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Freepbx 17 problems feedback


Can't connect Voicehost SIP

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Hi,

Have you checked if the trunk is connected using the following command: asterisk -rx “sip show peers”
?

You should verify if your trunk is connected and if you are receiving a “200 OK” status or any other response.

If the trunk is connected, you can check the full log of a call attempt using the command: grep XXXXXXX(dialed-phone-number) /var/log/asterisk/full. This will help you identify the reason why you cannot call through the provider.

Can't connect Voicehost SIP

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That only applies to the obsolete chan_sip and the OP should be using chan_pjsip.

Can't connect Voicehost SIP

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Correct, but still checking the trunk connection is the first step :slight_smile:

No audio on some PJSIP endpoints

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Trying to do Direct Media between two endpoints that are behind NAT causes this problem. It is random because depending on the router, firewall and other factors it can change how the SIP packets are handled. If the network doesn’t handle NAT problem and is sending a private IP in the SDP/media information then the other phone is trying to send media over the public Internet to a non-public IP which is going to fail and give you the issues you were having.

Can't connect Mittel 6930 to FreePBX

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Hi,

I’m having trouble connecting Mittel 6930 phones to my FreePBX. Looks like I have some settings wrong? Is there anyone who successfully connected Mittel phones to a FreePBX and can send me the right settings? This is what I have now:


Thanks in advance for your help!

Can't connect Mittel 6930 to FreePBX

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What issue are you having exactly?

No beeping sound when call is parked

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I am sorry if this is the wrong place to post this, but i’ve tried to contact Yealink Support to no avail.

I have two Yealink T43U phones connected to FreePBX 16.0.40

The user has a BLF for parking extensions on their phone and when they park a call, the BLF lights up as it should, but they do not get the audio alert (beep beep sound) while the call is parked. The audio alert works when the user puts someone on hold though. I don’t know if this could affect it, but there are a lot of users that have the same BLF configuration for the park extensions.(Meaning they see the same BLF light up at the same time.)

I have already factory reset one of the phones, but no luck.

Any help is appreciated.
Settings1
Settings2


Accountcode value not flowing in extension_custom.conf?

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Sorry for the confusion. It’s not working as far as I can tell. The first row is the one that matters as it calls to chow the value available to the dialplan (i.e. using the value set on the dialing extension), which shows nothing. Line 2 and 3 I built only as sanity check so I could set a value (which was missing) so I could continue testing the rest of the configuration.

So essentially, configuration:

[macro-dialout-trunk-predial-hook]
exten => s,1,NoOp(accountcode w channel is ${CHANNEL(accountcode)})

Log:

14104	[2024-05-27 20:27:10] VERBOSE[21587][C-0000005d] pbx.c: Executing [s@macro-dialout-trunk-predial-hook:1] NoOp("PJSIP/98200-00000095", "accountcode w channel is ") in new stack	

Call Control for Sangoma Phone Desktop Client Queues

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Has anyone been able to get the Call Control permissions (monitor, whisper, etc) working on Sangoma Phone Desktop ?
https://sangomakb.atlassian.net/wiki/spaces/SS1/pages/31260975/Sangoma+Phone+Desktop+Client+Queues

Same question here : Call Control (monitor, whisper, and barge)

I’ve got an ongoing ticket with Sangoma tech support but it isn’t going anywhere. The sales team has given me a demo of this feature and it is working for them but I can’t seem to figure out what I’m missing.

BTW I’m not looking for chanspy. We do have a solution for whisper and monitor features but having it native to Sangome Phone Client would be simpler for most clients.

No audio on some PJSIP endpoints

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I understand what you are saying and agree with it. I figured that out before posting. My question is what changed last week to make this start happening on just this system? Yes, I know there is a NAT issue so I made changes to address it.

Since NAT is enabled in the SIP setting direct media should be disabled. This is evident in the fact that this system has been working without issue for over three years.

I am trying to determine what changed to make this start happening. I did not enable direct media last week. I made no changes to the system at all. The only changes were to disable direct media when making the correction.

