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In my case there was no simultaneous registration, it was just one client trying to register, there were no active calls on the server, practically no load, except that I was watching fusionpbx webconsole. So it may be the website code which locks up database rather than registering sip clients.

Also it might not have been an issue for others who have low registration time out, I have registration time out of 60 minutes as my clients connect via TCP. My assumption is that the database gets updated (asynchronously) after the register response has been sent, so if database write fails...

Let me clarify the debug message posted earlier, I freshly start a SIP client and it tries to register, the sip auth challenge appears on the logs, then the sqlite warning appears. In the SIP client the status is NOW registered. However when you check the fusionpbx Status-> registrations - the...

Now and then when an user registers, the user is actually registered (as per his staus on the SIP client) but it doesn't get updated on the database, this results in no calls getting routed. If Sqlite is unable to write to database, it should not send an acknowledgement back to the client. The...

You probably need to pay some one to get your stuff working if you can't fix a gateway connection

The following lines had disappeared from the XML in the inboundroutes, false alarm - all well now <action application="set" data="domain_uuid=344f3a18-9840-4920-b0c3-4e205a05c18e" inline="true"/> <action application="set" data="domain_name=domainname" inline="true"/>

I had a setup where incoming calls used to hit the extensions without issue. I rebooted freeswitch few times as the toggling of ringback variable was not getting reflected unless freeswitch was restarted. Can some one let me know what did I mess up. When an inbound call hits freeswitch, I...

https://www.pbxforums.com/threads/32-second-call-disconnection.5464/#post-21667

I know this is an old thread, I came across an interesting behaviour,if you have early media then your late codec negotiation goes for a toss. As the early media's codec is used as A leg's choice. If you want late negotiation to work as you intend to, disable ringback and transfer ringback in...

ALG is disabled, I connect via TLS - the issue is with the RTP port which gets opened randomly for every call. I figured out that this random port for media causes an issue i.e the client sends out some port number as its listening RTP port, but symmetric NAT maps it to something else...

you may be right, I thought it was due to the nature of symmetric NAT which was causing a slight delay in RTP ports being opened. I just changed the setting on the router to a full cone NAT, - I have done few more things like disabling instant ringback - with this setup the RTP ports take a...

As mentioned earlier In the profile you need to enable "tls" and also set the value to true - you haven't done that , you need to add a field called tls and set it to true in the profiles - I assume you are trying to change the variables - which is not the place. Go to Advanced -> Sip Profiles...

You post a picture of your profile and I will check it up

The first part of speech lost is mostly due to the client behind NAT and the inward rtp port doesn't open up until the client establishes a connection with the server

In the profile you need to enable "tls" and also set the value to true, only then you will get the tls value to be true. You may need to restart the profile - rescan alone didn't work in my case.

I have a multi tenant setup. The Global preferred codec list is set to PCMA,Opus. Tenant 1 and 2 have inbound_codec_negotiation set to generous (I also tried scrooge) inbound_late_negotiation enabled media_mix_inbound_outbound_codecs is set to false in variables In the dialplan if I transfer...

I followed these instructions https://docs.fusionpbx.com/en/latest/additional_information/sip_tls.html No errors. After the profile has been re-scanned, the sip profile status shows the tls profile is running on port xxxx external_for_internet Profile sip:mod_sofia@192.168.1.2:xxxx...

<condition field="destination_number" expression="^(?:\+?countrycode)?(\d{10})$">

401 response is an auth challenge, your client will need to respond to it, 407 is also proxy auth reated. try using zoiper as a client and see whether you can progress.

I just made another test of capturing packets - this time, I just rang the phone without answering it. The first packet still had the high delta - so it appears that the first packet with high delta is NOT the voice packet but ring back tone. I guess the issue is something else.