Best posts made by BraswellJay

@black3dynamite said in Fedora install doesn't see SATA drives ...:

Try turning off SecureBoot.

Thanks. I checked and this was off.

I finally booted into gparted and it was throwing some errors that seemed to be because there was no partition table on any of the 4 drives. I added an unformatted partition on each and then then once I rebooted the drives were found by the fedora installer. I presume it would have also been found by the Centos installer at that point as well but I didn't actually try it.

We recently switched to freepbx with yealink phones and voip service and my users are saying that sometimes they hear a severe echo on incoming calls to the point that they can't understand the person on the far end. This is only happening very rarely, maybe 1 call every day or two but it has been reported by 3 different users so I don't think it is isolated to just one person or extension. I've got a 50M fiber circuit at this location that has no capacity issues at the moment so I don't expect bandwidth to be an issue

Is this likely coming just from the other party to the call? At first I thought maybe a cell phone caller with bad connection but I would think that symptoms in that case would be a garbled or stuttered type call but not necessarily echo.

Is there anything I can control related to echo when my piece of the call is all IP from phone to service provider? I thought echo was caused by impedance mismatches on copper lines and trunks but I don't have any such link in the call, at least not that I have any control over.

Does anyone have any recommendations on video conference equipment that integrates with MS Teams. We've got 1 big conference room (seats 15) and a few smaller ones that seat 5-8. I was starting to look at some equipment to handle the audio/video part but just wanted to see if anyone had any experience with a system such as this with MS Teams.

Thanks

@JasGot said in Email phishing attempt against one of our vendors was successful ...:

@BraswellJay said in Email phishing attempt against one of our vendors was successful ...:

Subsequently and on the same day, the vendor received another email that he thought was from one of our accountants directing him to ACH to a different (bogus) account.

What makes me also think it was a directed phish attack on your vendor, is that you say the vendor received an e-mail regarding another ACH account number on the same day, but you didn't say the message had any indication it was a follow up or correction to the earlier message.

Thanks everyone for the feedback. It does appear it was on the vendor end but it was a more sophisticated attack that did involve us being fooled as well even though the target was our vendor. From our investigation this is what we believe actually happened:

• Vendor owed us and was going to pay by ACH and requested details. These details were sent to him by our head of finance in an encrypted email which the vendor did receive.
• The attacker then spoofed our accounting team by sending us a phishing email that appeared to come from the vendor (the domain name used against us left an "s" off of the end of the domain name, thus appeared valid to our accounting team) stating that he had not received the ACH info (which the vendor had, this was the attacker phishing us). One of our accountants responded (to the wrong domain) once again giving the correct ACH details.
• At this point the attacker had all he needed to spoof an email that appeared to come from the accountant that had responded to him. The attacker used that info to send a phishing attack email to the vendor which appeared to come from our accountant but using the wrong domain name and contained the attackers ACH info.
• Vendor was fooled by this email and sent payment to the wrong account.
• Vendor ignored (for some reason, don't know why) the fact that when he went to ACH the money the company name appearing on his bank portal as the destination for the payment was not our company name.

One other detail is that both of the spoofed domains that were used in the attack were registered through google on the same day approximately 4 weeks ago which would suggest they were anticipating being able to use us and the vendor in a coordinated attack.

Re: 8 port FXO gateway needed

I have a site where I need an 8 port FXO gateway. I had been leaning toward the Grandstream gxw4108 but based on the linked thread above from a few years ago that seems maybe to be a low call quality device so I'm hesitant to try that now.

I see that Sangoma has an updated model, the Vega 60. Has anyone had any experience with this device with FreePBX? How was call quality?

This site currently has an old Nortel BCM but Centurylink just advised me that they are terminating our support agreement for that system at the end of April due to it's age.

This site is in a rural area and my only access is dual T1s so I don't think ditching the POTS lines and going SIP is an option.

Thanks!

@scottalanmiller said in If you are new drop in say hello and introduce yourself please!:

Welcome @BraswellJay

Thanks! I have lurked here for a while and learned a great deal from the posters here so thanks to all who have contributed.

I'm setting up a FreePBX instance and doing some testing. Right now I have two extensions, one a SIP extension using Zoiper softphone and the other a virtual extension with voicemail enabled, just for something to test with.

I seem to be able to make a call from the softphone to the virtual extension with no problem in that it seems to connect each time but the audio to hear the voicemail greeting is not received most of the time.

I've broke out wireshark to see if I could determine anything. One thing I noticed is that when my Zoiper softphone first registers with the server it seems to receive a 401 Unauthorized from the server before successfully registering. Is it normal to receive receive 401 Unauthorized before a successful registration? .248 is the server and .139 is the softphone.

