That would be correct as a transfer to a queue the queue handles the recording. In FreePBX world the queue makes the decision on the recording and that is done as a separate file.
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FreePBX not recording supervised transfer
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FreePBX not recording supervised transfer
Tony,
Sorry, I’m not following.
There is no recording of the Callee speaking with Extension 101. The two recording files are of Extension 100 speaking with the Callee, and then the Queue recording is Extension 100 and 101 Speaking during the Warm/Attended Transfer. Once the transfer is complete, there is no further recording.
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WebRTC with FreePBX
No secret to doing it. Treat it like a normal IP phone. Check you logs to see if the registration attempts are reaching the PBX and go from there.
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DAHDI AEX800 - Debian 12
i’ve trying install Digium AEX 800 on Debian 12, but with successful.
Here are some outputs:
lspci -vvv
02:08.0 Ethernet controller: Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express) (rev 11)
Subsystem: Digium, Inc. Wildcard AEX800 8-port analog card (PCI-Express)
Control: I/O- Mem- BusMaster- SpecCycle- MemWINV- VGASnoop- ParErr- Stepping- SERR- FastB2B- DisINTx-
Status: Cap+ 66MHz- UDF- FastB2B+ ParErr- DEVSEL=medium >TAbort- <TAbort- SERR- <PERR- INTx-
Interrupt: pin A routed to IRQ 11
IOMMU group: 9
Region 0: I/O ports at e000 [disabled] [size=256]
Region 1: Memory at f7120000 (32-bit, non-prefetchable) [disabled] [size=1K]
Expansion ROM at f7100000 [disabled] [size=128K]
Capabilities: [c0] Power Management version 2
Flags: PMEClk- DSI- D1+ D2+ AuxCurrent=0mA PME(D0+,D1+,D2+,D3hot+,D3cold-)
Status: D0 NoSoftRst- PME-Enable- DSel=0 DScale=0 PME-
@pbx01:~# dahdi_hardware
pci:0000:02:08.0 wctdm24xxp- d161:8002 Wildcard AEX800
@pbx01:~# lsmod | egrep dah
dahdi_voicebus 77824 1 wctdm24xxp
dahdi 258048 4 wctdm24xxp,dahdi_voicebus,oct612x,wcaxx
@pbx01:~# dahdi_genconf -vvvv
Default parameters from /etc/dahdi/genconf_parameters
Empty configuration – no spans
Generating /etc/dahdi/assigned-spans.conf
Empty configuration – no spans
Generating /etc/dahdi/system.conf
Empty configuration – no spans
Generating /etc/asterisk/dahdi-channels.conf
@pbx01:~# dahdi_cfg -vvvvv
DAHDI Tools Version - 3.1.0
DAHDI Version: 3.4.0
Echo Canceller(s):
Configuration
Channel map:
0 channels to configure.
===========================
Can someone give me some tip ou help me?
Thank you!!!
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Registering as a SIP station to another PBX
There’s no difference between registering as a SIP trunk and as a SIP station, as SIP doesn’t have a concept of a trunk.
Typical “extensions” expect the PABX to outbound register and outbound authenticate. They should reflect the contact user, as the request URI user, to follow the SIP specification. This is what FreePBX treats, by default, as the “DID”.
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Can't dial outbound calls
It’s being rejected by FreePBX’s restricted route logic, not by the provider. I’m afraid the online documentation has disappeared, so I can’t provide any more detail.
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Can't dial outbound calls
I see. Does anyone else have any ideas about FreePBX’s restricted route logic?
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Can't dial outbound calls
I stand corrected, I did totally miss that. I am actually not sure where that’s actually configured. Is this the same thing as Class of Service?
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Registering as a SIP station to another PBX
@david55, this is what I have read online, and seems to make sense.
When it comes to registering a typical SIP device, I’ve typically only had to point the device at the VoIP server IP, enter the extension number, and the SIP Auth Username and Password, and thats about it for a simple setup. When it comes to the SIP trunk config area of FreePBX, there are a LOT of options and I think I’m a bit confused on what I need to config, along with anything else on the Asterisk side that might be needed. Any insight, examples, or other helpful hints to get me going?
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Registering as a SIP station to another PBX
I would take the defaults on almost everything.
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Installing FreePBX on Asterisk 22 using install script?
