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Custom extension also possible by user interface?

October 25, 2021, 8:05 am
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@nxswart wrote:

Hi Guru’s,

So on this moment I have a call-file that I put in the outgoing folder and Asterisk then makes a call for me. I’m doing that by this call file:

SIP/$trunk/$number
MaxRetries: 2
RetryTime: 900
WaitTime: 30
Context: from-internal-custom
Extension: 16031
Priority: 1
Archive: Yes

And I have a piece of code configured in the extensions_custom.conf:
[from-internal-custom]
include => deb-reminder

[deb-reminder]
exten => 16031, 1, Wait(3)
exten => 16031, n, Playback(X)
exten => 16031, n, Wait(1)
exten => 16031, n, Playback(X)
exten => 16031, n, Wait(1)
exten => 16031, n, Playback(X)
exten => 16031, n, Wait(1)
exten => 16031, n, Playback(X)
exten => 16031, n, Wait(1)
exten => 16031, n, Hangup

Now I would like to remove the code in the extensions_custom.conf file and do it all trough the user interface of FreePBX.
I already created an extension 16031 and in that extension I see the possibility to set an announcement.
Only how can I hangup the call from the extension?

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FreePBX with Adtran TA924

October 25, 2021, 11:34 am
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@gregarican wrote:

So I migrated all of my configuration from an old FreePBX 13 on-prem to a new FreePBX 15 on AWS. The main difference is that I provisioned the SIP trunking and extensions as CHAN_PJSIP, rather than the legacy CHAN_SIP. All of the desk phones are working fine.

The only quirk I’ve encountered is I have Adtran TA924’s at our larger sites for breaking out some analog lines. Pointing the Adtran endpoints to the new FreePBX, the new SIP port, etc. was all pretty straightforward. I’ve verified those settings are all changed over okay.

Here is where the quirk is. The analog endpoints are SIP registered just fine. They can outcall just fine as well. But when incalling them the endpoints don’t ring. To remove inbound call routes from the picture I tested them out with another local desk phone calling them.

Would choosing CHAN_PJSIP cause issues for an older (Gen. 1) Adtran TA924? It’s just odd that the endpoints are SIP registered and can outdial just fine.

Here are links to some of the logs. First the Asterisk log for a call that didn’t ring, calling from local desk phone x613 to a cordless analog phone at x641 --> https://pastebin.com/MuUPTb9V. Secondly, a SIP debug capture from the Adtran itself, where I was test calling from my x101 desk phone to an analog fax machine at x644 --> https://pastebin.com/d1248SkR.

A couple of smaller sites only had a single fax machine. So I just use a Cisco SPA112 in those cases. Moving things to point over to the new FreePBX worked fine for them. They can outcall and incall okay.

Any ideas about the Adtran TA924?

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FreePBX upgrade 14 to 15 is broken

October 25, 2021, 12:37 pm
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@fastdrw wrote:

Freepbx upgrade fails in GUI. Instructs to run fwconsole ma upgradeall from CLI.
System then “breaks” and upgrades to FreePBX 16 with many broken modules.

(repeatable)

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Help with sip register please

October 25, 2021, 3:13 pm
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@troymakaro wrote:

I am new to FreePBX. I just installed it yesterday and I signed up for a free 20 day sip trunk test through sipstation.
I have a custom made soft sip client on 192.168.3.150 and my freePBX setup is on 192.168.3.177
Given the following errors, can someone please give me some direction how to solve this? This same client software does work using the asterisk server that I have at work so it must be something with my setup. Note I do not support sip tls yet.

[2021-10-25 14:54:30] NOTICE[25359] res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ from ‘<sip : 3aIgIdig6bRv @ 192.168.3.177>’ failed for ‘192.168.3.150 : 5060’ (callid: d519448-0-13c4-65014-446a2-711191fc-446a2) - No matching endpoint found

[2021-10-25 14:54:30] NOTICE[25359] res_pjsip/pjsip_distributor.c: Request ‘REGISTER’ from ‘<sip : 3aIgIdig6bRv @ 192.168.3.177>’ failed for ‘192.168.3.150 : 5060’ (callid: d519448-0-13c4-65014-446a2-711191fc-446a2) - Failed to authenticate

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How to match CID for call destination?

