very important for them.
For a while, we told our customers that it just wasn't possible, but I have recently come up with a pretty good replacement solution. Here are our internal instructional notes on the matter. I hope it benefits someone out there. Take care.
//The following best mimics the traditional analog "line" setup, where calls come into line 1 first, which everyone shares, then line 2, 3, 4, etc., enabling people to shout across an office space "John, please take the call on line 1", for example. MANY NEW CUSTOMERS WILL BE COMING FROM THIS TYPE OF SETUP
-Go to Parking, which will open the "Default Lot"
-Set number of slots to number of "lines" desired
*note the slots/lines start at 71 by default, up to the number of slots. 70 is the lot number/name if you will
-Set the Parking Timeout to 0 to disable it
*you can put a timeout if you want the call to go somewhere after the timeout, either Origin or other Dest.
-Use default Music Class
*be sure it is set to silent by deleting all MoH Default songs, or by having a professional MoH
-Pickup Courtesy Tone, None (it's annoying, it beeps by default when transferred to lot)
-Set alternate destination as preferred, though timeout=0 disables this-Create "normal" BLF keys with address of 71, 72, etc. on your endpoints, labeling them Line 1, Line 2, etc. or whatever
-For Polycom VVX phones, create a softkey for the "Line Hold" button
-On the web config, download/export the "All Configuration (except Device Settings)" file
-Open it in Notepad++ and add the following lines between
feature.enhancedFeatureKeys.enabled="1"
softkey.1.action="70$Trefer$"
softkey.1.enable="1"
softkey.1.insert="3"
softkey.1.label="LINE HOLD"
softkey.1.use.active="1"
*this enables custom softkeys, makes one that blind transfers calls to 70 (the default parking lot), enables the softkey, puts it in the 3rd slot from the left, labels it LINE HOLD, and makes it viewable only during a call, in that order-Now as calls come in, either direct, or via Ring Groups or Queues, whoever answers it can then put the caller on a "line" by pressing the "LINE HOLD" softkey, which will silently put the caller on the first available slot in the lot, and light up that BLF key for others to see. You can then yell across the office, "Call on line 1 for so and so!". To take the call, simply press the lit up BLF key. BOOM!
//THE END
Posts: 1
Participants: 1
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Freepbx Firewall not updating protected signalling port (for responsive firewall) when I updated Asterisk Bind port in Advanced SIP settings
@oreban4u wrote:
I recently setup a freepbx firewall for a client. I changed the default Asterisk bind port to reduce the risk of hacking. I setup the firewall and set the eth0 interface as External and Enabled the Responsive firewall because on occassion client requires remote access via softphone to the PBX.
All seemed fine until just recently when he travelled to a new destination and couldn't connect. I began to investigate (I added my IP to trusted zone so i could get in). I realised that even the sip trunks from External DID providers were unreachable. I determined that the firewall was still referencing port 5060 as the signalling port for Chan_sip even though i had updated the bind port in Freepbx interface. I was able to restore access by adding a custom service using the new bind port and setting it up as part of the "External" zone. This I know is not safe as it is unprotected by the firewall. How do I update the chan_sip port in the Firewall? Any help is much appreciated
Posts: 9
Participants: 2
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BLF Other Voicemail
@oracle_sod wrote:
Hi we use Sangoma S500 phones and I have a question about BLF setup
We have a line created with a voicemail box to catch our main line for out of hours. I have assigned a line button on the phone using the template and set it to BLF and the number to be *98XXXX
When the line has a voicemail the light goes red (as expected) however when the voicemail's have been cleared, the line goes green, is there a way to just make it turn off rather than have a green light ?
Posts: 3
Participants: 2
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Move to a new computer
@josephchrz wrote:
Hello Somehow still can't figure out why. My freepbx server motherboard die out no power to it. I replaced the power supply still nothing i replaced the memory and CPU and nothing. So i finally gave up and moved the hard drive to a new computer but totally different motherboard as well as more memory. I can get the server up and running but the network i can not access to get to the web GUI. Not sure what to do can someone please help me out? I went into the command and looked in the Eth0 all same settings from the old hardware that i setup still there. Not sure what to do.
Posts: 3
Participants: 2
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Making one trunk to report it has been registered in other country
@thebat451 wrote:
Hello,
please don't judge strictly, I am a noob in PBX.
Here is my situation. I have my FreePBX 2.11.0.43 running in one country and would like to make one particular trunk to look like (for VoIP provider) it has been registered in other country. Is this possible to configure? Letting all traffic to my PBX through VPN server does not help since all my trunk registrations go down.
Thank you!
Posts: 2
Participants: 2
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Questions about FreePBX
@jkawecki wrote:
Hello.
I have a few questions about FreePBX, Fax server module.
1. Is Freepbx supports T.38 protocol?
2. Is Freepbx (Fax server) has a fax archive? Where is it?
3. Can I transmission Fax using the phone system to another system user Email Box?Thanks for answer.
Best regards.
Posts: 5
Participants: 5
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Problem registering a sip extension
@radzero wrote:
I have a problem with extension 1008 is not registering and in the log i get this
[2016-06-02 21:53:59] SECURITY[20942] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="2016-06-02T21:53:59.328-0300",Severity="Informational",Service="SIP",EventVersion="1",AccountID="1008",SessionID="0x7fb2c034fce8",LocalAddress="IPV4/UDP/192.168.1.50/5060",RemoteAddress="IPV4/UDP/192.168.40.40/5060",Challenge="1b667289"Could someone give some help with this??
Thanks
Posts: 1
Participants: 1
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