@gitim wrote:

Cisco IP phone comunicator trying to register to freepbx I get this error
Apr 2 15:26:09 pbx in.tftpd[44133]: sending NAK (4, Missing mode) to x.x.x.x

What would be the issue?

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@Interlink wrote:

Hello, I have a freepbx 13 on asterisk 13.

What we are trying to accomplish is to have any inbound calls that exceed the call limit to rollover and play a busy tone. Currently they go back to our sip provider as a failed call and they forward calls to a roll over number (typically our cell phone).

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@Connex wrote:

For some reason Responsive Firewall only works with hard phones and soft phones. But when I log into the UCP and try using the phone module in there, RF will not automatically add it into exclusions like it always does with hard and soft phones. Is RF not designed to add the UCP phone as well? I have to manually add the IP to a trusted zone in order for the UCP phone to work.

Any ideas why?

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@digiteltlc wrote:

I set the inbound route “alert info” field for a distinctive ring actually working ok on route destination (simply an extension - ip phone)
If i set as inbound route destination a Misc Destination dialing a remote pbx extension (via iax trunk) , this last one rings but “alert info” field is ignored , or probably is not passed through a misc destination / trunk
Is there any way to achieve this ??
Thank you

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@3dxfood wrote:

Hi everyone!

I’m using an asterisk 13.18.4 for my voip server with a sip trunk my SP’s IMS server provided.chan_pjsip driver used.
Everything is ok but incoming call.When i received an incoming call i will got an 416 unsupported uri scheme error.It seems that asterisk doesn’t support a tel uri scheme.

Some useful logging here:

freepbxbjCLI>
<— Received SIP request (1223 bytes) from UDP:10.203.253.241:5060 —>
INVITE sip:s@192.168.88.144:5060;line=vrkljaj SIP/2.0
Via: SIP/2.0/UDP 10.203.253.241:5060;branch=z9hG4bKe28415501778241ef3d6-E2PtaN0
To: tel:+8610XXXXXXXX
From: tel:XXXXXXXXXXX;tag=ztesiph9ViWnRNGANbNkoNIquneNAI4-12-20481*gahj.4
Call-ID: 983xdBRFkJbzNRg_wvJd4cl9QewUSsSQaaL3HUU12hdei@zteims
CSeq: 1000 INVITE
Max-Forwards: 65
Contact: sip:10.203.253.241:5060;zte-did=4-12-20481-7842-12-639-65535
P-Called-Party-ID: tel:+8610XXXXXXXX
Supported: 100rel,histinfo,timer
P-Early-Media: supported
P-Asserted-Identity: tel:XXXXXXXXXXX
Accept: application/sdp,
application/isup,
multipart/mixed,
application/dtmf,
application/dtmf-relay
X-ZTE-Cookie: 7zs4rm3;id=dRtEEqImrnB8zj7z3Grl@ssf02jyc.bj.ims.chinaunicom.cn
Allow: INVITE,PRACK,ACK,UPDATE,CANCEL,BYE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY
Privacy: none
Min-SE: 90
Session-Expires: 600;refresher=uac
Content-Type: application/sdp
Content-Length: 257
Content-Disposition: session

v=0
o=- 576095279 2131747974 IN IP4 10.203.253.249
s=-
c=IN IP4 10.203.253.249
t=0 0
m=audio 23530 RTP/AVP 8 0 96
c=IN IP4 10.203.253.249
a=rtpmap:8 PCMA/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:96 telephone-event/8000/1
a=fmtp:96 0-15
a=sendrecv

<— Transmitting SIP response (421 bytes) to UDP:10.203.253.241:5060 —>
SIP/2.0 416 Unsupported URI Scheme
Via: SIP/2.0/UDP 10.203.253.241:5060;received=10.203.253.241;branch=z9hG4bKe28415501778241ef3d6-E2PtaN0
Call-ID: 983xdBRFkJbzNRg_wvJd4cl9QewUSsSQaaL3HUU12hdei@zteims
From: tel:XXXXXXXXXXX;tag=ztesiph9ViWnRNGANbNkoNIquneNAI4-12-20481gahj.4
To: tel:+8610XXXXXXXX;tag=z9hG4bKe28415501778241ef3d6-E2PtaN0
CSeq: 1000 INVITE
Server: FPBX-14.0.1.36(13.18.4)
Content-Length: 0

