Go to Admin>System Admin>Port Management and you’ll see the ports for HTTP provisioning.
In portal redirection, you’ll need to change the ip address / fqdn protocol from tftp to http or https. And then make sure the phone provisioning port is set to match the port setup under Port Management. Then factory reset the phone under web gui or by pressing Menu, then * key 3 times, then press and hold X button (Below the volume) for 10 seconds or so and the phone will restart and reset. You’ll need to have to provisioning port allowed on your edge firewall as well. I’ve seen portal redirection not work if the phones are on old firmware so If it doesn’t work, I would manually update the firmware and retest. They current firmware is a couple years old IIRC.
There’s no code or update that is going to change a configuration directive as important as this one.
I suspect your running an older FreePBX16 server. The rumor mill is that the “customized” version of CentOS used for the FreePBX 16 distro has not gotten the newer CentOS security updates for many months, now. They could have got in via an Apache2 vulnerability.
If it was me I’d assume the PBX was compromised since you obviously have it publicly accessible. PBXes are high value targets. If you (wisely) had your trunk provider block international calling they probably got in, discovered there’s nothing here they couldn’t get from a burner cell phone, and got out then forget to change the port back on the way out. Most people use 5060 and don’t go to the trouble of turning on tls on their phones so a cracker can muck about with the tls port as part of probing a PBX and most PBX admins wouldn’t know it.
With me I don’t put high value targets on the Internet I make the end users access them via VPN. But that’s a personal decision.
I’ve tried calling the phone with diffrent softphone, but i still got the same result.
But as i said, the phone appears as offline to asterisk, so thats why i get forwarded to voicemail every time.
Ths industry has kind of adopted this as the process based on feedback from the FCC and recent fines they have given out. It requires much more than 1 or 2 docs. This doc was put out by the independent Cloud Communications Alliance group and is recommended all trunking and cloud providers follow this for KYC.
There is an obvious gap in the P-series phone line…310, 315, 320, 325, 330…370!
Will there be a P350?
Just asking…because the P330 is not really a good replacement for the Digium D65. This was a great phone, by the way! The display was superior to the P330.
Huh, yea I am not sure why that would be an issue for Asterisk v20. How are you going about upgrading/downgrading between the versions? Are you using the asterisk-version-switch command?
I have a FreePBX instance that is taking care of hosting multiple SIP speakers in a school for paging/intercom. I would like to connect this FreePBX system to an on prem VoIP system the school uses for their normal phones. The simplest way to do this would be to light up a trunk between the systems but that occurs a large license cost on the side of the school district. I have been asked if I can instead register as a SIP ‘station’ to the phone VoIP server.
From some googling, it seems possible to configure a ‘SIP trunk’ to register as a SIP station, but its a little out in the weeds past what I really understand. Anyone able to shed a little light? Thank you!
Still need to know the version number of the FreePBX server you are using.
Did you try the force_rport command in pjsip.endpoint_custom_post.conf I suggested?
Remember to keep a close eye on the “intrusion detection” in the system admin module. On boot the phone will attempt to register once. If there’s any mistake like a bad password then sometimes the intrusion detection will block the phone. If that happens then the phone won’t continue to attempt to register. LATER models of Cisco phones WILL continue to attempt to register - so if Intrusion Detection has blocked them, and you correct the problem and unblock the IP of the phone - they will almost immediately register. But the firmware in this model does not appear to do that - it will just continue displaying the “registering” message forever and not make any attempt to register. You have to power cycle the phone.
Note also that with the latest 9x version of Firmware, Cisco made it impossible to send a remote reboot command to the phone. With earlier versions you could SSH into the phone and issue the reboot command.
You posted your SEPMAC 9 days ago. The following is the SEPMAC file I am using with my 7961G which registers in perfectly to chan_pjsip. My phone has a Cisco 7914 “outrigger” sidecar on it so my SEPMAC has more button definitions in it than a normal 7961 but otherwise it’s similar. The firmware I am using on my phone is:
Please note I have CAREFULLY gone through my SEPMAC and replaced all tabs with spaces and corrected all the indentation errors. I don’t think the phone cares about that but poor indentation in an XML file makes it harder to catch errors.
Some POSSIBLE issues in your SEPMAC file are:
You are defining UTF-8 at the top of yours and then you use the directive WInCharSet iso-8859-1 which is inconsistent. The phone ignores the UTF-8 at the top so it does not matter but I would test with or without the iso-8859-1 to see if it makes the phone change anything at all in it’s display.
Make sure in your tftp server logs that the phone is indeed properly loading the language files for Danish and those files are present in the tftp server
Your date template may not be right - this phone does not support all permutations of date template. You should probably use D/M/YA
If you have an internal NTP server, use that.
You have a duplicate listing of common Profile, one at the end and one in the middle
You have videocapability defined as 1 but this phone is not a videophone so that should be 0
daysdisplaynotactive appears to be ignored by this phone as do all the display parameters
your preferred codec is g711alaw although the phone’s datasheets seem to indicate it only supports g711ulaw although when I was testing either directive worked
you have a public IP in nataddress directive which should be blank unless you are accessing a freepbx server on the internet
Last and most important - when the phone first boots it reads in the SEPMAC file and then writes the parameters to it’s nvram. If you then later make a change in the SEPMAC file that the phone cannot parse or does not like - then the phone will SILENTLY ignore the SEPMAC file and use the nvram settings.
Which means that unless you make a trivial change - add or remove a line button or change the name of Line to Linje or vis-versa, some visible change to the phone’s display - you won’t know for certain if the phone is even loading in any changes in your SEPMAC
I found this very frustrating when testing SEPMACs. It would be far better if the phone simply issued an “error in provisioning file” on the display but it doesn’t.
The top problems in getting a phone to provision:
incorrect username/extension name/number
incorrect password
intrusion detection blocking the phone
Not setting the proper allow IP subnet range in SIP’s configuration on FreePBX
You could do it that way but if you run into problems another way would be to buy 2 gateways, one an FXO gateway and one an FXS gateway, and put the FXS on the school’s system and the FXO gateway on the FreePBX system, then plug them in back to back.
(this post mainly for amusement value to illustrate the futility of the “license every last little component separately” approach)
Latest FreePBX 17.0.19.7, brand new install, and this issue is still persistent.
Very simple to reproduce:
Ext 100 - Make an outbound call, Warm/Attended Transfer the connected party to a Queue
Ext 101 - Answer the call on the queue, speak to Ext 100 for a moment, then Ext 100 completes the transfer.
Ext 101 - Speak with party then end call.
You will get 2 recording files, the original outbound from Ext 100, then the Queue call communication between Ext 100 and Ext 101. The third portion of the call where Ext 101 speaks to the called party is gone.
This is also visible in the CDR.
Trunk and extensions and queue are all set to Force.
Outbound Attended Transfers do not produce recordings past the attended transfer.
Latest FreePBX 17.0.19.7, brand new stock install.
Very simple to reproduce:
Ext 100 - Make an outbound call, Warm/Attended Transfer the connected party to a Queue
Ext 101 - Answer the call on the queue, speak to Ext 100 for a moment, then Ext 100 completes the transfer.
Ext 101 - Speak with party then end call.
You will get 2 recording files, the original outbound from Ext 100, then the Queue call communication between Ext 100 and Ext 101. The third portion of the call where Ext 101 speaks to the called party is gone.
This is also visible in the CDR.
Trunk and extensions and queue are all set to Force.
I have read 2 separate articles and followed their FreePBX configuration instructions, step-by-step but am not able to get the sipML5 tool to register to the FreePBX.
Has anyone ever successfully done this? If so, I would sure appreciate your guidance.
