ALERT Grandstream UCM6100 PBX

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That's certainly correct in terms of use by the copyright holder; however, Grandstream may not be the copyright holder. Thus, their obligations may be governed solely by GPL2.

Certainly, Digium could grant a license for future commercial use of their code. Changing the nature of a license on previously released code may be problematic if the GPL2 terms already have been triggered.
Hey Ward -

Prefacing everything with "IANAL":

We discovered that the Grandstream UCM6100 user interface appeared to be based on the Asterisk GUI as well. Grandstream does not have a commercial license from Digium to distribute a derived work of the Asterisk GUI outside of the GPLv2; hence, it is our position that they have to abide by the GPLv2 for the "program", i.e., the user interface for the UCM6100. We have contacted them regarding this issue; I've received verbal confirmation that they will make the necessary corrections.

Matt
 
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wardmundy

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Hey Ward -

Prefacing everything with "IANAL":

We discovered that the Grandstream UCM6100 user interface appeared to be based on the Asterisk GUI as well. Grandstream does not have a commercial license from Digium to distribute a derived work of the Asterisk GUI outside of the GPLv2; hence, it is our position that they have to abide by the GPLv2 for the "program", i.e., the user interface for the UCM6100. We have contacted them regarding this issue; I've received verbal confirmation that they will make the necessary corrections.

Matt
Thanks, Matt Jordan. Making "the necessary corrections" would certainly appear to be in order. Publishing their embellished code in compliance with GPL2 also would be a nice touch.

"IAAL"
 

wardmundy

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Default privileges are a GEN-U-WINE train wreck. Just wait 'til those phone bills start rolling in for their customers. You can lead a horse to water, but...
 

jmcman

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OK, so I attended their webinar this morning on the product. Here's my synopsis...

It was mainly marketing with a little bit of technical fruit thrown in. I wrote down some things from the Q&A section that I thought may be of help or interest to others:
Q: How many UCM's can be linked together?
Up to 50 (SIP trunks) total. To connect different ones together, you use SIP trunks. So basically, up to 50 units.

Q: Is the CDR externally available?
You must enter the PBX and manually pull them out as an excel file. Not externally available presently, but it is in development.

Q: Do you support FAX to e-mail?
Yes, it will create a PDF file with the contents of the fax and e-mail it.

Q: Is hot desking available?
No, not currently.

Q: Is call recording availble on request (pressing via a softphone button)?
Yes it is. Softcode is *77

Q: Do you support dial-by-name or voice recognition IVR features?
No, there are no voice recognition features.

Q: Is there direct access to Asterisk via SSH, etc?
No, we are actually blocking these connections to asterisk specifically. There is a console available via SSH, but it is mainly to force an update of the device and do limited functions.

Q: Does your PBX include ringback if a call is rejected on a mobile phone?
If you are doing a call forward, yes you will get a ringback.

Q: Can you stack multiple units to just add more FXO ports?
Yes, absolutely. They are scalable in that regard.

Q: Can you backup one UCM6100 and use the configuration on another unit?
Sure, you can certainly configure one UCM6100 device and use it as a template or restore it for another installation. The backup restore is limited to the UCM device series.

Q: Is instant messaging available through the PBX with XMPP?
No, it is not currently.

Q: Is there an ISDN version?
There is not currently, but you can use an ISDN gateway paired to the PBX.
Another note is that all models can be powered by PoE, simplifying cabling.

Sadly, but unsurprisingly, they did not answer any of my questions. :lol:
Q: Shouldn't one of the pillars be security? I don't see anything about security... (The four pillars of the product are listed as voice, mobility, data, video)

Q: Did you hire an asterisk guru or team of gurus specifically? Or did you just reallocate existing internal manpower to develop this product?

Q: Is there any way to contact the engineering team directly for feature requests, enhancement requests, or serious security issues?

Q: Are you going to the release full source code for the product so it can be evaluated by others in the open source asterisk community?
Now, question 2 is of some significant importance to me personally. It makes a big difference whether you went out and hired some asterisk big shots to make a great product, or whether you decided some kid in your engineering department who can cook up an embedded asterisk install got recognized and the management decided to make a product out of his tinkering.

I'm still leaning towards this thing is a big, insecure nightmare just waiting to happen. :beatdeadhorse5:


At this point, I still cannot recommend this platform to anyone yet.

 

krzykat

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They're not making money off of the software. The software is there as a conduit to allow them to sell their hardware. I WISH !!! they would allow you to USE their hardware and package it with whatever software (IE PIAF) you want on it. It would OPEN their market share, not shrink it.
 
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wardmundy

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Release 1.0.2.17 for UCM6102/UCM6104/UCM6108/UCM6116 is now available for BETA test. This release has added the following new features among others:

INCLUDES 100+ BUG FIXES!

• Added DISA function
• Added Eventlist function
• Added LDAP synchronization for SIP peer trunk
• Added NTP server function and manual time settings
• Added pickup group function on extension page
• Added SIP "authID" support
• Added Remote-extension BLF
• Added VLAN Support

The firmware and release note can be downloaded from: http://www.grandstream.com/support/firmware
 

wardmundy

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I'd give mine to Tom to "improve" but I'm still playing. The good news: :santa: is just around the corner.

P.S. Found a couple of bugs that you may wish to review before you upgrade. I've gone back to the previous release.
 

krzykat

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If you get it working, you need to have GS pay you for the increase in sales that I guarantee they will see from that. Maybe get them to do a branding for PIAF and put some money into the PIAF development pot. Or at the minimum have PIAF be a master distributor and do it that way :)
 

wardmundy

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Just wanted to document how to get sip2sip.info INBOUND ONLY working with the UCM6100. You'll need an OBi202 for openers.

Outbound calling through sip2sip.info doesn't work unfortunately because sip2sip expects a SIP URI call in the following format: [email protected], and there's no way to do that on the UCM6100 presently.

In the OBi GUI Setup, you have to set a Service Provider ITSP Profile C (you can use any of the available letters) and a Voice Service SP3 (use any you wish).

For the ITSP Profile C General:
Code:
Add 223.| to the beginning of the DigitMap
Add sip2sip as the Name (in both places)
For ITSP Profile C SIP Settings:
Code:
ProxyServer = sip2sip.info
OutboundProxy = proxy.sipthor.net
For Voice Services SP3 Service:
Code:
X_Enabled=checked
X_ServProvProfile=C
X_RegisterEnable=checked
X_KeepAliveEnable=checked
X_AcceptResync=with authentication
 
AuthUserName=223XXXXXXX
AuthPassword=yourpassword
URI=223XXXXXXX
UCM6100 Setup goes like this...

For inbound calling, you'll first need to set up an Analog Trunk for the OBi. We used Channel 1 and called it OBi.

Next, set up an Inbound Route for your OBi Analog Trunk and point it to some extension, IVR, etc.

For grins, you can set up an Outbound Route (but it won't work as of this firmware). The trick here is to use a dial prefix (7,8,9, whatever) and then Strip off 1 digit. For OBi calls, you also need a Prepend (in our example, it would be **3 to tell the OBi to dial out through the SP3 service). The dial string should be 7223NXXXXXX where 7 is the dial prefix.

After you Apply Changes, you should be able to receive calls from other sip2sip numbers or from your sip2sip URI.
 

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