When running
sng_freepbx_debian_install.sh --noasterisk
on a system with Asterisk 22 installed I get the following:
2024-09-09 15:27:14 - Error at line: 1130 exiting with code 255 (last command was: fwconsole reload >> $log)
2024-09-09 15:27:14 - Exiting script
root@FreePBX-17-Asterisk-22-CM-Patch:/tmp#
the log says:
2024-09-09 15:27:12 - Reloading and restarting FreePBX 17
Reload Started
In Self_Helper.class.php line 214:
Unable to locate the FreePBX BMO Class 'Cdr'A required module might be disabled or uninstalled. Recommended steps (run from the CLI): 1) fwconsole ma
install cdr 2) fwconsole ma enable cdr
root@FreePBX-17-Asterisk-22-CM-Patch:/usr/src/asterisk-22# fwconsole ma install cdr
Detected Missing Dependency of: framework 17.0.1
Found local Dependency of: framework 17.0.19.9
Installing Missing Dependency of: framework 17.0.1
Updating tables admin, ampusers, cronmanager, featurecodes, freepbx_log, freepbx_settings, globals, module_xml, modules, notifications, cron_jobs...Done
<error>Error!</error>
<error>Unsupported Version of 22 </error>
<error>Supported Asterisk versions: 18, 19, 20, 21</error>
Given that Asterisk 22 is supposed to be out in less than a month - and it looks pretty much the same as Asterisk 21 - is there a plan to change the “supported version check” in all of the FreePBX modules to include the number 22 sometime soon?
Just askin! 
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FreePBX not recording supervised transfer
Ya that makes sense as the call to the queue is the actual extended doing the attended transfer. No surprise it loses the second part of the call recording.
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Sangoma Talk lost connection to PBX reconnecting
@dobrosavljevic yes sir. Yeah I can’t explain it. I’ve successfully upgraded on several other deployments that are also running Sangoma Talk without issue. The only difference I can think of is this is a bare metal vultr server whereas all the rest of my deployments are on vultr shared CPU VMs.
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Cisco 7941G gets a "Registration rejected"
Well if that what you think, I’m pretty sure what a range is mathematically and well understand the modulus operator and the RTP protocol, but who am I to disagree with your thinking?
I am also not the one who boldly posted
you may want to narrow this range down to say 16384 and 16390, and then map UDP ports 16384 through to 16390 to your phone from the outside. This will allow your phone to receive inbound RTP voice streams.
because it is flawed in math and comprehension , RTCP replies to 16390 on 16391 ( which comes from the far end) would not get back to Asterisk. further more, 3 and a half channels of media is useful for what diagnoses?
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WebRTC with FreePBX
Thank you, I’ll take a look at he logs.
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WebRTC with FreePBX
It’s been a long while since I did that but start with the live demo,
mostly wrong port or bad certificates are the stumbling blocks
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WebRTC with FreePBX
Thank you. Most appreciated.
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No Audio when call gets forwarded to a cell phone
Your PBX may fail to receive incoming RTP for several reasons:
- External firewall (if you have enabled one for your Droplet).
- FreePBX firewall (misconfigured; by default it should allow the Asterisk RTP range of UDP 10000-20000).
- FreePBX sending an incorrect IP address in the SDP, e.g., External Address misconfigured in Asterisk SIP Settings.
- Conceivably, the provider doesn’t send RTP until it receives some.
In most of these cases, if the PBX sends RTP, the RTP coming from the provider appears as ‘replies’ and would be passed by the firewall. Also, the provider may do ‘symmetric RTP’, sending RTP to whatever address and port it receives RTP from.
When you Force Answer, the PBX starts sending RTP (ringback tone, MOH, etc.), which appears to allow RTP to come in. When the callee answers, that RTP is sent to the terminating provider, opening the path for the callee’s RTP to come in and get passed to the originating provider.
If you can’t easily find what is wrong, instead of Force Answer, you might try setting Inband Progress on the incoming trunk, possibly along with Signal Ringing on the Inbound Route. With luck, that will send ringback tone (or music) as early media, opening the audio path without answering the call.
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Can't dial outbound calls
probably not here, it is an obfuscated commercial module, go to the source
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Exciting News: FreePBX 17 is Now Generally Available!
Good to see your same attitude towards the world on mailing lists carries on here too.
First, bit dot ly has been around a long time, it’s not going anywhere, and if it does, so what?
2, the rant is his opinion saying what he experienced, I’m so sorry it bursts some bubble.
I should suggest he come here to answer your very obvious nose out of joint response, but I’m not so petty that I crave to start a trolling match between you and him, did you comment over there? No? I don’t see it.
It kind of stands to reason that if its deprecated and now removed as stated on asterisk.org then its obviously dead, even I can see that.
As for your nit picking on protocols, I understood what he was getting at, after all I think his article is meant more for clueless, since if you are experienced, you wouldn’t need to read it in the first place - K.I.S.S. look it up.
I cant talk for him and wont, but the BTW at the end, he did say nothing was available for him to see, presumably that means nothing in the log either, since he said where what and where the log was you would imagine he would have looked, so what would you report it as Ted if you have generic banner errors but no substance, we know where it would end up - file 13.
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