This is the right idea.. the bigger challenge is disconnecting the call in progress after their credit has expired. - Daryl On 10/27/05, Ashutosh kumar <ashutoshk(a)chetu.com> wrote:

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(a)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: -- 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

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:

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 <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.+@.*") { # 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 <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 &lt;daniel.liu(a)cu88.com&gt; wrote: -- rrgv