I am working under the theory that there was a module update that caused this. The system is version 15 with Asterisk 18. I will update to 16 and see if the issue is resolved.

No audio on some PJSIP endpoints

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The only thing a module would do is either turn it on in Asterisk, or change Asterisk configuration to allow direct media to occur where it didn’t occur before.

Since Asterisk hasn’t changed, those are the only two things I can think of from a FreePBX/Asterisk perspective.

No audio on some PJSIP endpoints

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Direct Media is enabled by default, in FreePBX, for chan_pjsip extensions. You have to actively turn it off.

Accountcode value not flowing in extension_custom.conf?

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Where are you setting the accountcode to begin with? It has to be set by something.

Sng_freepbx_debian_install failure

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I’m using the script at GitHub - FreePBX/sng_freepbx_debian_install: FreePBX 17 Installation Script attempting to install on an AWS instance running Debian 12.5.

The installation script runs as expected for quite a while before failing to complete…

The last ~10 lines from the script output are:
2024-05-28 14:13:51 - Installing FreePBX packages
2024-05-28 14:13:51 - Installing ionCube packages.
2024-05-28 14:13:51 - Installing ioncube-loader-82 …
2024-05-28 14:14:13 - ioncube-loader-82 installed successfully…
2024-05-28 14:14:13 - Installing sysadmin17 …
2024-05-28 14:15:10 - sysadmin17 installed successfully…
2024-05-28 14:15:11 - Installing sangoma-pbx17 …
2024-05-28 14:16:50 - sangoma-pbx17 installed successfully…
2024-05-28 14:16:50 - Installing ffmpeg …
2024-05-28 14:17:29 - ffmpeg installed successfully…
2024-05-28 14:17:29 - Enabling modules.
2024-05-28 14:17:30 - Restating fail2ban
2024-05-28 14:17:32 - Installing FreePBX 17
2024-05-28 14:17:33 - Installing freepbx17 …
Timeout, server 35.92.209.30 not responding.
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And the last ~dozen lines from /var/log/pbx/freepbx17-install-2024.05.28-13.38.08.log are:
2024-05-28 14:17:33 - Installing freepbx17 …
Reading package lists…
Building dependency tree…
Reading state information…
The following NEW packages will be installed:
freepbx17
0 upgraded, 1 newly installed, 0 to remove and 4 not upgraded.
Need to get 188 MB of archives.
After this operation, 598 MB of additional disk space will be used.
Get:1 http://deb.freepbx.org/freepbx17-prod bookworm/main amd64 freepbx17 amd64 17.1-1.sng12 [188 MB]
Fetched 188 MB in 31s (6100 kB/s)
Selecting previously unselected package freepbx17.
(Reading database … 107054 files and directories currently installed.)
Preparing to unpack …/freepbx17_17.1-1.sng12_amd64.deb …
Unpacking freepbx17 (17.1-1.sng12) …
Timeout, server 35.92.209.30 not responding.
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At this point the server is unreachable by any means (but it is still running). It seems as though there’s some firewall restrictiction that’s blocking the ssh connection.

I’ve tried this installation process three times now, with the exact same result each time.

Before I dive deeply into the script to see what’s what, has anyone else ever experienced this behavior?


Accountcode value not flowing in extension_custom.conf?

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It’s set under extensions for each extension. I tried both via GUI (Extensions/Advanced) and Bulk handler (screenshots in first post).

Route notification calldDetails show time in UTC not in configured time zone

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We receive route notifications with the call details specifying the time in UTC.

We configured a timezone in System Aadmin > TimeZone area, but the notification call details show up as UTC.

Is there a way to correct this?

Accountcode value not flowing in extension_custom.conf?

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What are you calling from? Sangoma Desktop client? Zulu?

Is it safe to delete the contents of /var/spool/asterisk/tmp

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There is alot of files there 7.3 G worth and my disk is getting full. Is it safe to delete these files?

Sng_freepbx_debian_install failure

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Try to add some Public DNS servers (8.8.8.8 or 1.1.1.1) and hit again.

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