Source,Destination,Protocol,Length,Info
192.168.2.139,192.168.2.248,SIP,619,Request: REGISTER sip:192.168.2.248;transport=UDP  (1 binding) | 
192.168.2.248,192.168.2.139,SIP,604,Status: 401 Unauthorized | 
192.168.2.139,192.168.2.248,SIP,787,Request: REGISTER sip:192.168.2.248;transport=UDP  (1 binding) | 
192.168.2.248,192.168.2.139,SIP,683,Request: OPTIONS sip:[email protected]:56778;rinstance=b05461348c8fdb80;transport=UDP | 
192.168.2.248,192.168.2.139,SIP,661,Status: 200 OK  (1 binding) | 
192.168.2.139,192.168.2.248,SIP,712,Status: 200 OK | 
192.168.2.139,192.168.2.248,SIP,784,Request: REGISTER sip:192.168.2.248;transport=UDP  (remove 1 binding) | 
192.168.2.248,192.168.2.139,SIP,616,Status: 401 Unauthorized | 
192.168.2.139,192.168.2.248,SIP,784,Request: REGISTER sip:192.168.2.248;transport=UDP  (remove 1 binding) | 
192.168.2.248,192.168.2.139,SIP,567,Status: 200 OK  (0 bindings) | 
192.168.2.139,192.168.2.248,SIP,619,Request: REGISTER sip:192.168.2.248;transport=UDP  (1 binding) | 
192.168.2.248,192.168.2.139,SIP,604,Status: 401 Unauthorized | 
192.168.2.139,192.168.2.248,SIP,787,Request: REGISTER sip:192.168.2.248;transport=UDP  (1 binding) | 
192.168.2.248,192.168.2.139,SIP,683,Request: OPTIONS sip:[email protected]:56778;rinstance=3b911898cd7a2274;transport=UDP | 
192.168.2.248,192.168.2.139,SIP,661,Status: 200 OK  (1 binding) | 
192.168.2.139,192.168.2.248,SIP,712,Status: 200 OK | 
192.168.2.139,192.168.2.248,UDP,46,56778  >  5060 Len=4

I notice also that if I capture then the softphone is constantly sending RTP packets to the server but the server is not sending any RTP packets back to the softphone, which I presume is why I'm not hearing the audio of the voicemail greeting from the server.

@DustinB3403 said in FreePBX / Random loss of audio ...:

Audio issues tend to be due to codec issues.

Thanks, you were right it was codec issue which I hadn't even considered. I changed zoiper to exclude some of the one's it was saying were available and have had zero issues in about 15 test calls since then.

@scottalanmiller said in FreePBX / Random loss of audio ...:

The codec you probably want on both is Opus. Since you paid for that in your Gold package on Zoiper.

I didn't pay for Zoiper. I downloaded the client from their site and when it starts it offers me to upgrade to Pro or something like that but I just choose the option that says continue with a free account. This is Zoiper5. I did just install it yesterday right before I started testing so perhaps they are letting me use that as a trial but will go away? Not sure, but I haven't paid for it for sure.

I have several ubiquiti AP's and one ER4 that I've bought from Amazon.

I'm looking now at an Edgeswitch 48 and it looks to me like the best price is just to buy from Ubiquiti direct. Does anyone have any other source for ubiquiti equipment?

Thanks.

I've submitted a help request to Sangoma but they haven't got back to me yet and I thought someone may know the answer to this question.

I purchased the endpoint manager and sysadmin pro modules for our FreePBX installation. I tried doing through the upgrade on our server but it kept failing so I logged into the Sangoma portal and purchased from there, which appeared to be successful. I can look on the portal and see that the licenses have been assigned to the correct deployment ID. I figured they would email me an activation code for the modules or something but I haven't received it.

Is there a way to make our installation see that those modules have been purchased for it's deployment ID? Does the FreePBX instance contact a Sangoma server to authenticate the purchase or do I need to input an activation code somewhere?

Thanks

What's the best practice / recommended way to configure call park keys on a yealink t42s on freePBX?

I initially created a dsskey with a type of "call park" and value of the parking lot extension as a way to park a call and then had a few BLF keys to pickup parked calls. I had a lot of weird behavior under this configuration though. It seemed like a single extension could only park if the lot was empty. It never seemed to be able to park a second call into the next available slot. The display on the yealink would read "a call is already parked in that slot" if I tried to park a second call.

Eventually I stumbled on changing the dsskey from type of "call park" to transfer with a value of the parking lot extension. This worked as I wanted even with multiple pakred calls with the exception that the parking lot slot was not announced to the person putting the call on hold.

I was able to have the parking lot slot announced by going to the t42s configuration page Features->Transfer and changing "Transfer Mode via Dsskey" from blind to attended.

While this is the behavior I want out of the system, unfortunately the FreePBX endpoint manager overwrites the "transfer mode via dsskey" value and sets it back to blind when a configuration is updated.

What is the recommended way to configure call park functionality? Is using the dsskey transfer type functionality the best way to implement or was I doing something wrong with using the "call park" type dsskey?

I have defined a CID Name prefix ("RG :") on our primary ring group so that members can tell when a call came from the ring group or from someone dialing their extension. That part is working fine.

If a member of the ring group answers a call and then subsequently transfers the call to another extension, the CID name prefix is sticking with the transferred call and being displayed on the display of the extension that the call was transferred to.

Is there any way to remove the name prefix once any member of the ring group has picked up the call so that future transfers will not display the prefix?

