I have set up the ser to report accounting information. but I only get
the INVITE request, no BYE. I have set the log_flag=1, setfalg(1) and
record_route() in the ser file. I have no idea what is the problem.
Attached is the ser.cfg file
Dear Iqbal:
Thank you very much. I just want to plan a prepaid function in my scenario.
And wanna test a tool (asterisk / b2bua of vovida).
So I need another sound about what is the further.
Anyway, thank you for your valuable experience.
Charles
On Wed, 23 Feb 2005 00:19:25 GMT, Iqbal <iqbal(a)gigo.co.uk> wrote:
>
> um...dont understand, SER doesnt/shouldnt manage all.
>
> Also ammendmen to post below, asterisk_b2bua below is not asterisk and
> from teh cursory glance i took it looks like as if it needs radius..
>
> Iqbal
>
> On 2/22/2005, "Walter Willis" <walterwn(a)gmail.com> wrote:
>
> >b2bua work with "SER"???
> >
> >but the ser manager all.
> >
> >
> >
> >
> >On Tue, 22 Feb 2005 17:52:01 GMT, Iqbal <iqbal(a)gigo.co.uk> wrote:
> >>
> >> could ask this on asterisk :-) but answers inline
> >>
> >> On 2/22/2005, "Charles Wang" <lazy.charles(a)gmail.com> wrote:
> >>
> >> >Dear ALL:
> >> >
> >> >I find a program named "asterisk_b2bua" on
> >> >http://developer.berlios.de/projects/b2bua/
> >> >
> >> >And I also download them(two components) and try to test it.
> >> >
> >> >But I have not enough knowledge about asterisk. It seems a Software PBX.
> >> >Does asterisk_b2bua work? Does anybody ever try it?
> >>
> >> worked with asterisk, but this seems like a stripped version so I havent
> >> used this, and it seems in pre alpha stage so not sure about it
> >> >
> >> >I have questions about my scenario.
> >> >
> >> > |======================> UA2 (Internet)
> >> > |
> >> >UA1 ===> SER ===> Asterisk B2BUA ===> Trunking A (PSTN)
> >> > |
> >> > |
> >> > CDR + Prepaid + Handle Calls(Tear-down when call
> >> >during limited)
> >> > |
> >> > |
> >> > Authentication ( Radius / DB )
> >> >
> >> >Q1. Will Asterisk's B2BUA pass through the "To" (such as
> >> > 0939749xxx(a)ser.xxx.net.tw PSTN number) to Trunking A?
> >> > In another word, is the B2BUA not necessary to rewrite the phone number?
> >> >
> >> yup, you use extensions in asterisk, but use default catchall ._ to take
> >> all your calls and then route them out to ur pstn provider
> >>
> >> >Q2. How can B2BUA know when to tear-down this call if the call has some limits.
> >> > For example, UA1 has only 120 seconds to use this International call.
> >> >
> >> Because the rtp should go via the b2bua, and it is taking care of the
> >> billing so it knows how long the call has been in progress.
> >>
> >> >Q3. Is it necessary for Radius Authentication? Is it possible if no
> >> >radius exists
> >> > and SER and Asterisk use the same MySQL database.
> >>
> >> You can make them all exists, I dropped radius early one, it didnt seem
> >> to add much benefit to what I was doing, mysql seemed to do it all, no
> >> point adding radius although some billing software does like to use
> >> radius records, so it all depends on how ur billing.
> >>
> >> Asterisk by default uses its own DB, but this can be changed, but I am
> >> thinking that it might be better to keep them separate, because then ser
> >> DB can be moved off onto a diff machine, and u can use asterisk to
> >> maintain billing and keep that separate...ur accounst dept will want
> >> that guarded :-)
> >>
> >>
> >> >
> >> >Q4. What do I need to download and setup when I use Asterisk
> >> > to start this prepaid compoment? The first one shall be
> >> >asterisk-1.0.5.tar.gz.
> >> > Maybe some Mysql supported scripts or installaion. And any more??
> >>
> >> Just asterisk, not sure what the latest release is, am installing new
> >> version this week, so if I get round to it, I'll give feedback
> >>
> >> Iqbal
> >> >
> >> >Best Regard
> >> >Charles
> >> >
> >> >_______________________________________________
> >> >Serusers mailing list
> >> >serusers(a)lists.iptel.org
> >> >http://lists.iptel.org/mailman/listinfo/serusers
> >> >
> >> >
> >>
> >> _______________________________________________
> >> Serusers mailing list
> >> serusers(a)lists.iptel.org
> >> http://lists.iptel.org/mailman/listinfo/serusers
> >>
> >
> >
>
Assuming you were using a SER box to pass calls to a PSTN, is there
anyway to create some rules based on the cause codes it sends back?
