Trixbox + Old keyset (Merlin 410?)

Hi all, I am new to VOIP & TrixBox but comfortable with Linux and simple telephony. (I can punch a T1 down on 66 block ) A friend gave me an old AT&T Merlin 410 keyset with 6 extensions to play with.

It uses PSTN trunks, and digital (ha!) phones. However handy it is it doesn't have caller id or voicemail, what I was thinking was the following:

1. FXO for our main PSTN line with failover incase network or power dies
2. 1 FXS for each trunk on the keyset (4) (?)
3. TrixBox to route inbound calls based on callerid. Friends & Family on line one, work on line two etc. Jerks and telemarketers get sent to evil recursive menu options.
4. Finally set up a voip account to dial out in the rare event the main line is tied up.

To that end I was thinking a Linksys SPA-3102 would be the ticket and maybe a Linksys PAP2T for a few extra trunks.

Mind you this is really just to play with so I don't need to blow a million bucks. (My budget is about $150.)

Any thoughts?

Well, I have some reporting to do. Firstly, Trixbox is all set up and the system works.

I do have some problems:

1.) Calling out on the PSTN line introduces echo that is not present on sip to sip calls. It seems like echo cancellation sorta works, but it pops up annoyingly some times. I have fiddled with the gain to no avail.

2.) I want all calls preceded with a 9 to dial on the SPA3102 trunk. It mostly does this, except you have to dial very quickly. If you dial 9 and then pause I just get a busy signal. My outbound route dial pattern is "9|."

3.) Is there a way I can just have callerid's that don't have a name, or matching "unknown" go straight to voicemail? Black list works but seems like that'd be fairly tedious.

4.) Inbound DTMF from the SPA-3102 does not seem like it's getting to the Trixbox. i.e. using voicemail.

TIA

Hi, Sorry I missed you the first time around, but we'll try and get you around the first few problems:

1/ Echo on the PSTN is always a potential problem.
In this case your SPA is connected to Dear Ole BT, so you need to set cancellation there. Also check the line voltage, BT should be 48V (but isn't always) and Linksys like setting the SPA at 28.

2/ Again this is a setting in whatever you are using for dialing.
Most ATAs have a timeout setting for dialing and this should be set to about 4 seconds, possibly more. This gives plenty of time to decide what to dial next. Others have an immediate dial function and that should be turned off.
Also make sure you have the 'Dial String' enabled, or you will die of boredom before it dials, this is normally set as #.
Other ATAs allow/require complex dial patterns to be set. Make sure these are muted to the simplst possible ruleset- You want everything to go to Trixbox

3/ Try setting the CID entry in Inbound Routes to Unknown and annonymous.

4/ Check the DTMF mode is set the same at both ends. Think Linksys prefers inband RFC2833

rgower :

Hi, Sorry I missed you the first time around, but we'll try and get you around the first few problems: 1/ Echo on the PSTN is always a potential problem. In this case your SPA is connected to Dear Ole BT, so you need to set cancellation there. Also check the line voltage, BT should be 48V (but isn't always) and Linksys like setting the SPA at 28.

In this case it's Dear Ole Verizon The SPA reports a line voltage of 50V. Oddly however, I just did a yum -y upgrade and the echo thing went away. (Asterisk didn't update, just webmin, tar, and net-snmp.)

rgower :

2/ Again this is a setting in whatever you are using for dialing. Most ATAs have a timeout setting for dialing and this should be set to about 4 seconds, possibly more. This gives plenty of time to decide what to dial next. Others have an immediate dial function and that should be turned off. Also make sure you have the 'Dial String' enabled, or you will die of boredom before it dials, this is normally set as #. Other ATAs allow/require complex dial patterns to be set. Make sure these are muted to the simplst possible ruleset- You want everything to go to Trixbox

That did the trick, thanks!

rgower :

3/ Try setting the CID entry in Inbound Routes to Unknown and annonymous.

Now to see if that does it. Verizon has been nagging me of late to update my cell phone contract. Grrr.

rgower :

4/ Check the DTMF mode is set the same at both ends. Think Linksys prefers inband RFC2833

On my PSTN page I have:
DTMF Process AVT: yes
DTMF Tx Method: InBand+INFO

Line 1:
DTMF Process INFO: yes
DTMF Process AVT: yes
DTMF Tx Method: InBand+INFO
DTMF Tx Mode: Strict

Under SIP
DTMF Playback Level: -16
DTMF Playback Length: .1

Under the SPA trunk:
dtmfmode=rfc2833

Any other places to check? Thanks for the help. Much appreciated.