[Serusers] Re: Prepaid approach

Tue Nov 8 20:41:31 CET 2005

yes it works in asterisk v1.0.6 and lately 1.2beta2 with a new patch
released last week...

some notes:

B2BUA scenario:

|==========>(PSTN GW´s)
|
Nated_UA1 ===>> SER+rtpproxy <===> Asterisk B2BUA
| |
UA2 <======| |
Prepaid System
with RADIUS

1.- Get the Perl MD5 Package from:
http://www.cpan.org/modules/by-module/MD5/MD5-1.X.tar.gz

Install Perl MD5
tar -zxf MD5-1.X.tar.gz
cd MD5-1.X
perl Makefile.PL
make

2.- Get the B2BUA patch from:
http://download.berlios.de/b2bua/asterisk-b2bua-0.1.2.tgz

Unapack the B2BUA:
tar -zxfv asterisk-b2bua-0.1.2.tgz

3.- Get Asteriks version 1.0.6 with:
cd /usr/src
export CVSROOT=:pserver:anoncvs at cvs.digium.com:/usr/cvsroot
cvs login ---- the password is anoncvs.
cvs co -r v1-0-6 asterisk zaptel libpri

4.- Patch Asterisk
cd /asterisk
cp /your_download_dir/asterisk-b2bua-0.1.2/patch/asterisk-1.0.6-b2bua.patch
patch -p1 < asterisk-1.0.6-b2bua.patch

--------->

Modify app_getchannelstate.c (line 44) in this way ... look

Before ...

static int get_channel_state(struct ast_channel *chan, void *data)
{
struct localuser *u;
LOCAL_USER_ADD(u);
int res = -1;

After ...

static int get_channel_state(struct ast_channel *chan, void *data)
{
int res = -1;
struct localuser *u;
LOCAL_USER_ADD(u);

----------> esto evita un error en la compilacion de asterisk.

6.- Compile and install Asterisk
cd ../zaptel
make clean; make install
cd ../libpri
make clean; make install
cd ../asterisk
make clean; make install

If new to Asterisk, create default configuration files:

make samples

7.- Get the Asterisk PERL AGI module from:
http://asterisk.gnuinter.net/files/asterisk-perl-0.XX.tar.gz

Install with:
tar -zxfv asterisk-perl-0.XX.tar.gz
cd asterisk-perl-0.XX
perl Makefile.PL
make all
make install

8.- Replace the contents of the file /etc/asterisk/extensions.conf with
something like:

[general]
static=yes
writeprotect=no

[default]
; Internationa LD
exten => _011.,1,DeadAGI(/your_download_dir/asterisk-b2bua-0.1.2/agi/agi.pl)
exten => _011.,2,Hangup()
; National LD
exten => _01.,1,DeadAGI(/your_download_dir/asterisk-b2bua-0.1.2/agi/agi.pl)
exten => _01.,2,Hangup()
; Add other routes here

9.- Your /etc/asterisk/sip.conf file should look like:

[general]
context=default
port=5060
bindaddr=0.0.0.0 <http://0.0.0.0>
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729

[sip_proxy]
; For incoming calls only.
type=peer
host=10.10.10.1 <http://10.10.10.1> ; Your SIP proxy IP address
canreinvite=no

Edit the file /your_download_dir/asterisk-b2bua-0.1.2/agi/config.pl
In this file you specify the RADIUS and outgoing call leg parameters.

Run Asterisk
/usr/sbin/asterisk -vvvvg

10.- Below is an example of how you could route between SER and B2BUA.

#************************
# We check credentials for registers
#************************
if (method=="REGISTER") {
if (!www_authorize("your.domain", "subscriber")) {
www_challenge("your.domain", "0");
break;
};
save("location");
break;
};

#************************
# First we check the source of the call
#************************
# If the call comes from the gateway, no authentication is
# required
if (src_ip==10.10.10.2 <http://10.10.10.2>) {
log(1,"Call from pstn. \n");
# If the call comes from B2BUA, no authentication is
# required. The first leg of the call has already been
# authenticated.
} else if (src_ip==10.10.10.4 <http://10.10.10.4>) {
log(1,"Call from B2BUA. \n");
} else {
# We check user credentials
if (method=="INVITE") {
if (!proxy_authorize("your.domain", "subscriber")) {
proxy_challenge("your.domain", "0");
break;
};

