Hello, Mario.
1) This is pretty easy. Assuming the IP address of the Asterisk server is 10.11.12.13
(running on port 5060), you'd need something in your SER config that looks like:
if(method=="INVITE")
{
if(uri=~ "^sip:9[0-9]{7}@.*")
{
rewritehostport("10.11.12.13:5060");
route(1);
break;
};
};
route[1]
{
if(!t_relay())
{
sl_reply_error();
};
}
2) Assuming you have people's mailboxes on Asterisk (I'm not going to go into the
details on how to set up the Asterisk box here, but you'll need to configure the
extensions.conf for the user numbers you're expecting to use, and a voicemail.conf for
each mailbox):
You'd need to read up on failure routes in SER. Basically, you need to set a flag
somewhere in your config that tells SER that you're intending to use a special routing
block in case of a timeout (I also won't go into detail about setting timeouts here as
there are a lot of ways) using t_on_failure.
Something like:
t_on_failure("1");
perhaps somewhere in your invite block.
Then, you'd create a failure route in your config:
failure_route[1]
{
rewritehostport("10.11.12.13:5060");
append_branch();
t_relay_to_udp("your.asterisk.server", "5060");
}
Keep in mind, these are very SIMPLIFIED examples. You might want more in your failure
route. You might want something else. You'll definitely want more in the rest of your
SER config.
All that does is SEND the call to asterisk if someone's call times out (sending to
voicemail). If you then want users to be able to access their voicemail, the EASIEST way
is to set up a voicemail number to call... like...
if(method=="INVITE")
{
if(uri =~ "^sip:\*8500@.*")
{
strip(1);
rewritehostport("10.11.12.13:5060");
route(1);
break;
};
};
Then, if a user dials *8500, he'll be sent to asterisk (but first, it will strip off
the first character -- the *)
On the Asterisk side, you'd need an 8500 extension which looks something like:
exten => 8500,1,VoicemailMain
exten => 8500,n,Hangup
All that does is drop them into the Voicemail main application which will ask for a
mailbox number and password (you'll need to set up the mailboxes individually either
in the asterisk voicemail.conf or using asterisk realtime and setting them up in the
database).
3) This is tricky. Very tricky. There are lots of POSSIBLE methods for doing this, but
none of them are terribly pretty. Check out
http://www.voip-info.org/wiki/view/Asterisk+at+large
4) There are lots of links about setting up SER at
http://www.onsip.org Lots of
information in general about setting up both Asterisk and SER (and both together, I
imagine) can be found doing searches at
http://www.voip-info.org You can also search
google to read the archives for this mailing list by doing:
site:lists.iptel.org search terms
Hope that helps a little.
N.
On Tue, 12 Sep 2006 22:42:59 -0400, Mario François Jauvin wrote
Hi,
I have installed SER and asterisk on a redhat enterprise 4linux box. I would like to find
out how to:
configure SER so that when people dial 9,2226665555 it will redirect the call to
asterisk for processing and forwarding to PSTN I heard mentioned several times how
asterisk can provide voicemail for SER clients and I would like to know how to set this up
so that user can login to their mailbox I would also like to know how users can be
notified of their message indicators. In addition to the info above I would like to know
if there are links that describe this in greater details.
Thanks,
Mario