vicidial.org

Viciphone version 3.0 is now officially released.

With this release comes the following improvements:

• A complete rewrite of the core Viciphone code to support SIP.js v0.20.1
  
  • This does require 'rtcp_mux=yes' be set in the template you are using for your webphone
    
• Support for passing a settings container from Vicidial to Viciphone to allow additional settings to be added in the future without Vicidial changes
  
  • To take advantage of this feature an upgrade to Vicidial is required.
    
  • Viciphone will still work without the Vicidial upgrade, but some features will not be accessible.
    
• Added the ability to enable/disable various browser audio processing like Echo Cancellation and Automatic Gain Control
  
  • Does require the Viciphone settings container support in Vicidial
    
• Auto call agent conference on successful registration. This is similar to a feature that was added to some forks of Viciphone but it will work even if Viciphone and Vicidial are hosted on different domains.
  
  • Does require the Viciphone settings container support in Vicidial
    
• Dynamic translation support. This is a frequent request that has been implemented in several of the forks of Viciphone.
  
  • Does require the Viciphone settings container support in Vicidial
    
  • If you create a new translation, please share the values from your settings container
    
• Play progress audio instead of ringing when placing a call
  
  • The audio file can be changed in the Viciphone Settings Container by adjusting the region setting.
    
  • Currently supported regions include North America, Europe, and the UK.
    

If you already have Viciphone working on your cluster, make sure that:

Code: Select all

rtcp_mux=yes

is set in the template you are using for your webphones. Once that is in place you just need to change the Webphone URL in Admin -> System Settings to:

Code: Select all

https://phone.viciphone.com/v3.0/viciphone.php

Thank you Michael.

Where can we download the ViciPhone v3.0 source. All the links I can find on viciphone.com point to the v1.0 on GitHub. We would like to host the phone on internal servers.

Roger

With Agent ViciPhone v3.0 is there a way to show 'Registered' instead of '1XXXXXXXXXX - ACagcW178654678usernameusernameusernameusername'? On Admin ViciPhone v3.0 in Real-Time Main Report shows 'Registered'.

I will be releasing Viciphone v3.0 on Github once a client gets back to me on whether a specific bug from 1.0 and 2.0 has been resolved. I cannot replicate it in v3.0.

As for displaying "Registered" instead of the caller id of the active call, no I will not be changing that. The reason on the realtime report it does not show a Caller ID Number is because an Admin is not on a phone call.

In version 3.1 I do plan on reworking the interface to give more room to the status Display area. This will allow me to show things like "Registered/Un-Registered", "Incall/On Hook", and the caller id info at the same time.

Michael:

Please correct me if I'm wrong, but version 3.0 appears to perform a call from webphone into the server as soon as the registration completes, instead of receiving the call from the server as all previous phones (either web or softphone).

This way, if you changed your phone's default context (to disallow manual dialing from the phone), the login process fails because it cannot find the conference extension (eg. 8600051).

This means that you must either include the 86XXXXX dialplan in your custom phone's context, or go back to the default one.

Was this an intended behaviour, or is there any other reason to switch the phone login from "receiving" a call to Vicidial to "dialing to" Vicidial?

Yes that is a feature of Viciphone version 3.0 but it can easily be disabled in the VICIPHONE_SETTINGS settings container in Vicidial. Just got to Admin -> Settings Containers -> VICIPHONE_SETTINGS and set dialRegExten to 0.

The source code for version 3.0 has been released on Github:

https://github.com/vicimikec/ViciPhone/tree/v3.0

We use Vicibox 8.0
Asterisk 11.25.3

WebRTC works fine with version 1.0
I have tried putting the rtcp_mux=yes in template and sip.conf

And it is the error when I have google debug console open. So my guess is 11.25.3 can't do rtcp_mux?

Uncaught (in promise) DOMException: Failed to execute 'setRemoteDescription' on 'RTCPeerConnection': Failed to set remote offer sdp: The m= section with mid='0' is invalid. RTCP-MUX is not enabled when it is required.