};
# Not all the users are prepaid, so we check the database
# to see if the call will be routed through B2BUA.
# If every call is to be routed through B2BUA the "is_user_in"
# conditional is not required.
if (is_user_in("From", "prep")) {
rewritehost("10.10.10.4 <http://10.10.10.4>");
t_relay_to_udp("10.10.10.4 <http://10.10.10.4>", "5060");
break;
};

};

#************************
# Then we check the destination of the call
#************************
# We use a specific pattern to identify our SIP users.
# This can be replaced with a database lookup if a pattern
# is not possible.
if (uri=~"^sip:666.+ at .*") {
# Look user in the location database
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
} else {
# Try to send call to dest.
if (!t_relay()) {
sl_reply_error();
};
};
# Forward numeric uri's to PSTN gateways
} else if (uri=~"^sip:[0-9]+ at .*") {
rewritehost("10.10.10.2 <http://10.10.10.2>");
t_relay();
# Anything else is forbidden
} else {
sl_send_reply("403", "Call cannot be served here");
break;
};

################################################################

On 10/28/05, Daniel Liu <daniel.liu at cu88.com> wrote:
>
> I think it is better to use a B2BUA for prepaid.
>
> Asterisk can do this job. But I don't know the stability.
>
> regards,
> Daniel
>
> Ashutosh kumar write:
>
> > Hi,
> > OK, but how do you monitor the call while it is in progress?
> >
> > To disconnect the call, I am planning to use the session timers set in
> the
> > header prior to intitiating the call, which will offload the task of
> > montoring the call by SER, and disconnection-on-zero-credit will be
> handled
> > by our pstn gw. Am I wrong somewhere?
> >
> >
> > Regards,
> > Ashutosh
> >
> > -----Original Message-----
> > From: Ryan Pagquil [mailto:rpagquil at philonline.com]
> > Sent: Friday, October 28, 2005 3:33 PM
> > To: Ashutosh kumar
> > Cc: serusers at lists.iptel.org
> > Subject: Re: [Serusers] Prepaid approach
> >
> > Hi,
> > We are a bit similar in implementing prepaid service. Our users are
> > normally can call other users in our domain, and they are initially not
> > member of the pstn group in the ser.grp table. once they bought credits
> > they will be put in the "pstn" group and can call pstn destinations. But
> > when they run out of credit, we sends a bye message using sipsak on both
> > PSTN gateway and the user's IP phone.
> >
> > Regards,
> > Ryan
> >
> > Ashutosh kumar wrote:
> >
> >> Hi,
> >>
> >> While trying to implement prepaid solution using SER, I decided to go
> >> be a intuitive approach which is as follows.
> >>
> >> -When the user registers, he is placed in a "voip?groups table of ser
> >> (or radius server database), i.e he can make only pc-to-pc calls.
> >>
> >> - Later , or otherwise, when the user registers, he is shifted to a
> >> "pstn?groups table of ser , i.e noew he can make only pc-to-pstn calls.
> >>
> >> -Henceforth, whenever the user logins and tries to make pstn call, the
> >> call is approved only if group_radius_is_user_in(username) succeds,
> >> else the call rejected by SER.
> >>
> >> -To restrict the user from making pstn calls when his credits are zero
> >> (or beyond a threshold) , a dedicated cron job is scripted to move
> >> users from "pstn?group to "voip?when their
> >>
> >> account_cerdit=0.
> >>
> >> Am I right in using this approach, or are there any foreseeable
> >> problems which I might be overlooking.
> >>
> >> Thanks.
> >>
> >> Regards,
> >>
> >> Ashutosh Kumar
> >>
> >>------------------------------------------------------------------------
> >>
> >>_______________________________________________
> >>Serusers mailing list
> >>serusers at lists.iptel.org
> >>http://lists.iptel.org/mailman/listinfo/serusers
> >>
> >>
> >
> >
> > --
> > Ryan Pagquil
> > Infodyne Inc. - PhilOnline.com
> > 3603 Antel Global Corporate Center
> > Doña Julia Vargas Ave.
> > Ortigas Center Pasig City
> > Tel: 687-0715
> > Web: www.philonline.com <http://www.philonline.com>
> >
> >
> >
> > _______________________________________________
> > Serusers mailing list
> > serusers at lists.iptel.org
> > http://lists.iptel.org/mailman/listinfo/serusers
>
>
> _______________________________________________
> Serusers mailing list
> serusers at lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
>

--

rrgv
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