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@pbxgeek Thanks you, your tips help me solve the problem. This is the setup which works. 1. Setup the extensions 201 and 202 (without custom dial string) 2. Specify rtp_secure_media=optional:AES_CM_128_HMAC_SHA1_80 in variables 3. In groundwire sip app, Set Preferences>Security>SRTP>SDES...

I stripped down the codec to PCMU/PCMA, and made a call to ring group 600. It failed again. Here is the log. @pbxgeek What magic did you get to get both RTP/SAVP and RTP/AVP in the same message?

Can you elaborate, where do you see the error?

Sure. Yealink phone is not necessary, any softphone will do too. I my case 201 is Yealink and 202 is Groundwire on Android. Here is the setup: 1. Create two extensions say 201/202, and specify...

That didn't work. What i am trying to do here is pretty straightforward and i believe this has worked in the past. This is new installation and maybe the fusionpbx code has become worse than before.

I have SSL enabled in variables with certificates in place, The internal profile has default setting, If i make a call to *97 from an extension, i do get crypto as the following screenshot and trace shows 2024-08-20 16:36:13.509230 91.27% [DEBUG] switch_core_media.c:9003...

Hello, Most phones these days support SRTP/TLS and is not a bad idea to have better security if possible. On the other hand most Trunks/Upstream dont support SRTP. So freeswitch has to manage the following three scenarios, 1. Internal calls: client ----SRTP--- FREESWITCH-----SRTP-----client...

Thanks that was a typo. I have made the changes as you suggested. Unfortunately, it does not fix the problem. A trace of PSTN call to the ring group is shown below. You can see that rtp_secure_media is being set, but SDP does not have crypto lines. +OK log level [7] 2024-08-20 03:28:36.911278...

Unfortunately, i still get an [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION], [terminated][488] error after changing the dial plan,

Hello, I have a Yealink phone (W60B) setup with TLS sip and mandatory SRTP mandatory. The Yealink extension (201) has this dialstring assigned following guidance from documentation (tweaked), My upstream trunk does no support SRTP. The inbound call work fine if setup a dial plan like so...

Hello, I upgraded following the wiki instructions. Upgrade seems fine except some icons on the dashboard seems to be broken. The 'Recent Calls' is not showing the errors. The link points to "https://pbx/themes/default/images/icon_cdr_inbound_.png". It should have been...

Hello, I am using TLS/SRTP with groundwire app on android with FreePBX. The push mechanism is working as expected. I want to use FusionPBX (FPBX) instead, and stood up a server. Unfortunately, the push mechanism on FPBX worked fine when using UDP transport, but it doesnt work properly when...

Hello, I upgraded fusionpbx following the instructions here. Unfortunately after the upgrade the acl modification seems to be broken. When i try to add an ip address to the 'providers' acl, the webui shows that it is added. The fs_cli on the hand does not list the new ip address. Is there any...

Installed freepbx using binaries which got me the xml_cdr module. Seems source compilation was an issue (using LXD here).

I upgraded an old instance running fusionpbx 4.5.24 to 5.1.1 but retained the freeswitch installation (1.10.6-release-18-1ff9d0a60e~64bit). Strangely, this instance stopped doing 'Call Detail Records' after the upgrade. The mod_xml_cdr.so does exist and is loaded. Is there a change between 4.5...

Hello, I have setup a new install in Debian 12 and noticed that CDR is missing. This faq mention the XML CDR module, however it is missing on my system. The freeswitch version is FreeSWITCH Version 1.10.10-release~64bit and fusionpbx is Branch: 5.1 6092b59281b4758cef85227088e4a13493283dc8...

I changed the dial string to {absolute_codec_string=PCMU}sofia/internal/$3@172.16.2.3 and it worked. The dial plan which works is as follows: <extension name="transfer.International" continue="false" uuid="6d1673d9-bc23-4a2f-8da7-fa82c578b566"> <condition field="${user_exists}"...

Thanks, I looked at the Freeswitch page. Any idea what the following means: <action application="bridge" data="{originate_timeout=25},${group_call(bria@${domain_name}+A)},${group_call(deskphone@${domain_name}+A)}"/> I tried calling from the phone connected to the FXS port of the Obihai, and it...

Hello, I have an Obihai Obi110 device which is connected to a landline. I want to use it as an FXO device with Freeswitch/Fusionpbx. I have setup the SP1 Service and have specified 'li' in X_InboundCallRoute. In Fusionpbx my dial plan is as follows: <extension name="transfer.International"...