<— Received SIP request (415 bytes) from UDP:10.203.253.241:5060 —>
ACK sip:s@192.168.88.144:5060;line=vrkljaj SIP/2.0
Via: SIP/2.0/UDP 10.203.253.241:5060;branch=z9hG4bKe28415501778241ef3d6-E2PtaN0
To: tel:+8610XXXXXXXX;tag=z9hG4bKe28415501778241ef3d6-E2PtaN0
From: tel:XXXXXXXXXXX;tag=ztesiph9ViWnRNGANbNkoNIquneNAI4-12-20481gahj.4
Call-ID: 983xdBRFkJbzNRg_wvJd4cl9QewUSsSQaaL3HUU12hdei@zteims
CSeq: 1000 ACK
Max-Forwards: 70
User-Agent: ZTE-SBC
Content-Length: 0

Any solution to fix that?
Thanks everyone!

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@jfinstrom wrote:

Usually have people use the google public DNS servers 8.8.8.8 or 8.8.4.4
Cloudeflare with APNIC has released 1.1.1.1 if you would like a tertiary DNS source or a not google source you can use 1.1.1.1

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@sentinelace wrote:

We are trying to setup a phone system at an office using Hughes net. We may have to fall back to DSL but I was curious if this even works? I have 5 phones. All work,and internet is 45/1 but the voice is delayed. Not choppy but long delay

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@jgiebler wrote:

Has anyone developed an App, Web Code etc that would allow a user to initiate a call from their cell phone that will appear as/call from a Company Number (SIP trunk connected to a FreePBX Server)?

The Ask
What I envision is our users connect to an authenticated web page/app of some sort. When they login the system knows what “cell phone” they are on (based on read-only admin-populated fields) and what number(s) they are allowed to dial out from.

From there, they can enter a number they want to call. When the user presses call, the phone server initiates a call to them and to the other party and bridges them together.

Is this possible? Has it already been done?

The Background
We use Twilio for our SIP trunks. There is an App that someone created using the Twilio API called TwiDial to do this. It essentially does the same as the Google Voice app and creates a Conference Bridge between the Caller and the Callee. Both users receive a call from the phone system and when they answer, the calls are bridged together. The Twilio app could work for us, however, it is like $0.03 per minute and we cannot easily limit either side of the call (choice of company numbers or destination numbers).

Further Information
I understand there is some security implications of triggering calls from a web interface/App. This is why we want to limit the “outbound” call to an “approved cellphone number (read-only)” and require the user authenticate before being allowed to press “dial”

I know that we could set them up with a Softphone on their cellphones, but we currently do not allow any “public” IP phones. Any Softphones in use have to be connected over VPN. This was a security choice to limit attack vectors.

Our users frankly would find it too complicated to connect to VPN and then open a Softphone App. I know it seems trivial but it is not for them.

Thoughts
I suppose we could dedicate a virtual extension to the user that has “follow me” enabled with their cell phone number, and then user something like the below code to trigger a call from that extension.Its a little more management on our side to keep track of, update and manage virtual extensions as users move around, but it could work.

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@jerryriggin wrote:

When creating an inbound route, I need to put multiple, random CIDs to allow only them call the number. Pattern matching won’t work as the numbers are random phone numbers. I don’t seem to be able to make a list of any kind. I saw this question answered, “No solution” in 2010. Is there still no way to do this?

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@matthewljensen wrote:

I just opened an account with Vultr and set up a FreePBX 14 server with them. But I’m not sure what the best way to handle the firewall is. I have the ability of setting a firewall from the vultr side, and I obviously have the FreePBX firewall as well. I will be accessing the server from different locations, and I don’t want to get locked out of the server. Should I leave the vultr firewall open and exclusively use the built-in firewall. If so, how should I configure the built-in firewall? What about the responsive firewall, is this something that also integrates with port 80? I’ve read it’s not safe to leave the web portal open. I have a secure password on the admin web access though, should that be enough to ease my worries? What should I do?

