Does anyone know why the caller ID coming in does not populate in the asterisk phonebook like it used to? (prior to upgrading to Freepbx14)
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Asterisk phone book used to cache caller id entries but no more?
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Xlite config converted to FreePBX
Ok thank you for the info.
The below are the details (altered by me) my VSP has instructed me to use however for “type” I don’t see any documentation so I don’t see how this config could ever work!?
user= 043058370
type=user authenticated
Password=XXXXXXX
host=87.52.109.38
fromdomain= sip.freedom1.com
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FreePBX Security Alert (VoIP Server)
hi there…ok i havent done anything different. As shown in the screenshots the cert manager says it is satisfied and looking at the firewall there are entries in that were put in when I installed it. Why would this stop working?
turning off the firewall leaves one vulnerable.you’re talking about reverting a change. I have no idea what change(s) your talking about. I didn’t make any recent changes.
It may seem obvious to you guys but not to me, not at this time.
Leon
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Max number of channels used
Found an old thread from 2008 that no longer apply.
How do I find how many channels was used at one time?
Looking at the statistics in the dashboard the numbers “in use” does not match from the last hour and a day, week or month.
How to I tell or obtain the max number that was used in a period so I can determine if we have enough or too many channels.
old chain say… get that from the CDR but how or what to query?
Thanks
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Dead Air on Random Calls
We had a client today with choppy audio, calls went dead and then reconnected etc. We ran a constant ping from their lan to the internet and also from our office to their firewall, we instantly saw that there’s hiccups.
We asked them to contact their ISP, luckily a modem reboot solved the issue.
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Latest Backup Module not working
Yesterday there were a number of module updates including Backup & Restore v 14.0.3.11.
My backups report backup complete in about 1 second and they create a null file.
I created new backup schedules and drag and drop the templates into the table but they disappear when I save.
I disabled and reinstalled the module to no avail, is this a known issue and should I raise an Issue ticket?
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FreePBX Security Alert (VoIP Server)
Leon,
Certificate Manager is only telling you that the FreePBX firewall is correctly configured to allow the connection inbound. This does not take into account your router/wireless access point/I dont know what.
I have put extra effort into this in trying to get mirror1.freepbx.org to talk to your server directly to get the token and my connection is refused.
You should do what we have stated. Turn off the firewall and then attempt to update the certificate. I am not proposing turning it off all together.Turn. It. off. for now. To test. The whole “I haven’t done anything different” I understand why you are saying this but we have not accused you once of doing anything different. We are walking through a normal troubleshooting process and you have to cooperate with us instead of arguing about how you didn’t change anything.
There’s only so much I can do without having access to your server and I have exhausted all of those routes.
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Best way to trigger an audio file to be played over an extension
There are any number of posts here as to how to use “call files” in your case use applications (playback) and your prerecorded file
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DOS Vulnerability in Asterisk chan_skinny CVE-2017-17090
Just utilize asterisk-version-switch instead.
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Latest Backup Module not working
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FreePBX Security Alert (VoIP Server)
Hi andrew…ok that makes more sense. the only firewall is the linux firewall. I’ll try and get some time to work on this in the next day or so and see what happens.
I’m not trying to argue just you need a little more words than being really terse.
I’ll report back once I can play with it
thanks leon
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Protocol 'tftp' is not enabled. Please enable in System Admin module
This topic was automatically closed 24 hours after the last reply. New replies are no longer allowed.
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ERR_CONNECTION_REFUSED After trying to connect to extension
I have a fresh install of STABLE (LEGACY)
10.13.66-32bit
Release Date: 2016
FreePBX 13 • Linux 6.6 • Asterisk 13
I am using a unifi usg and 24 poe switch.
FreeBPX is installed on an 2006 iMac
Whenever I try to connect a softphone to an extension the internal connection gets refused and I have to reboot to get to the gui. As it is FreePBX is unusable as I can’t connect to an extension. While the connection is blocked the server remains connected to my external SIP account.
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VEGA sometime can not access
hello：
I face some issues to access the VEGA gateway by browser. sometime, the VEGA can not access anymore. after trying few times, they may access. It make me crazy.
