@robbo007 wrote:
I have a Grandstream HL503 with a PSTN line connected to the FXO port. I want it to route incoming calls to an internal extension 300.
Under the Grandstream HL503 I have configured in basic settings:
Unconditional Call Forward to VOIP: my freepxb server and the trunk I have created.The FX0 port is registered.
when calling the PSTN number I get the error:
Posts: 3
Participants: 1
@willow wrote:
Hi
I have successfully updated our PBX from version 12 to the latest version via a restore and have configured the server to run in a VM.
I have 25 Cisco endpoints and 5 remote workers and the old live version 12 server is currently sitting on our LAN in our standard IP range.I intend on moving the endpoints onto the new server and changing from 3 x PSTN lines to 4 SIP trunks and wanted to know the best network solution. I have a linux firewall with multiple zones so i could put the PBX on a dedicated zone and not on the general IP range on a DMZ and use the built in firewall.
I have a TFTP server also on our normal range which i guess ill have to move to the new range?I have read various opinions but wanted to get an exsperienced persons point of view.
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Participants: 1
@hwy419 wrote:
Want to connect an external BRM system to Asterisk and get channel/recording information.
http_custom.conf:
[general]
servername={redacted}
enabled=yes
bindaddr=127.0.0.1
bindport=8080
enable_status=yesari_general_custom.conf:
[general]
enabled = yes
pretty = yes
allowed_origins = *ari_additional_custom.conf
[{redacted}]
type=user
password={redacted}
password_format=plain
read_only=yesOutput of “http show status”:
{redacted}*CLI>http show statusHTTP Server Status:
Prefix:
Server: Asterisk/13.38.1
Server Enabled and Bound to [::]:8088
HTTPS Server Enabled and Bound to [::]:8089Enabled URI’s:
/httpstatus => Asterisk HTTP General Status
/ari/… => Asterisk RESTful API
/ws => Asterisk HTTP WebSocketEnabled Redirects:
None.Any reason it’s not binding to the port I’ve configured in http_custom.conf? Haven’t tried connecting yet but I’m worried now my user isn’t configured either.
Posts: 2
Participants: 1
@sgseidel wrote:
FreePBX 15.0.17.12 / Current Asterisk Version 16.15.1 / all modules up to date
The VPN Server module had been working well as far as I’m aware. I recently added a Yealink SIP T-22P phone to FreePBX and was trying to set up a VPN client for the T-22P extension. Now when I try to do anything in the System Admin==>VPN Server module, I get a pop-up dialog box that looks like this:
Rebuilt VPN certificates
The value for CERT_DAYS_VAL has been changed. To apply this new value, please click on Rebuild button or Cancel button to abort.
Rebuilt CloseWhen I click on the “Rebuilt” [sic] button, the process runs indefinitely. I ran this overnight and this morning it was still running.
Note that the CERT Days Remaining field at the top of the VPN Server module says 3638 days
In the Dashboard there is an error:
!Security Issue!
VPN Cert Days value is Changed
This is a critical issue and should be resolved urgentlyWhen I click on the “Resolve” button in the Dashboard, it takes me to the VPN Server module and
the “Rebuilt VPN certificates” pop-up dialog described above.sudo grep -i vpn freepbx.log
[2021-01-30 22:41:59] [freepbx.INFO]: [NOTIFICATION]-[sysadmin]-[VPN_Cert] - VPN Cert Days value is Changed (CERT_DAYS_VAL has been changed. You have to rebuild the VPN certificates.) [] []
[2021-01-30 23:07:24] [freepbx.INFO]: [NOTIFICATION]-[sysadmin]-[VPN_Cert] - VPN Cert Days value is Changed (CERT_DAYS_VAL has been changed. You have to rebuild the VPN certificates.) [] []
[2021-01-30 23:10:31] [freepbx.INFO]: [NOTIFICATION]-[sysadmin]-[VPN_Cert] - VPN Cert Days value is Changed (CERT_DAYS_VAL has been changed. You have to rebuild the VPN certificates.) [] []
…Anyone available to help progress this issue?
Posts: 1
Participants: 1
@IanJanus wrote:
When I try to go into Module Admin via the GUI it crashes with the following error " Exception
RPM command errored, Delete /dev/shm/yumwrapper/* and try again. Exit code 2 - see FreePBX log for more info."
