Dahua VTO2202F "Villa Door Station" -- SIP Groups & other weirdness

[Sean Phelan]

I have been experimenting with a Dahua VTO2202F (PoE version) for a week or so. I've found a few nice features, several missing features and some stuff that is so weird that I think I must have misunderstood what's happening.

The door station hardware is elegant and looks like it will last, and I like the simplicity of just one single push-button.

The firmware supports much of the standard Dahua http API, including snapshots and subscribing to an event feed. I was a bit disappointed that SNMP and Syslog are missing: they are not in the web interface and if they are in the API then the names have been changed.

A more fundamental missing feature, as far as I can see, is that pushing the doorbell button does not seem to generate an event - certainly not one that appears in a "Subscribe to Event Message" request for All eventCodes. Is this really the case? Am I missing something?

Another feature that I do not understand is the SIP "Group Call". The documentation talk about "calling every extension" and the firmware tries to number multiple extensions with a '#' suffix, along the lines of "9901#0, 9901#1, 9901#2" etc. This is not part of the SIP URI specification, and although I don't think that strictly speaking it is a protocol violation, it doesn't seem to be supported by any other SIP clients I can find (my Siemens/Gigaset basestation will accept a '#' in an address, a Linphone SIP client will not).

Also, I'm not convinced that the VTO (door station) does make group calls, in spite of claiming to do so. I suspect that one of Dahua's VTH stations acts as a master and additional stations are connected to the primary VTH and not to the VTO. This, of course, would make it difficult or impossible to use anything other than Dahua's own client devices if you want to have the door station call out to more than one VTH. I hope I am mistaken about this; any and all advice will be very welcome !!

I also have a problem with one-way audio (i.e. works to VTO but silence the other way) when connecting to the Siemens/Gigaset VOIP basestation. Apparently one-way audio is a common problem with SIP/VOIP services, and is usually caused by a firewall or by codec incompatibilities. The devices are all on the same LAN segment so I don't think it's a firewall issue. And the Gigaset contains all the common codecs. Two-way audio works properly using a Linphone client instead of the Gigaset. Again, any and all suggestions will be appreciated!

Thanks all !!
Sean

 

[The Automation Guy]

What are you using for your SIP server? Does it allow for programming a single extension number that really calls multiple extensions? My phone system (PBX in a Box - based on FreePBX) allows for this and I actually have it set up to "intercom" multiple extensions. I have the softkeys on the phone programmed for this. So one softkey dials the extension that will intercom call all extensions, another one might only call the all of the kids bedrooms rooms and playroom, etc, etc, etc. If your SIP system allowed for this, it would be easy to have the VTO call one of those programmed extensions that really calls multiple extensions at once.

 

[Sean Phelan]

I have been using the VTO's internal SIP server. I was thinking about running a separate server; I've taken a quick look at Sip Witch, but I'll also look at PBX in a Box.
Your approach -- client calls a single number that triggers the server to call multiple numbers -- sounds practical and straightforward; I'll try that.
Thanks
Sean

 

[The Automation Guy]

Honestly the intercom feature is the most used thing. We certainly don't get any calls on our home phone (especially since I created a "anti-spam" IVR that answers any incoming call to the house and asks the caller to press "1" to be connected). But having the ability to call one or multiple extensions and have the phone on that extension pick up automatically in speaker phone mode is a game changer. This allows people to hear and respond without getting up and answering the call or pressing a "talk" button to reply (like my parents 70's style intercom system). Now of course that still doesn't mean that my teenage kids will actually respond and say something back, but that's a problem I can't fix with technology!

This, along with the fact that phone service only costs us about $3.50/mo (including e-911 service), is why we have kept our phone system.

PS - I'm happy to answer any questions related to PIAF or FreePBX, etc. I've set several systems up (including one for a former employer), and while this isn't my actual job, I'm happy to help anyway that I can.