Do i need to upgrade the firmware?
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Latest Backup Module not working
This evening I downloaded 14.0.3.12 and so far so good. Watching it.
Thank you Andrew.
Edwin
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Incoming calls to SIP Trunk not working after 10.13.66.22 -> SNG7 upgrade
I used the now-production, latest script to upgrade from 10.13.66.22 to SNG7.
Everything went somewhat smoothly (had some modules that I needed to manually re-install, but nothing too bad.)
Extension-to-extension calls work, outgoing calls through our SIP trunk work, but incoming calls through our SIP trunk get a “this call cannot be completed” message. All calls (both directions) worked fine on 10.13.66.22.
Our SIP trunk provider is Broadvoice, and they use IP address authentication (no need to use user/secret). Approx 6 months ago I tried to get the trunk working on 10.13.x with pjsip, but was unable to, so I left it on chan_sip. All of the extensions are on the local LAN and are using pjsip.
My sip Settings for outgoing are:
type=peer
trustrpid=yes
sendrpid=yes
qualify=yes
nat=never
minexpiry=30
maxexpiry=3600
insecure=port,invite
host=206.15.150.13
dtmfmode=rfc2833
disallow=all
detectfax=yes
context=from-trunk
canreinvite=nonat
allow=ulaw&g729
Incoming:
User Context: default
User Details: [empty]
Register String [empty]
Again, those settings worked perfectly for both in and out on 10.13, but incoming does not work on SNG7 while outgoing does work.
Any advice is greatly appreciated, and am also happy to share any info/logs if they would be helpful!
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Incoming calls to SIP Trunk not working after 10.13.66.22 -> SNG7 upgrade
FYI, this is Broadvoice’s support page on configuring this product with FreePBX:
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Vega 50 configuration - Call does not release when far end drop the call
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Quick calling via names
Does anyone know if this can be done?
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FreePBX 14.0.1.20: Incoming calls from outside network do not go through
Environment: FreePBX 14.0.1.20 running on a Dell PowerEdge server.
SIP Trunk Configs:
Outbound:
Trunk Name: ESVC-TRUNK
host=siphosthame
username=sipusername
secret=sippassword
type=peer
Inbound:
USER Context: ########## (our main phone number)
description=ESVC-inbound
host=siphostname
type=friend
dtmfmode=auto
allow=all
insecure=port,invite
canreinvite=no
context=from-trunk
We have two VoIP servers on our premises: A Barracuda Networks CudaTel that we’re transitioning away from and the new FreePBX server. Both are pointed to our ISP’s SIP host, with separate accounts. We have one Polycom VVX400 phone provisioned on the FreePBX server with extension 50. Provisioning works fine and we can make outbound calls on that phone no problem. We have an inbound call route defined that is supposed to pass any DID number that touches it. There is an inbound DID associated with extension 50, and it also shows up in the Inbound Routes.
If we try to call the DID line from an IP phone in the same office as the FreePBX server, it goes through (a phone connected to the CudaTel). If our ISP tries it from their office, it also goes through. No calls from outside our network or our ISP’s network go through. We get the following error messages when following the log:
[2017-12-14 12:24:01] WARNING[11388][C-0000009e]: chan_sip.c:17235 check_auth: username mismatch, have <ESVC-TRUNK>, digest has <s>
[2017-12-14 12:24:01] NOTICE[11388][C-0000009e]: chan_sip.c:26312 handle_request_invite: Failed to authenticate device "757XXXXXXX" <sip:757XXXXXXX@162.XXX.XXX.XXX>;tag=as2c984c3f
Where 757XXXXXXX is my cell phone number and 162.etc is the IP address of our ISP’s SIP server. These errors occur whenever someone from outside our office tries to call the DID line. So it appears that the server is trying to authenticate the outside calls? I’ve looked around this forum and done some deep-diving on Google, but I’m coming up short. Any suggestions? I can provide config files and/or more detailed log entries, if need be, including what we see with a successful call.
I’ve also checked and re-checked the firewall settings. There’s nothing there that I can see that would be blocking the traffic. We are not running the firewall on the FreePBX server, since we have a corporate firewall at our perimeter.
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