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Experts, How can we allow the call be diverted to a Voicemail when a customer is waiting on the Call Centre Queue press the key as configured on the "Exit Key". I am using fusionpbx 4.4.11

Experts, I am looking at a way to play a wav file to Called party (Leg B) before bridging the call to Leg A. Once the called party confirms with a DTMF say "1" I will proceed to bridge the call. I tried using "group_confirm_key" and group_confirm_file. Most of the part works fine when the call...

Experts, I am using Mod Call Centre in the fusionpbx. Is there a way to increase the Agent Timeout (Ringing) from 60 seconds to more. I tried to add the Call timeout in the call centre dial plan but it does not seem to work. Can you please guide me what mistake I am doing? <extension...

Experts, Is there a way that we can grab a particular call from the Queue and assign that to any agent? Assuming there are 10 calls in the queue but I want only grab the 7th call and assign that to a particular agent, is that possible?

Was any solution found to this Sir, I am also facing the same issue

Experts, I am using the fusionpbx mod_callcenter. I have a requirement where when the Agent after completing the call clicks on a transfer button, I would like the A leg (Caller) to be routed to an IVR -> say a post Call survey type. When I use the UUID_Transfer both the legs are getting...

So is there any setting where we can change the value from qop to auth? Want to know how it differs?

Noted Sir, this is what the RFC says. qop-options This directive is optional, but is made so only for backward compatibility with RFC 2069 [6]; it SHOULD be used by all implementations compliant with this version of the Digest scheme. If present, it is a quoted string of...

Hello Experts, I was just wondering if there is a way to disable the qop=auth parameter on the 407 response below in freeswitch. I understand this parameter is optional according to RFC 2617. The reason I want to try is, we are using a WebRTC to SIP Gateway (Mizu) and when the 407 is sent by FS...

Thanks and Noted Sir.

To add - I found one more thing - the issue happens when the application (Google Chrome) tries to negotiate TLS v 1.3. It is also strange that for 10 calls the negotiate v.12 and the 11th call for no apparent reason is negotiating v1.3. I am not sure though that could be a reason.

Experts, I am facing a very strange issue - We have many clients connecting to freeswitch via SIP WSS and the client side is using JSSIP. Suddenly for some calls made by the agent I can see "Websocket abruptly disconnected" and freeswitch side shows no logs. I am using FS 1.10.6 and Fusionpbx...

Noted Sir, Thank you.

Hello Experts, I am facing an issue with Record_session. My current setup is Leg A via WSS and Leg B on SIP. When I use Record_session on the dialplan the B leg audio is recorded but not the A leg. If it is a SIP to SIP call both legs are getting recorded. Is there any setting I need to enable...

Hello Sir, I was just wondering if it is possible to have "sitcky agent' setup on the Fusionpbx. i.e when lets assume the Agent has attended to a call from a particular customer say with calling number (12345). when the customer calls again say within 5 days or specified time interval and if...