I thought maybe I could do this by using the Change External CID Configuration setting but I've not been able to make this work, it still preserves the prefix.

@DustinB3403 said in FreePBX / Queue Login/out state on BLF key ...:

Looks like your question was answered here from last year.

https://community.freepbx.org/t/solved-queue-blf-hints/49665

Thanks, this did it.

I had put an extra * between the toggle code and the extension. Which interestingly enough still worked to allow an agent to login/out but didn't work with the BLF hints apparently. Once I changed as per the link you referenced it worked as expected.

For those that are interested I changed the BLF value in the EPM definition to :

*45__line1Ext__*730

(no * between *45 and the extension number):

Now a core show hints displays :

*45250*[email protected]: Custom:QUEUE250*730   State:Idle            Presence:not_set         Watchers  1

So I got the go ahead to switch our centurylink PRI over to a SIP solution. I had been testing with voip.ms and everything has been really good with their service and that was who I intended to go with. My manager though asked about continuing with Centurylink as the provider so I told him I would get quote expecting it to not be very competitive price wise. To my surprise though the quote is competitive.

To be sure one thing I don't like is that it is not metered in that it is a flat fee for a fixed number of concurrent calls but based on my usage calculations the cost will be in line with what I would have expected on metered service at voip.ms.

One thing that does concern me though is that their doesn't appear to be a lot of information on setting up FreePBX with Centurylink SIP and that which I have found leads me to believe it is not as straightforward as some other providers.

See:

https://wiki.freepbx.org/display/FPG/Setup+Centurylink+SIP+IQ+trunks ; and
https://www.savelono.com/linux/how-to-setup-a-centurylink-iq-sip-trunk-for-asterisk.html

Has anyone successfully used Centurylink IQ SIP service with FreePBX?

Go into your @Skyetel endpoint setting and make sure it is talking to your PBX on port 5160 also.

This seems likely to be my issue then. I had left it at 5060 in the skyetel portal as you showed above and initially had it port forwarded to 5060 on the FreePBX. When I realized that I had set port 5060 up as the chan_sip port instead of the pjsip I changed the firewall to forward 5060 to 5160 on the FreePBX. I'll try making the change in the Skyetel portal directly to 5160 and just direct port forward 5160 in the morning.

As @JaredBusch figured, this fixed my issue. Once I made this change I had two way audio and no call's being taken down prematurely.

Does anyone have recommendation on a good network switch to use that will be located in a harsh environment. The location will be extremely dusty and very hot especially during summer months. Plan to put in a NEMA enclosure that will have little air flow inside.

Minimum of 12 RJ45 ports with POE capability and 1 SFP port for link back to server room but 16 or 24 RJ45 POE would be better for expected future needs.

Thanks.

We're installing an inventory management system and the vendor is using a RHEL server. They have installed a Samba share to put client installation files to install on windows machines.

I'm not having any issues on a Windows 7 machine but the Windows 10 machines are not able to see the share. It does not allow access. I'm thinking this is some kind of SMB version issue but I'm not familiar with Samba shares from Linux so not 100% sure. I did enable SMB v1 on the windows 10 clients thinking that would be necessary but that did not have any effect.

Does anyone know of anything I could check on the Windows clients so that I can see the share?

Thanks

We're looking at adding an intercom solution to facilitate conversation between our operators on one side and on the other side truck drivers that are waiting on a truck scale while product is being loaded. I can work with a direct point to point intercom or we do have a FreePBX phone system and the operations room already has a phone in that location that could be used for that side of the link.

The concern I have though is that the speaker/microphone on the truck scale side will be in a fairly noisy environment since even idling those trucks generate a fair amount of noise. The proposed mounting position for the equipment on the truck scale will be a good 3-4 feet from the driver as they are sitting in the cab of their trucks and I'm concerned that may be too far away for them to hear or for the microphone to pick up what they are saying.

Has anyone worked with a vendor that makes an intercom speaker/microphone solution that would work well in a noisy environment such as this? I would prefer a SIP solution but would consider other if that's what is needed.

I've been playing with a fanvil phone and the only thing I can't seem to make work like I want is transfer to voicemail with the dsskey.

On our system the transfer to voicemail code is #<ext>. This seems to be working fine if I directly key in the extension. However, what I'd like to do is be able to in place of dialing the extension, is be able to hit the dsskey of the extension instead. So #<hitDSSkey>. When I do it this way though I simply get a straight blind transfer to the extension but not direct to voicemail.

Here is the configuration of the key on the phone :

The fourth column (BLF/BXFER) I thought maybe I could change this to BLF/DTMF but that didn't seem to have any affect.

I know Fanvil's are kind of new but has anyone who has tried one know how to program the transfer to voicemail on the dsskey's?

Thanks

Has anyone used a good system for managing the look of email signatures?

We use o365 for email with a mix of users on the web app and the local outlook client install. We've had a standardized signature for a while now but it was just something we taught users how to do for themselves. The bosses are wanting to have a solution where it can be centrally managed now.

I found this from just googling but don't know if it is any good :

https://www.exclaimer.com/email-signature-software

Thanks