27 Oct
2005
27 Oct
'05
10:08 p.m.
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
27 Oct
27 Oct
10:36 p.m.
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,
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(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
28 Oct
28 Oct
1:02 p.m.
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(a)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
2:37 p.m.
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:
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(a)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
7:44 p.m.
Ashutosh,
You may want to look at the thread "Detecting runaway calls" we have
been discussing these type issues there.
Basically if you control the PSTN gateway and/or there is support for
session timers then you are in good shape, and this should work.
- Daryl
On 10/28/05, 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:
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(a)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
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
8 Nov
8 Nov
11:21 a.m.
New subject: [Serusers] mediaproxy
Hi all
Is it possible to run mediaproxy on a server with two nics (in different
network segments of course) and let it route traffic from one nic to the
other (doing a real proxying)?
Thanks for your help
Frank
9 Nov
9 Nov
3:59 p.m.
New subject: [Serusers] mediaproxy
Not that I am aware of. rtpproxy has support for options in
force_rtp_proxy() used for implementing a bridge.
g-)
----- Original Message -----
From: "Frank Fischer" <frank.fischer(a)digitalnomads.ch>
To: <serusers(a)lists.iptel.org>
Sent: Tuesday, November 08, 2005 10:21 AM
Subject: [Serusers] mediaproxy
Hi all
Is it possible to run mediaproxy on a server with two nics (in different
network segments of course) and let it route traffic from one nic to the
other (doing a real proxying)?
Thanks for your help
Frank
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
29 Oct
29 Oct
5:16 a.m.
New subject: [Serusers] Re: Prepaid approach
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(a)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 搗oip?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
損stn?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 損stn?group to 搗oip?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(a)iptel.org
http://mail.iptel.org/mailman/listinfo/serusers
--
Ryan Pagquil
Infodyne Inc. - PhilOnline.com
3603 Antel Global Corporate Center
Do馻 Julia Vargas Ave.
Ortigas Center Pasig City
Tel: 687-0715
Web: www.philonline.com
_______________________________________________
Serusers mailing list
Serusers(a)iptel.org
http://mail.iptel.org/mailman/listinfo/serusers 
8 Nov
8 Nov
9:41 p.m.
New subject: [Serusers] Re: Prepaid approach
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 <daniel.liu(a)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:
--
rrgv
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:
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
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(a)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
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http://lists.iptel.org/mailman/listinfo/serusers
6959
days inactive
6972
days old
8 comments
7 participants
participants (7)
-
Ashutosh kumar -
Daniel Liu -
Daryl Sanders -
Frank Fischer -
Greger V. Teigre -
Rafael R. GV -
Ryan Pagquil