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If you're referring to my post, we had to brute force it by just rewriting the SDP headers using our SBCs. I'm pretty sure it's related to the issue this person is talking about, where it's set in the session attributes, but not in the media (or vice versa, I can't remember), so freeswitch just...

Just to clarify, is this alluding to the lack of 'CALL_REJECTED' in the list of originate_dispositions that constitutes a missed call?

I am not affiliated or contributor to the FreeSWITCH project in any way, but you can go through the mailing list to see 2 decades of various memory leaks. For what it's worth, the past 3 or 4 years have been pretty solid... or I haven't hit the code paths that trigger them.

Don't forget to ensure that your deploy hook doesn't run in the middle of the working day. FreeSWITCH usually takes a good minute to get back into a functional state, not to mention killing any active calls. As for the sofia port debacle, I have definitely seen it with my own eyes.

For what it's worth, I've typically put a cronjob in the early hours of Sunday morning to restart freeswitch; primarily because of many historical memory leaks with freeswitch, stuck calls that remain in the core database, and for certificate renewals.... or if you're messing around with the...

I'm fairly certain you can just re-run the renewal command in a cronjob, I guess every day. /usr/local/sbin/dehydrated --cron --config /etc/dehydrated/config

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.