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I am trying to understand what constitutes the value that appears in the PDD column in the CDR When I dial a local extension, the PDD on fusion PBX is 1.5s to 2.5s - however when I call a PSTN line, it is just 0.82 - I don't understand how that is possible (some times it is 0.38). I did a...

The last entry in the internal profile was wss-binding with value :7443 and the param was disabled. I enabled the param to test out webrtc, however when I did a rescan of the profile, the profile disappeared, the console is complaining that there is no such profile. I checked using 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...

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...

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...

I have a weird problem on internal calls alone, i.e between extensions, the first hello of the callee is never heard by the caller - I just did a packet capture and the first RTP packet alone has a huge delta, it also has a status message saying Payload changed to PT=8 - this message occurs on...

Is it possible to play music on hold via freeswitch when the proxy-media is set to true? In my profile I have late-neg true proxy-media true hold-music local_stream://default disable-hold false...

I have 2 extensions connected to fusion PBX, no external gateway involved - both the sip clients are Zoiper clients from Android device. When mobile A calls mobile B - there is 0% received packet loss at mobile A, however at mobile B the loss as per zoiper is 300%,I can see that the received...

/var/log/messages file is filled with all the ping messages, register messages, I want to get rid of this - these are not even DEBUG/WARNING - how do I switch off this appearing in this file.

I have a setup like this a double NAT scenario. FS (192.168.0.4) -> NAT (Public IP 1.2.3.4) -> INTERNET -> NAT (Public IP 5.6.7.8) -> PHONE (192.168.1.100) Closed the issue as it is not a freeswitch or fusionpbx issue but an external route issue with the ISP

I have fusionpbx/freeswitch running behind a NAT router with a dynamic IP which gets DDNS resolved when it changes. I have sip clients connecting from the internet to the server. Everything works well even when the IP gets updated except the external sip IP address - I have set it to...

I have two domains, one for extensions from LAN and another for extensions that remotely login via internet. Every user has an extension in the LAN and also in the other domain. A user will either be logged in the LAN domain or in the remote domain. If a user's LAN extension (LAN-123) is not...

Newbie alert, I am trying to strip few characters if present in the destination number and updating the destination number inline in a dialplan. However the dialplan which is next in line, doesn't seem to get the updated variable. I am trying to prefix 0 to the destination number but the next...

Please see attached log, I have set an inbound route using Dialplan->Destinations However when an incoming call is received from the gateway, it hits the Dialplans in the Dialplan Manager obviously my expected dialplan is not in that list as it is under Inbound routes. How do I make the inbound...

I am trying to forward an inbound call to an internal extension. I have created an inbound route with a condition of destination_number and performing a transfer. When the call arrives, the dialplan gets executed send 1082 bytes to tcp/[192.168.1.11]:58484 at 11:45:12.987849...

I am having an external gateway which sends an invite for an incoming call like this - can you let me know how do I forward it to an extension based on the gateway uuid This was solved - my gateway had an incorrect context resulting in inboundroutes not getting hit. INVITE...

I have a fresh install of fusionpbx and I have created 2 domains and 2 extensions in each domains. I have managed to setup of cross tenant dialing using this link. When I use a SIP client connected to an extension to send SMS/text message to another extension in the same domain, it works...

2021-07-31 14:19:06.234080 [DEBUG] switch_ivr_originate.c:1407 Play Ringback Tone [%(400,200,400,450);%(400,2000,400,450) I have two internal extensions, one is running Zoiper Client and another is running Sipnetic. When I call from Zoiper to Sipnetic - I don't hear a ring back tone at...

I have few internal extensions and I have an external gateway connecting to PSTN, the external gateway supports only PCMA. In the variables section, I have this global_codec_prefs - PCMA, Opus media_mix_inbound_outbound_codecs = false outbound_codec_prefs = PCMA,Opus My intention is to have...