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@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
srvlookup=yes
disallow=all
allow=ulaw
allow=alaw
allow=g729
[sip_proxy]
; For incoming calls only.
type=peer
host=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) {
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) {
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");
t_relay_to_udp("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.+@.*") {
# 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]+@.*") {
rewritehost("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@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@philonline.com
]
> Sent: Friday, October 28, 2005 3:33 PM
> To: Ashutosh kumar
> Cc: serusers@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@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
>
>
>
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>
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rrgv