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@vespino wrote:

My client has 4 phone numbers which they want to use as CID:

Prefix empty => CID1
Prefix 0 => CID2
Prefix 1 => CID3
Prefix 2 => CID4

I have set the dail patterns to “.” to be able to call every number.

The first problem that appears is when calling 0xxxxxxxxxx (no prefix, 10 digits, starting with a 0 because all domestic numbers start with a 0) it’s using CID2 and not working because the number is incomplete.

Is it possible to use the prefered prefixes and being able to call all numbers? Or is the easy way out not using 0 as prefix?

I have set up 0 as a prefix on another server, but on that server I have only allowed domestic numbers, so the problem described is not at hand.

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@snhroc wrote:

Hello, I am trying to use the conversion tool to convert the data from an lightwaight raspberry pi raspbx implementation into a normal Freepbx distro on a different server.

I am able to get through much of the process fine, set up the donee machine, set up the donor and it appears to go through all the steps. However right at the end of the process on the destination server, I get a gzip error. See log below. Any ideas??

[/] Donor now sending module ‘contactmanager’, table 'contactmanager_group_entri[-] Donor now sending module ‘contactmanager’, table 'contactmanager_entry_numbe[] Donor now sending module ‘contactmanager’, table 'contactmanager_entry_image[|] Donor now sending module ‘contactmanager’, table 'contactmanager_entry_userm[/] Donor now sending module ‘contactmanager’, table 'contactmanager_entry_xmpps[-] Donor now sending module ‘contactmanager’, table 'contactmanager_entry_email[] Donor now sending module ‘contactmanager’, table 'contactmanager_entry_websi[|] Donor now sending module ‘contactmanager’, table 'contactmanager_entry_websi[|] Donor now sending module ‘pagingpro’, table ‘pagingpro_scheduler_exclusions’[] Donor now sending module ‘sangomacrm’, table ‘sangomarcrm_suitecrm_users’ …[|] Donor now sending module ‘xmpp’, table ‘xmpp_options’ …

Trying to retrieve the backup from the DONOR machine …
Decrypting and extracting …
gzip: stdin: not in gzip format
tar: Child died with signal 13
tar: Error is not recoverable: exiting now
error writing output file
Error!
There was an error restoring the conversion data.
Please retry the conversion
Cleaning up…Done!

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@jairusan wrote:

I know there are other posts on this topic, however, it is not clear to me yet if running yum update / yum install to resolve issues showing on the dashboard it’s safe.

Any feedback will be greatly appreciated.

Example:

Version: FreePBX 13.0.194.5

Missing HTML5 format converters < I will need to install lame for mp3 etc…

Can I just do yum update and yum install lame?

Sincerely,

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@beau wrote:

Received a strange e-mail from my cloud hosting provider (Vultr) today:

Dear Customer,

Recent network security audits have detected some issues on your instances. Please review the following reports and help us to ensure the security of our network:

== Portmapper servers ==
Portmapper is a service usually used with NFS. When this is not properly firewalled, it can be abused to conduct DDOS attacks. We recommend that all portmapper services be behind a firewall, and restricted to only IPs that need to contact them.

For Linux machines, please add firewall rules to block port 111 on both UDP and TCP:

iptables -I INPUT 1 -m tcp -p tcp --dport 111 -j DROP
iptables -I INPUT 1 -m udp -p udp --dport 111 -j DROP

Please see https://blog.cloudflare.com/reflections-on-reflections/ for more information on reflection attacks.

The following IPs have been detected running open portmapper servers:
XXXX - at 2018-04-03 10:23:03

Is this something used by FreePBX? What are your recommendations to correct the issue without breaking anything?

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@toborgps wrote:

Hi everyone!