October 25, 2021, 10:54 pm
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@guillaumesoucy wrote:

Hello,

I need to send call form a specific phone number to "Terminate Call"I read on another topic on this forum that I need to use CID match.

I try but I don’t understand where to put the number to block:

Screenshot%20at%202021-10-26%2001-45-23
Screenshot at 2021-10-26 01-45-23.png719×478 33.5 KB

It tell to use an inbound route and enter the number stating by a underscore. Do I need to put this number in DID or CID?

Thanks!

Guillaume

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How to delete recordings?

October 25, 2021, 9:17 pm
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Dashboard not showing for around 2 minutes

October 26, 2021, 2:26 am
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@dysmas wrote:

After a manual install of FreePBX, dashboard was working normally. Then I installed around 10 modules. Some time later I noticed that when opening FreePBX, or clicking on the Dashboard menu, the progress bar stops at 90% (as usual) but does not disappear. The dashboard is below, but masked by the white rectangle below the progress bar. After around 2 minutes, the dashboard appears normally.
In the console, during the time where the Dasboard remains masked, I see this message :
The console logging API (console.log, console.info, console.warn, console.error) has been disabled by a script on this page.

How can I identify this script and understand where it comes from ?

Otherwise, FreePBX works fine, I don’t know another problem.

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Grandstream GDS3710, no video preview through FreePBX

October 26, 2021, 3:33 am
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@jra2fael wrote:

Hello,

I have a Grandstream GDS3710 video doorbell and a GS3510 video intercom connected to FreePBX through pjsip, and when the GDS3710 calls directly the GS3510 to it’s IP there is a preview of the video on the screen before answering but through FreePBX it is not working.

I tried the solution by @danielf below but unfortunately without success:

I’m getting the added header in the logs, but I don’t know if it is because the URL of the streams seems to have changed, there’s only an RTSP stream I tried with it but without success.

I’ve seen there may be something to configure with pjsip to allow the preview video but I didn’t find exactly what to do.

I’ve tried a couple of things without success, is there any way to see what could be missing during the call to make this work?

Does someone have a solution to this problem?

Thanks

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New Sysadmin Pro Feature - VPN server port is user configurable

October 26, 2021, 12:35 pm
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@lgaetz wrote:

With the edge versions of System Admin, Firewall and Endpoint Manager, you can now configure the port for the OpenVPN service in System Admin, configure endpoints to connect to the VPN using this port and configure the firewall to allow VPN traffic to this port. The primary purpose of this setting would be to set up the System Admin VPN behind a NAT device where 1194 UDP is already in use, but there may be some value in just using a randomly selected high port number for obscurity.

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Sip clients unregistering randomly Stale nonce error

October 26, 2021, 5:17 pm
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@Steveg70 wrote:

I have about 200 clients registering and a handfull of them seem to drop registration randomly and we get errors on the console showing…

[2021-10-26 23:29:26] NOTICE[7582]: chan_sip.c:17471 check_auth: Correct auth, but based on stale nonce received from ‘“CLIENTA” sip:10000110@clients.mysip.pbx;tag=1820798efde56ed6o0’
[2021-10-26 23:29:27] NOTICE[7582]: chan_sip.c:17471 check_auth: Correct auth, but based on stale nonce received from ‘“CLIENTG” <sip:10000027@clients.mysip.pbx;tag=56d93b6a93922684o0’
[2021-10-26 23:29:27] NOTICE[7582]: chan_sip.c:17471 check_auth: Correct auth, but based on stale nonce received from ‘“CLIENTW” sip:10000024@clients.mysip.pbx;tag=351a7fda11b42c09o0’
[2021-10-26 23:29:29] NOTICE[7582]: chan_sip.c:17471 check_auth: Correct auth, but based on stale nonce received from ‘“CLIENTZ” sip:10000026@clients.mysip.pbx;tag=5d0640ccbcf1163bo0’

We have done the usual checked nat is yes, disabled alg and made sure client is authorized though firewall. Client registers fine but after a few minutes looses registration.