I have tried yum update and fwconsole ma upgrade all
and all the modules appear to be up to date.
Any idea how I sort this issue?
Thanks
Posts: 4
Participants: 1
@ebony76 wrote:
Hey folks,
I’ve been banging my head for a while here and I typically figure things out, been doing these installs for over a decade (doesn’t make me an expert, just been following directives of smarter folks for that long)
I’m running CentOS 7, Asterisk 16.16.0 & FreePBX 14
I’ve done the prerequisite setup as per wiki, and all the packages, dependencies and software installed without a hitch, Asterisk compiled with no issues.
Asterisk started with ./start_asterisk all good.
It’s when I go to install FreePBX that the issues occur:
./install -n
Assuming you are Database Root
Checking if SELinux is enabled…Its not (good)!
Reading /etc/asterisk/asterisk.conf…Done
Checking if Asterisk is running and we can talk to it as the ‘asterisk’ user…Error!
Error communicating with Asterisk. Ensure that Asterisk is properly installed and running as the asterisk user
Asterisk appears to be running as asterisk
Try starting Asterisk with the ‘./start_asterisk start’ command in this directoryAsterisk is running with the correct ownership
root 30844 0.0 0.0 113420 988 ? S 20:28 0:00 /bin/sh /usr/sbin/safe_asterisk -U asterisk -G asterisk
Any ideas, I’ve been digging but haven’t yet found a fix.
Thanks in advance for your help and all of you that have bravely been manning this forum for many years!
Posts: 17
Participants: 1
@bzaayer wrote:
I have 2 different phone systems that are having random phones, both on-premise and external phones, drop connections. This problem started about 10 months ago when I changed over the extensions from chansip to pjsip. Since then random phones will loose connections. To get them back I will reboot the phone and about half the times that will reconnect the phone. One phone system is Freepbx 13 and the other is 14. I am using Grandstream GXP-2170 phones. Any help would be greatly appreciated. This is becoming a problem.
Brian
Posts: 1
Participants: 1
@GRIFFCOMM wrote:
Hi
I notice there appears to be separate ports for each service in the PBX
System Admin > Port Management
80 : Admin
81 : UCP
82 : REST API for Apps
83 : REST API
84 : ProvisioningAssuming provisioning is using HTTP, is there any reason why the system wont let me amalgamate Admin, UCP and Provisioning to a single port? I also assume (not checked) that the REST APIs are also HTTP based, so these could also use the same port.
What’s the reason behind this?
In a grand scheme its likely external phones will be VPN so this isnt a “huge” issue as only Admin would be required, but the UCP means another port has to be opened for what appears to be no real technical reason other than i cant select the same port to use.
Many Thanks
Posts: 1
Participants: 1
@dan_ce wrote:
Hilariously/tragically, this authority says do it on the PBX
If you’re trying to fix echo problems with Asterisk, it’s probably best to make sure the Linksys box is not trying to do its own echo-cancellation, which should be left to the Asterisk server
Whereas ‘Darcy’ who sounds like he knows what he’s about, says do it on the ATA adaptor
Neither Asterisk or Freeswitch will echo cancel your PAP2T. Echo
cancellation over IP is very problematic, and hardly ever attempted. The PAP2T should be echo cancelling for itself, and they usually do a fairly good job of this.
Posts: 1
Participants: 1
@markcrobinson wrote:
Can I tell the system that it does not need to send any email at all when someone leaves a voicemail?
Posts: 1
Participants: 1
@WeebHiroyuki wrote:
Hi, so I’ve successfully set up FreePBX, configured TLS, and logged in properly, but whenever I dial an extension, take *65 for example, it hangs up after 1 second for some reason.
Posts: 1
Participants: 1
@HackerHarvey wrote:
Hi Folks,
So today everything was working well all phones connect and calling. I reset my root password and re-ran the firewall setup and for some reason, all phones are showing what I believe is an SSL error. I updated the SSL certficate and I still can’t connect. Please see the log below. Apart from that I am stumped as it was all working before.
