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Hi, I have installed Flexisip and Kamailio on two different servers, I can able to get connected to both via the Linphone app, but can't able to make it via SIP.js in browsers, can this work with browsers..?

this is for sip clients only right..?? , coz we use sip.js in the mobile app which is webrtc, for that also we need to use like this..??

So, in order to do that, I must have a proxy server that will keep the mobile sip status always registered by sending proxies SIP messages to the server

The 3CX Management Console may show a mobile app extension as "registered" even if the mobile app is offline because the extension is registered to the 3CX server itself, not directly to the mobile device. When the mobile app connects to the 3CX server, it registers the extension with the server...

I am not asking on the mobile side, Is there any possibility in freeswitch to make it always registered here

Scenario: 1. Agents A and B are registered via mobile, both can get a call if their mobile app is open. 2. If Agent B is closed and kills the app on mobile. in this state, Agent A is trying to call Agent B. 3. Due to Agent B is not registered, Agent A gets an 'extension is currently unavailable'...

Actually, I have a service in a server that will trigger the push notification if someone makes the call, but here the issue is the one who call the mobile extension received a user unavailable status while calling. how to overcome this

Hi, we are working on the mobile app to make calls from FreeSWITCH via sip.js. when the app is open the extension status is registered and we can able to make calls, but when the app is closed the call is not coming. How to make extensions always available even the app is closed?

when press *4 the log shows like below 2023-04-17 06:08:42.436124 95.80% [DEBUG] switch_rtp.c:1982 rtcp_stats_init: audio ssrc[476567410] base_seq[2641] 2023-04-17 06:09:03.876145 93.97% [NOTICE] switch_channel.c:1123 New Channel sofia/internal/101@testSIP.com...

it says dtmf is disabled, but other dtmf are working fine.

it also shows 2023-04-17 04:52:15.136114 92.93% [WARNING] sofia.c:10149 IGNORE INFO DTMF(*) (This channel was not configured to use INFO DTMF!) 2023-04-17 04:52:17.316116 92.37% [WARNING] sofia.c:10149 IGNORE INFO DTMF(4) (This channel was not configured to use INFO DTMF!)

Dialplan: sofia/internal/100@testSIP.com Action bind_digit_action(local,*4,exec:execute_extension,att_xfer XML ${context},${bind_target},${bind_action_target}) Dialplan: sofia/internal/100@testSIP.com Action digit_action_set_realm(local) Dialplan: sofia/internal/100@testSIP.com Regex (FAIL)...

Hi, sorry for the delay.. it's still not working, I was using this dialer to make calls.

ok, so it will be like 'sip:*4003@domain.com'..?

I have updated the bind_action_target XML, but still it doesn't work. Testing Scenario:- Ext A - 001 Ext B - 002 Ext C - 003 1. Ext A makes a call to Ext B - Works fine 2. During the call, Ext B wants to transfer the call to Ext C, so Ext B dials *4003# can click call, but the call...

Hi, I am working on the dialer, I want to make an attended transfer and conference between calls. i have updated the Dialplan->bind_digit_action as follows but when i dial *4{ext_num}# the call get disconnected.

HI, I have recently started exploring fusion PBX, can someone help with the attended transfer..??