Uniden UIP1868 "Whole House System" and Asterisk?

[ZED28]

Originally Posted by gmsmith

UM...wow...

Thanks for the response. I appreciate the details.

I am having a problem having it recognize DTMF tones during calls, like for example voicemail password and such. I have dtmfmode set as RFC 2833 in the sip config on asterisk and I set out of band telephone events to: RFC 2833, am I missing something?

Now its time to cancel my ebay auction and keep this puppy.

I had the exact same problem. I also tried the same settings that you did but it never fixed it. Unfortunaltely, I never persued the problem when I found out the phone wasn't going to let me assign an extension to each handset so I returned it at that point.

I loved the phone, don't get me wrong. In fact, I purchased the same POTS versions of the handsets and am using them with ATA's instead. The speakerphone on these puppies are awesome!

Did you manage to get two lines registered? The thought process I had was this...

The phone came with one handset and a base so seeing two lines for extensions in the SIP settings seemed to point to each phone having its own extension. I had a 2nd handset that I added to the system and was hoping to see a third. But as stated above, the minute I had mine connected to my Asterisk box, I only ever saw one line to use for an extension, it removed the 2nd line all together.

BTW, I called Uniden for support on this phone, their response was interesting... "we only make this model phone to work with Packet8 or Vonage, the one you have is an experimental model that was never supposed to be sold to the public". So, no support from Uniden! I believe they're more likely to be Packet8 phones that have been "unlocked" but that's just my theory.

But none of this is helping you with your DTMF issue! Sorry bud, wish I could be more helpful but I just didn't persue it enough. I would be interested to find out if you ever fix it though and if you got the 2nd line to appear.

[gmsmith]

Only did one line...was just planning on using it to replace the current multi-handset base we have (some old crappy AT&T thing). I will play a little bit with the DTMF issue and perhaps call in a chip at Uniden (family friend blah blah blah), though the DTMF thing shouldn't be a big deal to resolve. Even my wife loves this phone...so I need to take advantage of that to push the move to VoIP in the house

[ZED28]

Yeah, my buddy is the one who got me looking at Asterisk about 6 weeks ago. We already have AT&T's Call Vantage with two lines which is a pretty okay solution but figured I could do a much more versatile thing with my own PBX.

Well here I am, about $1000 later and am almost ready to cut off AT&T! It was a hard sell to my wife who kept saying "let's just go back to POTS!" but it's a sweet setup, everyone has their own phone (2 kids, the wife and me) and hopefully, 4 seperate DIDs as soon as TelaSIP can get them to me.

[gmsmith]

Finally got a chance to look back at this...

dtmfmode=auto

And all is well with the world. Time to buy a few more handsets

Greg

[mfflynn]

"dtmfmode=auto"

Does this go in the asterisk configuration file for the extension or does it go in a phone setting somewhere (if so, where is it)?

Would this problem cause the asterisk server to not recoginze the tones for someone logging into their voice mail?

Thanks guys.

[gmsmith]

Originally Posted by mfflynn

"dtmfmode=auto"

Does this go in the asterisk configuration file for the extension or does it go in a phone setting somewhere (if so, where is it)?

Would this problem cause the asterisk server to not recoginze the tones for someone logging into their voice mail?

Thanks guys.

dtmfmode=auto goes in your asterisk configuration for the specific extension. You should have a dtmfmode= already in the extension, just change it to auto.

And it does specifically that, allow DTMF tones to be heard while on a call.

[mfflynn]

dtmfmode=auto goes in your asterisk configuration for the specific extension. You should have a dtmfmode= already in the extension, just change it to auto.

And it does specifically that, allow DTMF tones to be heard while on a call.

I modified the line "dtmfmode=" to "dtmfmode=autp" in the sip.conf under the extension the phone is setup on and restarted the asterisk server and the phone. Things like entering your voicemail password on that phone do not work. Is there another conf file that has that switch in it?

[gmsmith]

make sure you don't have a typo there

dtmfmode=auto (not autp)

[mfflynn]

Typo in the reply, not in the config. So I ask again:

I modified the line "dtmfmode=" to "dtmfmode=auto" in the sip.conf under the extension the phone is setup on and restarted the asterisk server and the phone. Things like entering your voicemail password on that phone do not work. Is there another conf file that has that switch in it?

[mfflynn]

I still haven't solved this problem are there any other suggestions?