<131>Feb 1 23:58:26 WEB [2514:2533]: WEB <3+error > 106.445.854:Post msg : [RPLAC:GID_ACC] [], 0x10000, 3, 0, []
<131>Feb 1 23:58:27 WEB [2514:2533]: WEB <3+error > 107.755.500:Send msg : [DONOW:NULL] [app_vpPhone], 0x60d02, 0, 0, , 5000
<131>Feb 1 23:58:27 WEB [2514:2533]: WEB <3+error > 107.756.455:CallDskMsgTimeoutEx [0x60d02] lret[0]
<131>Feb 1 15:58:31 cfg [653.654]: CFG <3+error > get attr can not find item priv.auto_provision.mac_local_cfg_md5 err
<131>Feb 1 23:58:39 WEB [2514:2514]: WEB <3+error > 119.382.085:Send msg : [DONOW:NULL] [app_vpPhone], 0x60d03, 0, 1, , 5000
<131>Feb 1 23:58:39 WEB [2514:2514]: WEB <3+error > 119.382.824:CallDskMsgTimeoutEx [0x60d03] lret[0]
<131>Feb 1 23:58:39 WEB [2514:2533]: WEB <3+error > 119.419.790:Post msg : [COMON:] [], 0x10000, 3, 0, []
<131>Feb 1 23:58:39 WEB [2514:2533]: WEB <3+error > 119.448.167:Post msg : [COMON:] [], 0x10000, 2, 0, []
<131>Feb 1 23:58:43 WEB [2514:2533]: WEB <3+error > 123.582.262:Send msg : [DONOW:NULL] [app_vpPhone], 0x60d02, 0, 0, , 5000
<131>Feb 1 23:58:43 WEB [2514:2533]: WEB <3+error > 123.583.097:CallDskMsgTimeoutEx [0x60d02] lret[0]
<131>Feb 1 23:58:48 WEB [2514:2514]: WEB <3+error > 128.303.169:Send msg : [DONOW:NULL] [app_vpPhone], 0x60d03, 0, 1, , 5000
<131>Feb 1 23:58:48 WEB [2514:2514]: WEB <3+error > 128.303.961:CallDskMsgTimeoutEx [0x60d03] lret[0]
<131>Feb 1 23:58:48 WEB [2514:2533]: WEB <3+error > 128.337.960:Post msg : [RPLAC:GID_ACC] [], 0x10000, 3, 0, []
<131>Feb 1 23:58:48 sua [1824]: NET <3+error > [000] New binding with 46.101.55.136
<131>Feb 1 23:58:49 sua [1824]: NET <3+error > [255] depth=2:/O=Digital Signature Trust Co./CN=DST Root CA X3
<131>Feb 1 23:58:49 sua [1824]: NET <3+error > [255] depth=1:/C=US/O=Let’s Encrypt/CN=R3
<131>Feb 1 23:58:49 WEB [2514:2514]: WEB <3+error > 129.470.826:Send msg : [DONOW:NULL] [app_vpPhone], 0x60d02, 0, 0, , 5000
<131>Feb 1 23:58:49 WEB [2514:2514]: WEB <3+error > 129.471.732:CallDskMsgTimeoutEx [0x60d02] lret[1]
<131>Feb 1 23:58:49 sua [1824]: NET <3+error > [255] depth=0:/CN=sip.**************.co.uk
<131>Feb 1 23:58:50 sua [1824]: NET <3+error > [255] SSL ERROR ZERO RETURN - SHUTDOWN
<131>Feb 1 23:58:50 sua [1824]: NET <3+error > [255] EVP lib in (null) (null)
<131>Feb 1 23:58:50 sua [1824]: DLG <3+error > [255] tls recv message failed, error_code[6]; socket:remote_ip[46.101.55.136], remote_port[56974]
Posts: 2
Participants: 1
@jlizzotte04 wrote:
Silly question that I believe I already know the answer to, but we host many clients, some of whom have many extensions. Often times they change employees. If there any variables we can use on User Management to grab the info from the extension? Example:
Extension 5503 belongs to Fred. Fred gets fired. The HR person asks me to change the name to the new employee. Bob. I have to rename all the pertinent info for his extension. But I also have to duplicate my efforts in the User Management section. But wait, there’s more. I have to also change it in Endpoint manager (I already assumed there was no solution to that too)
Just sounds like it would be beneficial to have a variable that gets edited once, and inserted. Thanks
Posts: 1
Participants: 1
@leorke wrote:
I can’t enter the Operator Panel, got the message: err_connection_refused
I can’t enter the User Control Panel, don’t know the user and password.
