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CSG SIP Comm Gateway CSG-500

We just received (probably the first) SIP Comm Gateway. Documentation is very limited or I just cannot found it. I do have the SIP Instruction manual, but is it very weak. So far, I am not able to ping the box when I it is tied from the units network port to a switch. This network is a local net for testing and building only and so far not internet accessible. I do have two 8400i TPs, and 3 NIs on the local net. Soon, this test station will have internet access. This SIP unit is set to default setting, and the DHCP is enabled. Any clues?

From the manual it sounds that the Touch Panel Protected Setup should automatically have the SIP button in the ?Other Settings? options. Since I cannot ping the unit, I assuming the SIP button will not appear? Am I correct that the SIP button should automatically appear when I start communicating with the unit?

So far, I have not found any documentation on how the AMX program is going to control this box (make a phone call from the code).

The only way I am currently talking to the Box is through the web-based configuration tool when it is directly tied to my PC.

Does anybody have any information that can help? Where should I go for documentations? I saw nothing under AMX Search devices web page.

Thanks,
Brian.

Comments

  • briffeebriffee Posts: 2
    CSG-500

    I am know able to ping the unit on my local net. I still do not see any SIP button showing up on the TPs. I found out that we are 1 of 5 companies trying this box out. And, That I will be receiving directions from AMX.
  • You'll find the Instruction manual here:
    http://www.amx.com//techdocs/CSG-500.544.580.InstructionManual.pdf
    We waiting for our 1st unit, so I can't give more information yet, sorry.
  • viningvining Posts: 4,368
    From the CSG-500.544.580 Operations Reference manual.
    The options on the SIP Settings page (FIG. 3) enable you to establish network settings for using your
    touch panel as an IP phone. You may need to load a Duet module to enable the touch panel to receive SIP
    calls. The Duet module translates between the standard interface and the device protocol. It parses the
    buffer for responses from the device, sends strings to control the device, and receives commands from
    the UI module or telnet sessions. Refer to the documentation supplied with the Duet Module for more
    details.

    I like the line from above "You may need to load a Duet module to enable the touch panel to receive SIP
    calls.".

    I would think they should know and it would say you either need to load a Duet Module or not say anything at all. Of course I can't find a module for it anywhere that would allow calls to a CO or anything specifically supporting the SIP Gateway. There's the intercom module but the one I found was a standard Netlinx module.

    For set up material and more documentation I found good documents here:
    https://www.digium.com/en/supportcenter/documentation/viewdocs/AA50
    you'll need to create an account to access them but at least they have all the documation on these devices although I don't know what's accessable in this AMX packaged version.

    I plan on playing with this gateway when I have idle time to see what tricks it can or can't do. My thinking is this will most likely become a standard in every resi system so I want find out the pitfalls and shortcoming in this AMX version of the Digium AA50 before it needs to be installed somewhere and actually work.

    Anyone got one working? Any tips? Any complaints? Anything at all? What phones? Polycom?
  • Spire_JeffSpire_Jeff Posts: 1,917
    The other thing I would like to know is: Has anyone tried using an AMX touch panel as a station on a phone system with SIP support?

    Jeff
  • viningvining Posts: 4,368
    Spire_Jeff wrote:
    The other thing I would like to know is: Has anyone tried using an AMX touch panel as a station on a phone system with SIP support?
    That was the point of the quote I posted that says you "may" need a Duet module to make calls. I would assume the TP would just be another VOIP client and a module would be required to handle the dialing and what ever logic a VOIP phone would have.

    The Digium documentation is fairly good as far as the PBX configuration goes but I don't know how much of their original web interface remains after the exterior shell change and the AMX web interface installed.

    I have the CSG-544 sitting on my desk but I've yet to put it on the network and log in but I'm hoping this abitlity to intercomn and make phone calls on the TPs will be easy and bullet proof. I played with the intercom part and that works well so we'll see.
  • Spire_JeffSpire_Jeff Posts: 1,917
    I wonder if they are saying you "may" need it because you don't have to use the duet module that is available if you roll your own code? A better way to put this would be along the lines of: For ease of use, you can use the duet module. If you prefer custom functionality, the protocol is available to allow custom programming.

    Just a thought,
    Jeff
  • travtrav Posts: 188
    SIP Stuff

    First off, let me point you in the direction of this post which answers some of the questions raised.

    As far as controlling calling from the netlinx, if you have the SIP enabled firmware for the panels, then it is simple as catching the numbers pressed on the touch panel in code, and then sending the entire string back to the panel as a send_command to get it to dial! As easy as a microwave dinner. (but tastier)

    As far as your other question, has anyone used it with a SIP enabled phone system, yes, I've got an asterix server in a VM and it registers directly with that. We've used a NetCom SIP phone, very basic bit of kit, and found that if that registers with the server, then the voip panels _generally_ will as well.

    Good for corporate installations, but again, we have found that they generally want the AMX stuff off their main IP PBX. So it is connected to a CSG then a SIP trunk to their PBX with the appropriuate dialing rules.

    The Latest firmware for the 1000vi, and the 700vi Have the SIP stack in the firmware, the 8400i is awaiting a release with the SIP stack + network fixes rolled in, and erm, not sure about the 500vi and the 5200i.

    The duet module kinda fakes the SIP stack (a poor analogy but it will do) for intercom purposes. But if there is SIP enabled firmware there isn't any need to use the Duet module.
  • viningvining Posts: 4,368
    Trav,

    thx for the insight and the link. I hope to get a handle on this during the holidays but need to get a couple of phones first. I have some VIOP intercom stuff from Digital Acoustics that I'd like to incoporate with the SIP but that needs a SIP software app to run on a PC to act as a bridge.

    Are you using an AMX piece or a Digium Asterisk Appliance?
  • travtrav Posts: 188
    AMX Chat

    Jump into AMX Chat and I can fill you in a bit more...
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