I want to be able to redirect a call if one line says it's full or has
an error to a second route or PSTN.
Best regards
Hello,
I have installed ser 0.9 from cvs and create a new db with ser_mysql.sh but
when I start ser I have this error:
Feb 23 00:45:20 spada04 /usr/local/sbin/ser[16112]: register_udomain():
Invalid table version (use ser_mysql.sh reinstall)
Feb 23 00:45:20 spada04 /usr/local/sbin/ser[16112]: domain_fixup(): Error
while registering domain
Feb 23 00:45:20 spada04 /usr/local/sbin/ser[16112]: ERROR: fix_expr :
fix_actions error
and if I try to run ser_mysql.sh reinstall I have this:
Feb 23 00:46:55 spada04 /usr/local/sbin/ser[16151]: table_version(): Invalid
number of rows received: 2, grp
Feb 23 00:46:55 spada04 /usr/local/sbin/ser[16151]: group:mod_init(): Error
while querying table version
Feb 23 00:46:55 spada04 /usr/local/sbin/ser[16151]: init_mod(): Error while
initializing module group
Laurent
Hi all!
Is there a way to rewrite a SIP packet over several headers?
I want put some data from a certain header and put it into antoher
header - e.g I want to (don't ask why) put the call-id into the userpart
of the Contact: URI
Is this possible within the ser.cfg (e.g. using avp)?
regards,
klaus
Hi there,
Does anyone know a code snippet to activate sems in failure_route ?
I want to redirect to voicemail in case of busy line or timeout.
I can't make a t_newtran() in the failure_route, so if i put the
vm("/tmp/am_fifo","voicemail") function in this block, the sems answers
with his default message but the ruri (and therefore the mailaddress of
the recipient) is unknown and it is not possible to terminate the call
with a BYE-request!
Are there in addition any side-effects by using the mediaproxy?
regards,
Philipp
Dear Sir/madam,
I just finished installed and configured ser, now the sip clients like
windows messenger and the sip phone can sign on, but the problem is
after the sip clients signed on, I can not see those clients are in
on line status in the windows messenger, how can i fix it? thank you
regards
Jeffrey
Hi,
I load the module NATHELPER :
modparam("nathelper", "natping_interval", 10)
modparam("registrar", "nat_flag", 6)
When I write:
if (nat_uac_test("3")) {
force_rport();
fix_nated_contact();
};
I run ser and I have : parse error (121,22-23): unknown command, missing
loadmodule?
Hi,
In your onreply_route, just add an append_branch()
And it will work.
Br
hw
Rumor has it yes, but I have not been successful. I posted a question
last week, and was told to search the
archives. I did, but
not much came up!
http://www.google.ca/search?hl=en <http://www.google.ca/search?hl=en>
&q=302+site%3Amail.iptel.org&btnG=Google+Search&meta=
I think I'm on the right track, but don't have time today to play with
it. Here is what I got so far:
#
# $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
#debug=3 # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
debug=3
fork=no
log_stderror=yes
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
#children=4
fifo="/tmp/ser_fifo"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
#loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/xlog.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
#loadmodule "/usr/local/lib/ser/modules/auth.so"
#loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
#modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
#modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this
config),
# uncomment also the following parameter)
#
#modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
modparam("tm", "fr_inv_timer", 15 )
modparam("tm", "fr_timer", 10 )
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
break;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
save("location");
break;
};
};
if (method=="INVITE") {
rewritehostport("216.143.130.70:5060");
t_on_reply("1");
t_on_failure("1");
t_relay();
break;
}
}
onreply_route[1] {
xlog("L_NOTICE", "Reply status %rs: Reason %rr\n");
if (t_check_status("302")) {
xlog("L_NOTICE", "We got a 302!!!!!\n");
## ADD CODE TO SEND OUT INVITE!! ###
};
}
failure_route[1] {
xlog("L_NOTICE", "Failure status %rs: Reason %rr\n");
}
Let me know if you find any information.
Paul
-----Original Message-----
From: serusers-bounces <at> iptel.org [mailto:serusers-bounces <at>
iptel.org] On Behalf Of Martin Koenig
Sent: Tuesday, February 22, 2005 11:00 AM
To: serusers <at> iptel.org
Subject: [Serusers] Turn 3xx into new Invite
Hello,
is it possible to take 3xx-responses in a reply_route and turn them into
a new INVITE on Ser using append_branch or simliar?
Regards,
Martin
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