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While I had good success with the setup I documented in 2018, I have since moved on to other projects and have not spent any time further refining this. I plan to do so and will post again when I have had some more time to work on it.

Time for some new open-source VoIP projects to come along.

It works if you share the core db... I had it working using the notes provided in my thread, but no one else can seem to believe or reproduce the success?

Many UDP SIP packets were truncated, cutting off important information in INVITE packets. We worked with the homer team to get this solved but it seems their capture agent struggled with fragmented packets no matter what settings we used. We run video in our network and homer failed to produce...

Homer is so buggy and frustrating, we went commercial and bought VoIP Monitor (voipmonitor.org) - and haven't looked back. Spend the money and enjoy your life more.

Yes use SIP SRV records for the load sharing/balancing on each SIP domain; the individual hostnames are different. I think DB is the most sensitive part. Would be nice if there were a failover/LB option for DB connections. With MySQL I have used mysqlproxy and mysql-router in the past. I don't...

It means if you want the "guaranteed" route then your dial string needs to start with 999001. Start with 999002 for the other route. Whoever this provider is, don't use them, because this scheme sucks.

Can you elaborate more on this? How does it work?

Hey Roget, I just found that FusionPBX has this built in - some other guys were discussing it on the forum. I guess it's a hidden / undocumented feature. Do a search for recent posts on the subject of fax.

Hey, I didn't know this existed. Did you guys follow documentation or a tutorial to make this work? Looks like it's not on the docs site. Other than what you've mentioned above -- what else do we need to know?

The regex (FAIL) lines are just showing call routing logic, letting you see the decisions the dial plan is making. They don't represent an error or major failure. For example This FAIL is just saying that in this call the user_record variable does not match "all" thus this particular route in...

Could you please explain which number is the DID and which number is the PSTN number you're forwarding to? Also cut some of the extraneous stuff out of the log so that it is readable. I do not see an error in the logs above.

It uses the speed dials in your Contacts. You set up a contact and when you add their number, also add a speed dial code. Then you can speed dial that contact's number with *0[speeddial]

The phrase method works for me. You may need to look at your logs to find out why it isn't working for you.

Hmm... Make a "phrase" using the Phrases application. Then set up your Destination to first play the phrase then go to your ring group or extension.

How do you sell your software when there is no page on your site for making a purchase?

Looks like it would have to be a feature request. The extensions screen doesn't give the option. According to the FreeSWITCH wiki you would need to specify the parameters "reverse-auth-user" and "reverse-auth-pass" ; see https://freeswitch.org/confluence/display/FREESWITCH/XML+User+Directory in...

How is the phone authenticating the INVITE?

Yes, it's easy. Just add H264 to the global_codec_prefs variable and then flush cache and reload your SIP profiles.

I have done this on a different platform. It is complex. Here are the steps in a nutshell. Set up an e-mail domain and parser. If you want a shortcut, there are services like Sendgrid that will receive e-mail for you and deliver them to you by HTTP API with all the parts separated out into...