We have shortend session expiry to 300, then 60 did not help,
We tried pedantic=no this also did not help

Extensions are using chan_sip

thanks in advance

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Ways recordings can be downloaded?

October 26, 2021, 10:06 pm
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@Juste wrote:

Is there a way recordings could be downloaded otherwise than one at a time ?

That would save me a lot of time if I want to download 2 years worth of recordings.

Thanks

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Some modules cannot be installed on FreePBX 16

October 27, 2021, 3:10 am
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@AndrewZ wrote:

I had to upgrade to FreePBX 16 earlier than planned, almost everything works well, with a few exceptions: two modules - cdr and cel - cannot be installed/upgraded due to php errors.
Is there any fix for that?

root@freepbx:~# fwconsole ma downloadinstall cdr
No repos specified, using: [standard,extended,unsupported] from last GUI settings

Downloading module 'cdr'
Processing cdr
Verifying local module download...Verified
Extracting...Done
Download completed in 0 seconds

In DB.class.php line 153:
                                                                              
  count(): Parameter must be an array or an object that implements Countable  

root@freepbx:~# fwconsole ma upgrade cel
No repos specified, using: [standard,extended,unsupported] from last GUI settings

Downloading module 'cel'
Processing cel
Verifying local module download...Verified
Extracting...Done
Download completed in 0 seconds

In DB.class.php line 153:
                                                                              
  count(): Parameter must be an array or an object that implements Countable

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The FreePBX 15 web interface does not start after updating the framework to the latest version.

October 27, 2021, 3:29 am
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@VaAlSh wrote:

После обновления framework до последний версии не запускается вебинтерфей FreePBX 15.

HTTP ERROR 500

Подскажите с чего начать, куда копать?

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Which SIP client

October 27, 2021, 5:13 am
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@Matthai wrote:

Hi,
may you advise me which SIP clients are good? I am using different systems - primary Linux and iPhone, but also Windows, Android and very occasionally MacOS. I have tried several clients and Linphone works only on Android. On iPhone calls are just getting rejected, on Linux it was not working at all. On Linux Blink was quite nice, but finally I ended up with Zoiper. I have tested it on Linux, iPhone and android and it is working great. However, it is not opensource and free.

I am using RasPBX as a server.

So in short - which PJSIP clients are recommended for all mentioned operating systems (Linux, Windows, MacOS, iOS, Android)? I am specifically interested in opensource clients.

Many thanks for your ideas and thoughts.

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Properly Defining DSCP for PJSIP endpoints and trunks

October 27, 2021, 7:09 am
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@gregarican wrote:

I have a FreePBX 15 deployed with PJSIP throughout. No chan_sip used. I noticed that there are some spotty call quality issues. My sites all have prioritized DSCP 46 traffic on their routers and switches to help transport a bit. When I look in the FreePBX, here is what I see.

Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress....................>
==========================================================================================

Transport:  0.0.0.0-tcp               tcp      3     96  0.0.0.0:6062
Transport:  0.0.0.0-tls               tls      3     96  0.0.0.0:6061
Transport:  0.0.0.0-udp               udp      3     96  0.0.0.0:6060
Transport:  0.0.0.0-ws                 ws      3     96  0.0.0.0:5060
Transport:  0.0.0.0-wss               wss      3     96  0.0.0.0:5060

I assume that tos is analagous to DSCP, correct? In this case I need to set it for 46, EF, 0xB8. Where exactly is this setting found in FreePBX 15?

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VM file to E-mail

October 27, 2021, 7:32 am
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@ilouiemiami wrote:

Guys, I have two instances of Freepbx set up and I’m having an issue on both of them. The issue I’m having is that when someone leaves a VM, it’s supposed to be sent to an e-mail as attachment. I do receive an email but, on one I get a .wav file that says INVALID file, and on the other I don not get a file at all. Would someone be kind enough and point me in the right direction, please. I’d like to fix this issue ASAP. Thanks

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Network settings stuck on unconfigured and cant change DNS settings

October 27, 2021, 8:47 am
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@chris43 wrote:

Hi

while attempting to make some network changes 2 problems has developed.

when attempting to change the gateway the web page has unconfigured highlighted. the correct (old) configuration shown in the Static ip, netmask and gateway.

after clicking static and changing the gateway and clicking save interface the dialog box save changes appears and the save and apply is selected.

as it states in the dialog box a reboot may be required the system is rebooted. after reboot there are no changes in the network settings they appear as above.

the next issue when accessing the system admin DNS to change the dns server after clicking on submit the changes do not apply. could this be related to the above problem on the network settings unconfigured being highlighted?