My version: 14.0.16.4
Thanks for support
Posts: 4
Participants: 1
@GRIFFCOMM wrote:
Hi
The PBX is located behind a firewall and port 80 and 443 are in use, this means Lets Encrypt wont work on this box, however i need items like ZULU which needs a certificate. I can get certificates, so whats the best way to automate saving the external certificate in to the box?
Most are valid for 60-90 days and i dont want to be uploading a cert every 60 days. Assuming there’s no module i can load, the other way would be the exposé the cert path through a share so i can copy the new PEM and CRT file in OR a scrit inside freePBX that runs to collect the certificate PEM and CRT files…
Any ideas?
Many Thanks
Posts: 4
Participants: 1
@antonis77 wrote:
Hi ,
I know that this topic has been discussed already but i can t find a solution to that.
but i cannot see a video preview in my phones
I am using FreePBX 15 as a SIP server. Both doorphone and GXP phones are registered on the SIP server.
I ve enabled video support on FreepBX.Granstream support says that
In invite you find link for video. FreePBX remove this line so phone have no idea where get video.
Can we do something to solve this ?
Posts: 1
Participants: 1
@netphoneusa wrote:
This issue does not happen on the SPA series phones. It only happens on Yealink
I have 2 ring groups
#1 = Sales Ext 151 152 153 154
#2= Parts Ext 251 252 253 254
My extension is 101.
Call comes in. They want sales. I use my BLF key for the sales group. All 4 extensions start ringing. No one picks up - they are all away from their phone. I then decide to press the same BLF key that is no flashing red as it rings the sales group to try and pick the call back up on my ext 101.
On the Cisco SPA phones - no issues. I can retrieve the call
On the Yealink phones (T29 and T54) nothing happens. It will not pick back up again and let me have the call back so once I send it to that ring group I can’t get it back.
Any ideas what causes this?
Posts: 1
Participants: 1
@incognito wrote:
Hi There,
I am having a problem that is limited to one extension/user. When I call one of my internal extensions then I conference in the problem extension (112). Extension 112 will drop from the call after 30 seconds. If I call extension 112 direct it does not happen. If I call extension 112 and then conference in other extensions it does not happen. It happens consistently if I call another internal extension first and then conference in 112.
Setup:
- Freepbx 14
- PBX is hosted offsite (Vultr)
- All extensions are using PJSIP with Direct Media set to No
- All extensions are using Grandstream GXP 2160 phones
- All phones have NAT traversal set to Keep-Alive
- All extension are remote from the PBX and from different external networks.
- SIP ALG is disabled on all routers. All routers are the same Unifi Security Gateways.
I am not seeing any timeouts in the Asterisk log files but I do see an entry for:
Channel PJSIP/112-0000210f left ‘native_rtp’ basic-bridgeAny help is appreciated. This is just a weird obscure issue that is only happening in a very specific order/scenario.
Thanks
Posts: 1
Participants: 1
@Tragik wrote:
Hello,
I am having some issues with a FreePBX system dropping calls consistently at 32 seconds. I’m running FreePBX 15.0.17.17 with Asterisk 16.11.1.
I’ve seen a lot of other posts with similar issues as mine and it usually comes out to be a NAT issue, but I can’t figure out where my issue is. I am forwarding UDP port 5060 (pjsip not chan_sip) and RTP ports 10000-20000 to the internal IP of the PBX and have firewall rules block it if the IP address isn’t coming from the Twilio IPs I am using for my trunk. If i go to asterisk SIP settings and detect network settings it fills in the external IP with the correct IP. I have all of my internal networks added under asterisk SIP Settings as well. Here are logs for a call that dropped: https://pastebin.com/uM5955cd
Posts: 1
Participants: 1
@mike_b wrote:
I have a question regarding forwarding to an external number. Is it possible to forward a call to an external number, but have the caller leave a message on extension if there is no answer?
So, following scenario: call comes in and the caller selects an extension. The extension is forwarded to an external number, and there is nobody to pick up the call. At the moment, the phone simply keeps ringing. I would like the call to ring the extension a number of times (say 5 times), then switch back to the extension voice mail box so the caller can leave a message there. Possible?
Posts: 2
Participants: 1