I found a sap400, wich is FXO and seems to work fine with asterisk. It
offers the posibility to plug 4 analog phonelines.
Thanks again.
Kosa
- Un mundo mejor es posible -
Ioan Indreias escribió:
Both SPA2102 and SPA9000 have FXS ports. You need to use SPA3102 (or
other ATA which have FXO
getting one device to plug both is
fine, but two devices to connect one by one would be good too. The
cheaper the better this time.
Thanks in advance.
--
Kosa
- Un mundo mejor es posible -
--
_
-- Bandwidth and Colocation
to unlock the spa2102 with no succes at the moment, any
links or hint will be very appreciated.
I'm and absolute newbie on asterisk, btw.
Thanx!
Kosa
- Un mundo mejor es posible -
--
_
-- Bandwidth and Colocation Provided by http
Most of the times it works, other times phone on FXS rings, I pick it
up and all I get is a dial tone.
this might also be MWI in SPA-style.
regards
adam
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Hi,
i've just finished setting up gtalk connection with asterisk. it works
nice, audio is full duplex.
i just have one question which i could not find an exact answer to. Is
gtalk able to send dtmf codes? Because i'd like to listen to my
voicemails while away from home.
I've been googling
permit udp any host 192.168.5.0 range 1 2 and then I didn't
home users typically use /24 netmask. If this is the case, i don't
understand why do you write keyword host following a network address.
either specify a valid host address, or write 192.168.5.0 255.255.255.0
to specify the
Hi,
I'm currently looking at FXO options to provide a POTS line to Asterisk to
trunk calls with.
i've had some problems setting the disconnect tone correctly to my
country. As a matter of fact, i still do, as the calculated values does
not always hang up the phone.
Other than this i
Marc Patino Gómez wrote:
in most cases it works well but, if my internet connection is down
Asterisk tries to Dial voipprovider, but it can't resolve the dns name,
so it waits 60 seconds to jump to the following priority...
Any ideas to solve this problem? I can't use the IP of the
Paul Hales wrote:
Is there a way to specify multiple email addresses in voicemail.conf for
a specific user?
why don't you use the /etc/aliases file for this purpose?
regards
adam
___
Sign up now for AstriCon 2007! September
Hi List,
i've set up a cisco 7912 for my asterisk box. I've had problems with
VAD and CNG. After googling a bit, i've found an article about asterisk
not supporting these two protocols, therefore it's better to turn them off.
Since then i did not found answer to my two questions, maybe
Greetings,
i've been posted a message to this list in july, which had one response.
Thanks for that idea! Unfortunately asterisk is only a hobby, and did
not have much time dealing with the problem since. My original letter
was long, i wouldn't post it again, the archive url is
Hi guys,
sorry for the long e-mail, i'm only trying to give as much information
as i think is relevant to my problem (console log, sip.conf and
extension.conf parts). I've sent this e-mail a couple of days ago, but
it bounced back today.
i've been practicing with callback for a while, but i'm
Hi guys,
sorry for the long e-mail, i'm only trying to give as much information
as i think is relevant to my problem (console log, sip.conf and
extension.conf parts).
i've been practicing with callback for a while, but i'm at a dead end.
I hope somebody can help me to move on.
i have
Hi folks,
my goal is to access voicemail (there were some posts about this) but
not by dialing numbers. As asterisk sends voicemails in e-mail, it's
cheaper for us to read e-mails on our cell phone (3g, gprs), and the
message is attached there.
i've looking around in voicemail.conf and
Hey Noah,
Noah Miller wrote:
whatever. That would save you from having to have a dedicated line
for outside access to your extensions. Or, if you have a live person
currently i just play with *, registering multiple voip accounts to be
reachable from multiple sip networks. I'm also
Hi,
Crazy Boy wrote:
But, I want to check my voice mails by dialing our DID number from a
outside telephone.
there must be an easier way, but since i only have asterisk and a couple
of ATAs (spa 3k), i've set one up to give a dial tone to the incoming
caller on the FXO port. This way,
Hi guys,
i've installed asterisk to handle multiple voip accounts. I've looked
at CDR configs, and managed to have cdr-csv files growing after each
call. It would be easier to check my locak asterisk cdr's than logging
into each account and check them at the provider website.
i found that
Hi, and thanks for the suggestions!
Matt wrote:
Sounds like your VoIP provider is incorrectly sending you an Answer before
the call actually completes. I would contact your VoIP provider.
I suppose it could also be possible that YOU have an Answer() statement
that
is only on your VoIP
Yossi Ben Hagai wrote:
The Playback command is auto-answering the call. you can use
Playback(please_wait,noanswer) to fix it.
thanks a lot to everyone who answered, this, of course solved this
issue, it's also in the doc, i just didn't have the idea to look at
playback's manual :(
19 matches
Mail list logo
