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It's possible, but I can't imagine it's 'supported' whatever that means for your support contract. It's something I would only attempt if I really needed a good reason to do so and I'm not entirely sure how this would improve security; I would hazard a guess it would actually reduce security...

I think you want to just use the normal 'Conferences' app instead: https://docs.fusionpbx.com/en/latest/applications/conference.html The Conference Centre is for creating rooms (which are defined by PINs)

Slightly old thread, but I hate seeing a question unanswered. You can modify the clean up tasks in: /etc/cron.daily/fusionpbx-maintenance

So digging a bit deeper, I believe the say: to TTS functionality is part of the FreeSWITCH IVR code, so you are right. The official docs make no mention of this, but the FreeSWITCH documentation has always been crap: mod_dptools: say mod_dptools: speak As for the module error, I've had a look...

I think "speak" is what mod_flite is for. I'm pretty sure "say" only works on the predefined phrases in phrases_en.xml or similar, mainly used for creating automated clocks or other predefined prompt flows. I think that Youtube video just got mixed up during the demo. As for the open bug, I...

I have personally never needed to use STUN/ICE, etc. I only rely on the rport mechanism of SIP and FreeSWITCH/mod_sofia's ability to deal with NAT on the client sites. The phones are behind a mix of consumer routers and some enterprise routers, Cisco, Juniper, Drayteks, Sonicwall etc. I believe...

FYI. It still manifests, but I think it's related to this uncommitted PR: https://github.com/signalwire/freeswitch/pull/2393 (which the author accidentally left a semi-colon in the commit).

Might also seem obvious too, but make sure the firmware is latest. I've had a a handful of SIP related Yealink issues over the years. Also, it's possible a SIP ALG is stripping it. You can confirm that with a packet capture from the Yealink itself.

Never seen that before, is that coming from the Yealink? You can tell with 'sngrep' on the FusionPBX and looking at an INVITE. We have similar issues when a phone is configured to send INVITEs to a PBX, but they aren't REGISTERED. The nobody username does pop up in mod_sofia when a username is...

Slight update: Actually, replacing the a=inactive with a=sendrecv does work. We tried using a=recvonly to simulate kind of inactive, but that didn't work, so I think it's more where the SDP state is very quickly modified at the start of the call. Hard to say really.

Hey all Not been around here for a while, but I've been investigating a recent problem with our FusionPBX servers where callers are reporting one way audio (Outbound call to third parties means no inbound audio, third party can hear caller fine) and it appears to be related to Broadworks...

Well, I discovered what it is, it was the Cisco SIP ALG. I never thought to clear the NAT translations after disabling and the Cisco kept pushing SIP traffic through it long after a typical UDP timeout, I presume it just tracks them like TCP sessions. You have to wonder why router...

Hey everyone. We have a client who is experiencing problems with failed transfers, and I'm just wondering if anyone has any bright ideas about what the typical causes of the nightmare_xfer scenario tend to be: 4c5f0ee2-0a68-4a98-9f7d-2775eabd310b 2025-06-02 12:34:59.578055 100.00% [DEBUG]...

Did you modify any of the dialplans at all during the initial testing? My only thought is that I don't know if both the call_recording_on_demand and user_record can live in harmony both being enabled, presumably due to conflicting bind_digit_actions.

It's *5 and *6 to mask and unmask the recording (silence). You can also use the SIP Record param that some handsets support on DSS keys like on Yealinks, but I find the problem is that they don't start in the 'on' position. The way I prefer to do it usually is by using different extensions...

Yeh, I was thinking of that, just playing something like 'enabled' or 'disabled' to one or potentially both parties after pressing the mask/unmask.

Yeh, pretty much. Anything it sees in the access list is ignored.

Event guard uses the Advanced -> Access Control allowed nodes as a white list. Can add a new Access Control List called Customers or any name you like set to default deny then add the customer CIDR to a new node set as allowed.

Hi all Anyone got any smart ideas to fix this request? Background: User wants recordings on by default, but wants to be able to disable the recordings when they are going to be listening to sensitive information. They are going to be using a Yealink or a Snom handset. Potential Methods: 1...

Good to hear, the only reason why I elaborated is a lot of people come here and expect to just be able to install some free SIP app off the store and push to work out of the box and think it's a problem with FusionPBX and they think there's some magic toggle you can enable. Ah yeh, was going...