On 9-Dec-05, at 7:18 AM, Kim Culhan wrote:
On Thu, December 8, 2005 10:43 pm, William Lloyd said:
The analog FXO cards are a nightmare, I would advise staying away
from them. If you need a couple FXO/FXS ports buy a sipura 3000.
Can you post an example asterisk sip configuration for using
a Sipura 3000 for FXO/FXS ?
Found this example config looking around the net:
http://geekgazette.com/index2.php?
option=com_content&task=view&id=28&Itemid=26&pop=1&page=0
The sipura 3000 is a little gem as far as I'm concerned. Unlike the
grandstream models with FXS and FXO, this device treats the FXS and
FXO ports COMPLETELY INDEPENDENTLY.
They are ideal for use at home for example. Chop the bell line
coming into your house. Put the sipura 3000 in the middle. FXO to
bell, FXS to the line going into your house.
When power is off the device has a relay inside to just connect the
FXS and FXO together. I've not check if this can be disabled. worst
case use a knife to cut the traces on the board inside.
When it's operating it sends all incoming calls to asterisk.
asterisk does voicemail whatever it needs and switches the call back
out on the FXS port.
The sipra has an insane number of options that can be tweaked.
generally I just run the factory default with SIP auth ID, password
and proxy hostname/ip. Needs to be setup twice. Once for the FXO,
and once for FXS. Use separate authnames so that asterisk can
identify them. I typically use names like sipura1fxs and sipura1fxo.
On asterisk it's just a simple sip.conf
something like
[sipurafxo1]
type=friend
context=from-pstn
host=dynamic
dtmfmode=rfc2833
secret=xxxxxxx
canreinvite=no
[sipurafxs1]
type=friend
context=from-sets
host=dynamic
dtmfmode=rfc2833
secret=xxxxxxx
canreinvite=no
I make the password the same in both ports.
The first digium card was ok after I disabled most of the stuff in
the bios. Adding a second card required digium to remotely log
into the
machine to help get it working properly. On the plus side the
free Digium
tech support for hardware was excellent.
Can you post to the list the changes Digium made
to get that working ?
This was with Quad PRI cards not analog.
Sorry, can't help you. They did it remotely. They played with
hdparm I think. Also zttest and zttool. I didn;t dig into it. The
client was breathing down my back to get the system upgraded and back
online. when you have multiple PRI's offline....
Also, were you running FreeBSD when Digium
remotely logged into the machine ?
I run FreeBSD on my laptop and anywhere else I can use it.
Unfortunately I don;t think it's stable enough for me to use in high
end production _asterisk_ servers that I put in the field. so the
answer is no ;-)
There are so many little issues that can come up in VOIP on a
"supported" hardware platform like Linux that it's not a battle I'm
prepared to fight at this point.
I have a TE110P spare card that I would like to try in a freebsd
machine if I have time. I've been hoping todo that for a long time
so far.
I'm a big FreeBSD advocate but you gotta do what makes business sense.
I'm running asterisk on a FreeBSD soekris box with a sipura 3000 at
home for example.
I have a couple of TDM Digium cards that I purchased. I've long
given up on them. they are buried in a box of old cards. One time I
walked across the room, picked up a regular phone hooked up to the
FXS for of the TDM card, I got a little static shock from the
handset. Asterisk server goes dead. Rebooted it. It' came up but
the FXS port was dead from then on. It's not telecom quality gear.
the echo problems were horrendous. I played with the echo cancelers
for hours.
In fairness, I've done the same thing to Sipura 841 phones. I bought
4 six months ago and 2 are dead already. 1 for static and the other
just refuses to boot past a certain point for some reason. They are
total junk also. the 941 look nice but I've none yet.
for my own desk I use Cisco 7960 or 7940. I love them. The 7960 has
6 line buttons Each goes to a different asterisk server for testing
or whatever.
-bill
tnx
-kim
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