PBX Version:

13.0.197.31

PBX Distro:

10.13.66-22

Asterisk Version:

13.29.2

Thank you in advance for any help

Chris

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Paging Group rings softphones

October 27, 2021, 9:26 am
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@PYoung wrote:

We have users with a desk phone (Yealink) , Sangoma Connect and Zulu all with the same pjsip extension. Ex : extension 201 with multiple contacts possible. Now we have a paging group (301) which includes extension 201. When we page the Yealink phone accepts the page. No problem here. However Sangoma Connect rings once and shows a missed call. Zulu does not ring but shows two missed calls for “unknown”.

I get that the softphone needs to activate auto answer to accept paging and some don’t support it. The things is we don’t want to receive paging on Sangoma Connect or Zulu or other softphones.

I noticed the Multicast option in Paging Pro might be a solution but this particular PBX has multiple sites and some with an unmanaged network so they are not going over a VPN. This won’t be possible.

Any ideas here to not have Sangoma Connect and Zulu included in the paging group but keep the deskphone ?

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Intrusion Detection won't start - zulu logfiles - gui fix?

October 27, 2021, 9:50 am
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@mvogel4949 wrote:

I’ve been having quite a few systems with intrusion detection stopping and not going able to start via the GUI. The solution has been to access CLI and

touch /var/log/asterisk/zulu_out.log

Is there a solution that works just through the GUI? Updating a module, rolling a module back?

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RasPBX bad sound quality when using a dongle

October 27, 2021, 12:16 pm
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@Matthai wrote:

Hi,
I am completely new here, so excuse my silly questions.

I came across RasPBX and installed it (current version is raspbx-10-10-2020) on my RaspberryPi 3. OS (based on Raspbian) includes Asterisk 16.18.0 and FreePBX 15.0.17.55. I am located in Slovenia, Europe.
I am also using Huawei USB dongle and have set up trunk to mobile network via that dongle.

For clients I am using Zoiper 5 (on Linux, iOS and Android).

Another thing. I had problems with UDP - after 30 seconds (of established connection) call has been dropped. I suppose there has been a problem with SIP ALG, I switched it off on my router, but the problem persisted. So I switched to TCP connection and now everything works just great. However, I am not using TLS/SRTP/ZRTP encryption, but my RasPBX device and clients are operating within secure OpenVPN network. BTW - OpenVPN server is located somewhere else (in Slovenia), so VPN connection between OpenVPN server and RaspPBX and other clients is going through the internet.

Now the problem.

I have tested the incoming and outgoing call via USB dongle and first test were quite good. I am able to call from my GSM to RasPBX and the client can answer the call and sound is of a good quality. I can also call from a Zoiper client to the outside number and call quality is good.

So I decided for some more tests.

I have asked a colleague from USA, Ilinois for a test. Call from one Zoiper client to the other (through the VPN) is working very good. Sound quality is really good.

But then I asked a colleague to call my GSM through USB dongle.

In that case call quality was bad, it was really hard to understand what he was talking.

I am not an expert, but have a feeling this is codec related. This is the list of my codecs under Settings - Asterisk SIP Settings:

  • g722
  • alaw
  • ulaw
  • g729
  • gsm
  • g726
  • g723
  • speex

In a logs, I can see lines like this:
2021-10-27 19:50:07] NOTICE[26589][C-0000000b] translate.c: 2604 lost frame(s) 2605/0 (slin@16000)->(g722@16000)
I assume that means g722 is being used, I have also found out Asterisk is internally using slin coded and then performs transcoding, but don’t know why or if this could affect the sound quality.

What is weird to me is that call from one PJSIP extension and the other has great sound quality, but if call goes through USB dongle, sound is much much worse.

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