Auto Dialer - prospecting

We are making outbound cold calls, any suggestions on a Auto Dialer that could be used with FreePBX so that we can have the rep calling out on the same number he would get call back on.

Do you need a manual or an automated process? the most important thing for outbound is to make sure that you are respecting local laws and regulations.

This said, you may want to have a look at our WombatDialer that works well with FreePBX

Oh good, since someone from Loway is here hawking their wares I can ask some vital questions.

Are you guys ever going to update your documentation to not be from the era of 1.4-1.8 of Asterisk? You have highly outdated dialplan examples for your product. You have examples/help documents that still have pipe (|) delimiters which support for was completely removed in Asterisk v13, so like 8 years ago.

To go with that, when are you guys going to actually support Chan_PJSIP? Not a single thing in your website or documentation shows support for Chan_PJSIP. It’s only been about 10 years with Chan_SIP now being completely dead/unsupported with removal in 2023 (Asterisk v21). In fact a lot of things will be removed in Asterisk v21 just like some things have been removed in Asterisk v19. Have you done anything about that?

This is seems to be the only Asterisk third-party commercial solution that hasn’t made a single effort to update their software with the modern versions of Asterisk in the last decade. So when are you guys getting around to that before hawking this some more in 2022?

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I appreciate your optimism, and I’m sure chan_sip will be removed in due time, but there are still a ton of systems out there that are running on chan_sip - wish it wasn’t so. There are people running both SIP and PJSIP, usually because they want to support webrtc but don’t want to touch existing dial-plan. And in any case, even if it had been removed in 2020, we’d keep supporting SIP for a few years because we know that a lot of systems are very infrequently updated. I’m not sure whether it is an asset or a liability, but Asterisk/FreePBX very often works so well that it ends up being forgotten and to live on untouched basically forever.

This said, our defaults for setting up are now all PJSIP based (and have been so for the last, I don’t know, 5 years) while very old documentation might not have been updated, but usually is either dated or clearly marked as “Asterisk 1.4”. When there are doubts, our support services will try and help making things clear.

Do you guys not follow the project? Asterisk Module Deprecations - Asterisk Project - Asterisk Project Wiki. The due time for the removal of Chan_SIP is October 2023 with Asterisk v21. Chan_SIP has been noload since Asterisk v17.

So your QueueMetrics 22.02 release is old documentation? That does mention 1.4 but it’s always “1.4 or higher” there’s even sections with " Enabling ACD call attempts recording on Asterisk 1.0 and 1.2". Again, you’re saying this entire section of your online user manual just hasn’t been updated in the better part of a decade for reasons? About QueueMetrics :: Loway Documentation Center

Is this documented somewhere because I can’t find a single thing to support this outside of PJSIP is used for WebRTC.

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