Asterisk 11 went End-of-Life in 2017, it is missing a lot of features and bug fixes that the newer versions have. We would strongly recommend upgrading to Asterisk 16.

Has 16 been fully tested for support? Last I heard 16 was still undergoing testing and was not yet suitable for production use.

Asterisk 16 is solid in production, we have installed Asterisk 16 on over 100 production dialers at this point. It's very reliable and has been for over a year now.

Hi Mr Florell, The only reason why we have kept with such a old version on all our servers is, that every time we go to new asterisk, the AMD goes out the window even with settings container. With Asterisk 16 did this ever get better or comparable?

A few things:

• Viciphone has never officially supported Asterisk 11. It worked in the early stages, but the standards for WebRTC changed and it stopped working. I suppose the people behind the SIP.js library may have found a work around for this at some point, but I never tested it. Viciphone 3 will never support Asterisk 11 though because as you said rtcp_mux=yes is 100% a requirement and Asterisk 11 cannot do it.
  
• The AMD application is horrible. Always has been. At best it is 80% accurate with its determinations. That means 1 in 5 calls where a live person picks up it will treat them like an answering machine and probably hang up on them. That is a live customer contact you are just throwing away. To make matters worse they will just hear dead air and then get hung up on. When that happens most people will block your number. If you are calling into the US that will lead to your number's reputation going down in the Robocall Mitigation systems carriers have been forced to use. This will lead to your numbers being marked as a SCAM and your connect rate will go down.
  
• Asterisk 11 has horrible security vulnerabilities. One nasty one will cause Asterisk to stop responding. It doesnt crash so Vicidial cannot restart it. It keeps running. It just cannot process any more calls. Even if you have your firewall locked down someone from within your organization could still lock up your systems.
  
• Vicibox 8 has not received OS updates in years. Many packages running on it have security vulnerabilities.
  

AMD isn't really at fault for the changes you are seeing. In Asterisk 12 there were massive changes to the core of Asterisk. These changes forced us to change the way answered calls route. The net result caused two big changes. The first is that calls without any RTP audio actually make it to the routing extensions where they used to get statused as LRERR. This means that the AMD application is now having to deal with these RTPless calls. The second is that the settings that we have in extension.conf for the AMD application are no longer optimal. They were provided to us by a client MANY years ago. Their company had spent months tuning them. No one has taken the time since these changes to retune the AMD settings. While we write call center software, we are not actually a call center. We do not have the ability to experiment with things like this. We are not placing calls to real people. We do not have agents to give us feed back on the changes. It is up to call center operators to do these experiments.

What we have done though is fixed a few bugs that popped up in AMD due to these changes. One such bug was this one: https://gerrit.asterisk.org/c/asterisk/+/13247. Not only did we fix a timeout issue, but we also added a check for calls where no audio data is received. When that happens it returns NOAUDIODATA for the AMD Cause. Matt recently added to the AMD Agent Route Options the ability to immediately hang up NOAUDIODATA calls when they reach the AMD processing AGI script. While this does not prevent the calls from taking up AMD's time it does prevent them from making it to the agents. Just update to the latest SVN trunk of Vicidial and add "NOAUDIODATA-Hangup-ENABLED" to the settings container.

So my recommendation to you is to upgrade to the latest Vicibox which has tones of security vulnerabilities fixed, upgrade Vicidial to the latest SVN Trunk, add "NOAUDIODATA-Hangup-ENABLED" to the settings container, and spend some time tuning the AMD settings in extensions.conf. When you have good settings for AMD, please share them to the Vicidial issue tracker so we can update extensions.conf for everyone.

Just to follow on from the comments regarding AMD, I've found the optimal settings can vary wildly depending on where you are calling, the carrier being used, and load on the machine, so it's best to experiment and find the optimal settings for your campaign(s). It's not perfect and there will be false positives (humans treated as machines) and not to mention those damn greetings where someone says "hello" followed by a short pause, then their greeting, AMD treat these as humans.

You can even get clever with the AMD agi script if you spot a pattern in the stats on the amd report to override the default actions.

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