Usage of Telecom

[Marie.O]

I wonder how many people are using the telecom (asterisk) part of LinuxMCE. How do you like it? Does your HDD fill up with all your old mailbox messages, or do you clean them out.

Inquisitive minds want to know.

Thanks

rgds
Oliver

[bulek]

Hi,

I'm trying to use it daily, although have always a bit of hazardous expectation whether Asterisk will die or not (since I'm using Cisco 7970 this happens quite often)...

IMHO, call routing is a bit too complex for user to setup. I usually create two "dummy" users, Housephone and Doorphone and then reroute incoming calls to them and use only call routing for them...

What I spot is that voicemails are currently not sent by email and are also hard to delete... I have an separate plain old POTS line with telephone in parallel to Cisco 7970 and must admit that members of my family still like old phone more... Didn't figure it out why...

I do like possibility to connect to Asterisk from work and receive calls normally as being home...

Regards,

Bulek.

[samuelmukoti]

I love the Asterisk/Telecom features that LMCE brings.  I immediately bought Grandstream GXW4104, and set it up for my analog calls (since i dont have VOIP service in my country) i also got a Broadvoice account all my international FREE calls.  I'm loving it.

Can't wait for my N810 to arrive so i can use it as a SIP Phone too...

I would love it if some things worked better, and whole experience was better, e.g. how incoming calls are handled when watching a movie etc..

But for now, i absolutely love having Asterisk bundled with LMCE!

[tschak909]

Right now,  certain features are only available with the SimplePhone on the media directors. Namely, since SimplePhone is a DCE device, it can emit DCE events to tell the system to pause a movie, etc. This is not currently possible with phones connected to the Asterisk DCE device. It's made more interesting by the fact that not all phones emit reliable off hook information in a vendor-neutral manner.

As of now, I attach a Phoenix Solo USB conference mic to each media director, and use that as my phone, dialing from the orbiter as appropriately.

-Thom
 

[golgoj4]

1. Use broadvoice every day with a linksys ata, works great
2. I setup my pots line, but i dont use it, was more for testing

All in all, it works pretty well IMO, there are some small bugs but no deal breakers.

-golgoj4

[Marie.O]

[..]As of now, I attach a Phoenix Solo USB conference mic to each media director, and use that as my phone, dialing from the orbiter as appropriately.

Did you modify the SimplePhone app to support the USB mic?

rgds
Oliver

[tschak909]

yeah, I still have some work to do to make it packageable for others, but yeah.

-Thom

[golgoj4]

there was ONCE that i was able to use regular mic's connected to the 1/8 input that worked on md's but not the core/hybrid. Very odd...

[bulek]

...

As of now, I attach a Phoenix Solo USB conference mic to each media director, and use that as my phone, dialing from the orbiter as appropriately.

-Thom
 

Hi,

thanks for info. That reminded me also that I'm searching for the mic that could be attached under TV's MD and be usable for telephony. All ordinary micrs for only on max 1 m distance. Are there any other recomendable microphones that would pick your voice from couch approx 3m away at decent quality ? 

I remember reading about Crown PZM mics and doing speech recognition in smart home environment with them, but AFAIK, they we're connected to really expensive conference amplifier...

Thanks in advance,

regards,

Bulek.

[tschak909]

As I said, now I am using a USB conference mic. In my case, the Phoenix Solo.

I still need to do a number of modifications to the SimplePhone to make it more hospitable for normal use, but it is possible.

-Thom

[skeptic]

I pretty much don't use my home phone (POTS), but keep it around for faxing.  I've been meaning to switch to asterisk/LMCE but it doesn't appear to be a simple 5 minute setup, and I've been distracted with MAME and possible UI stuff.

I guess I'm just posting because while I voted "don't use it" I am interested in it and I'll be following any new related threads.

[golgoj4]

I pretty much don't use my home phone (POTS), but keep it around for faxing.  I've been meaning to switch to asterisk/LMCE but it doesn't appear to be a simple 5 minute setup, and I've been distracted with MAME and possible UI stuff.

I guess I'm just posting because while I voted "don't use it" I am interested in it and I'll be following any new related threads.

if you have an ip phone, i would actually file it as a simple 5 min setup.
my linksys ata took no time to configure as an extension
my sip clients on computers took more time to install than setup.

A couple things i am interested in and have been trying to figure out on my own

1. How do i integrate faxes? Asterisk seems to be capable of this, but im learning it slowly as im doing a lot a once.
2. Changing the default message to one you record / upload. I have tried this multiple times, but it appears broken. Italics are there because im more than likely missing something.
3. Thom mentioned about the 'off hook' bit. Which really doesn't concern me terribly.

[skeptic]

hmmm...  Maybe i should take another look at it then.

[Marie.O]

1. How do i integrate faxes? Asterisk seems to be capable of this, but im learning it slowly as im doing a lot a once.

There are a couple of ways. hylafax is one way, recvfax is another project tightly integrated with Asterisk. How ever, keep in mind, if you receive your calls over VoIP, you won't have much success with faxes.

2. Changing the default message to one you record / upload. I have tried this multiple times, but it appears broken. Italics are there because im more than likely missing something.

How did you try to record new greetings, and what greetings are these?

rgds
Oliver

[dlewis]

I've seen many posts around the issue of having a second firewall layer (before the LinuxMCE firewall) and having issues with Asterisk. I've read a few docs on the net and I think I might have some up with a possible "solution"... (DISCLAIMER: I have not tried this myself since I don't have LinuxMCE currently set up. It might not work, so try at your own risk) Thanks goes to the voipcoop.org guys for supplying this information:

If you are behind a NAT firewall, you may need to open the following ports to your FreePBX box: For IAX protocol, UDP (NOT TCP) port 4569 (IAX). For SIP protocol, UDP (NOT TCP) port 5060 (SIP) AND ports 10001-20000 (RTP, must also be defined in /etc/asterisk/rtp.conf). All these ports are *UDP*, opening the *TCP* ports will NOT help anything and may expose your system needlessly.

If behind a NAT, create an /etc/asterisk/sip_nat.conf file with AT LEAST these three lines: 1) nat=yes 2) externip=your.external.IPaddess (or externhost=your.external.hostname) 3) localnet=192.168.0.0/24 (assuming your network uses 192.168.0.x addresses). Then "asterisk -rx sip reload" at the CLI.

For firewalls in general, go here: http://www.voip-info.org/wiki-Asterisk+firewall+rules

This would seem to work for SIP, but I don't know about SCCP. Can anyone try this and let us know? Thanks!

-D