I recently purchased 2 Cisco 7940’s with the latest SIP firmware already pre-installed. One of these phones in now located in Minnesota and one in Colorado. The one in Minnesota is working fine with Freepbx and was a breeze to setup. The freepbx machine is hosted on a raspberry pi 3 with ethernet in Colorado. The phone in Colorado will not send a ping to the freepbx server located on the local network. The log files show nothing about this phone trying to ping, the other phone is pinging via the external IP address. The local phone just will not connect. Any advice or help is appreciated. I have read a-lot about TFTP being required however I have no idea on how to do this.

I am new to Freepbx as well as Asterisk. Thank you all!

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@sm175595 wrote:

Hello guys,

I have successfully set up several OBI1062 phones with FreePBX and EndPointManager.

If I go to my phones IP address I can’t log in.

I’ve tired

extension : secret
admin : secret
extension : password (under “user manger settings”)
admin : password

nothing works, either i’m totally missing something or idk what.

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@sm175595 wrote:

I just got my server up and running. For the most part things are working.

My question is when I delete an extension (chanSip) it doesn’t actually remove it. I still see it in the end point manager. I cant use that same number when making a new extension. To the point my ObiHai508 server wont auto-provision anymore. Going into user settings and deleting it there again doesn’t help.

My question is this. It seems from other posts that this is a daunting process to delete an extension. If this is true i’m going to have to decide whether to dedicate my time to figure out how to delete and extension or just reinstall FreePBX.

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@sm175595 wrote:

I just got my new phone system up and running. The only thing I can’t get is the Obi508 to work.

It did at one point auto provision. I know because I couldn’t log into it anymore. It just didn’t ring.

Where do I go to submit this problem? or what do I do from here? I already made a post about not being able to log into my phones after the provisioning so I could see the settings.

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@fdickey wrote:

You know how to set up a SIP trunk in FreePBX? You know how to configure FreePBX and get a customer up and running with IVR, extensions, voicemail, etc?

We’d like you to join our fast growing Cloud operations! Sangoma sells a SIP Trunking Solution called SIPStation and Cloud PBXact and we need someone with good interpersonal and technical skills to help onboard our customers.

You will be joining a great team with lots of experience to support you!

Here is a full job description, if you are interested, please contact fdickey@sangoma.com

Onboarding Customer Engineer
Neenah, WI

Full-Time

Reporting to the Onboarding Manager, the Onboarding Customer Engineer will be responsible for completing scheduled network checks, implementations and training appointments. The ideal candidate will have a strong work ethic while maintaining the customer support values which are crucial to Sangoma’s success.

Job Responsibilities:

• Assisting new customers with initial set up of their SIP Station (SIP Trunking) or Cloud PBXact (Hosted PBX) accounts.
• Prompt, consistent attendance.
• Adherence to appointment times (contacting new clients at scheduled dates and times).
• Perform over the phone training on Sangoma features, call flow and devices.
• Complete network checks to confirm compatibility and quality of service.
• Testing of call flow, devices and features to assure configuration is accurate and functioning properly.
• Personal ownership and sense of urgency on every appointment or project.
• Work to meet and exceed metrics to ensure appointment success.

Qualifications:

• Technical knowledge of unified communication platforms and features. Asterisk, FreePBX or PBXact knowledge are strong assets.
• Network knowledge of firewalls, routers and requisite upload/download speeds to optimize the Sangoma services.
• Excellent trouble-shooting skills to address network, device and account issues.
• Strong communication skills and attention to detail.
• Excellent customer service skills.
• Team work – knowing how to listen, share, cooperate and learn together as a team.

Compensation and Benefits:

• Full-time position
• Direct hire
• Paid training
• Generous paid time off
• Medical, dental, and vision coverage
• 401K program
• Team building events
• Employee referral program
• Relocation Assistance

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@JGAN wrote:

As you can probably tell I’m just getting into FreePBX and Asterisk,
After the caller enters the IVR and selects an option I want them to be connected to extension 200 and then a DTMF sent to the extension. I put a macro into extensions_custom.conf:

[macro-senddtmf]
exten => s,1,Dial(SIP/200)
exten => s,2,SendDTMF(*74)

Would I then create a custom destination in FreePBX with the target set as:

macro-senddtmf,s,1

Or is this somehow different then the context FreePBX is expecting as the target?

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