Hi all,
i configured ser.cfg to support mysql db. I wanted to
know in wchich table i should put the PSTN Gateways
and dailpeers. Because i have more than 5 PSTN
Gateways i wanted to use it in ser with some dialpeer.
Is it possible to take the gateway ip and port from
mysql table according to dialpeer route?
Thank You
Yours,
Abdul Lateef
Computer Programmer
HATIF COM
Mob: +974 - 5405022
Tel: +974 - 4883068
ICQ: 276994704
YM!: abdul_zu
Fax: +974 - 4883063
Doha Qatar
http://www.hatif.com
__________________________________
Do you Yahoo!?
Read only the mail you want - Yahoo! Mail SpamGuard.
http://promotions.yahoo.com/new_mail
Hello, i am runing ser 0.10.99-dev8 and 0.8.14 on a
different machine.
When calling from a sip device (Linksys RT31P2) to
PSTN (Cisco 5300), the ACK message is not logged to
mysql. Only INVITEs (without 200 status) and BYEs are
logged.
The other scenarios sip to sip and pstn to sip, log
all the messages i need. (INVITE 200, ACK and BYE)
I am sure there is something missing in ser.cfg.
Any clue on what i am doing wrong??
Kind regards, Pablo.
# ------------------ module loading ----------------------------------
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/textops.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 2)
# -- auth params --
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
# -- rr params --
modparam("rr", "enable_full_lr", 1)
# -- acc params --
modparam("acc", "db_flag", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "report_ack", 1)
# ------------------------- 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
if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (method=="INVITE" || method=="BYE" || method=="ACK") {
setflag(1);
};
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
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 (src_ip==66.100.100.70 and method=="INVITE") { ## Calls from PSTN
forward(uri:host, uri:port); ##
break; ##
};
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("66.100.100.73", "subscriber")) {
www_challenge("66.100.100.73", "0");
break;
};
save("location");
break;
};
if (method=="INVITE" || method=="BYE" || method=="CANCEL" || method=="ACK") {
setflag(1);
acc_db_request("", "acc");
};
if (uri=~"^sip:1786[0-9]*@66.100.100.73") {
prefix("5000#");
rewritehostport("66.100.100.70:5060");
forward(uri:host, uri:port);
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
On Fedora core 3 with all needed packages installed and config.php
configured properly I can't login to serweb user_intrface or admin
When I first load the user or admin index.php I get many errors in
apache error logs like the following:
[client 192.168.25.2] PHP Notice: Undefined index: extrahtml in /var/www/html/phplib/oohforms.inc on line 72, referer: http://192.168.25.3/iptel/user/
[client 192.168.25.2] PHP Notice: Undefined property: isfile in /var/www/html/phplib/oohforms.inc on line 250, referer: http://192.168.25.3/iptel/user/
[client 192.168.25.2] PHP Notice: Undefined property: hidden in /var/www/html/phplib/oohforms.inc on line 253, referer: http://192.168.25.3/iptel/user/
.....snip
[client 192.168.25.2] PHP Notice: Undefined index: frozen in /var/www/html/phplib/oohforms.inc on line 284, referer: http://192.168.25.3/iptel/user/
When I enter username/passwd and submit I get more errors in logs
(Notice, index frozen etc.) and also A Bad Username or Password message
and I return to login screen. There is nothing wrong with passwords, and
I can connect to ser db with mysql client just fine.
I use serweb_2004-07-27 and ser-0.8.14.
I have the vars in php.ini according to the instractions.
Please point me to any possible solution.
Thanks.
Just some thoughts based on my experience...
After months trying to make everything work using rtpproxy-mediaproxy with
almost everything accomplished but video, I tried to switch to stun solution.
All my problems are gone now, I have audio, video, presence and instant
messages working like a charm. And most important media server doesn't flow
thru my server so network load remains very low. I have been testing for some
days now and I'm quite happy since I still have to stumble on major problems.
Now some considerations... On a poll onsip.org STUN usage is very low and
rtpproxy-mediaproxy rule as NAT trasversal solution. Why don't people use
stun? Has it some major drawbacks I still haven't found? What are main
advantages of rtpproxy-mediaproxy solutions?
I'm really curious to know serusers opinions about this issue.
thank you all for your two cents ;)
--
Giovanni Balasso
giaso(a)yahoo.it
FYI - below link worked pretty well for me to install serweb properly.
http://www.aarnet.edu.au/events/conferences/2004/sip/Servers/ser-install-RH…
Thanks,
Karun
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]On Behalf Of Leo Papadopoulos
Sent: Wednesday, June 29, 2005 1:56 PM
To: serusers(a)lists.iptel.org
Subject: [Serusers] Docs to install SERWEB please..
Dear SER Users,
What is the best document to use to help me install SERWEB? Where might I find these documents? I already have SER running with MySQL.
Leo Papadopoulos
E-mail: leo(a)telecomCTO.com
Web site: www.telecomCTO.com <http://www.telecomcto.com/>
And even worst.
There are some kind of NAT that STUN does not work.
You can check the mailing list i think some people call it "crap nat".
Regards,
Ricardo Martinez.-
> -----Mensaje original-----
> De: Andres [mailto:andres@telesip.net]
> Enviado el: Martes, 05 de Julio de 2005 17:17
> Para: Giovanni Balasso
> CC: serusers(a)lists.iptel.org
> Asunto: Re: [Serusers] NAT considerations...
>
>
> Giovanni Balasso wrote:
>
> >Just some thoughts based on my experience...
> >After months trying to make everything work using
> rtpproxy-mediaproxy with
> >almost everything accomplished but video, I tried to switch
> to stun solution.
> >All my problems are gone now, I have audio, video, presence
> and instant
> >messages working like a charm. And most important media
> server doesn't flow
> >thru my server so network load remains very low. I have been
> testing for some
> >days now and I'm quite happy since I still have to stumble
> on major problems.
> >Now some considerations... On a poll onsip.org STUN usage is
> very low and
> >rtpproxy-mediaproxy rule as NAT trasversal solution. Why
> don't people use
> >stun? Has it some major drawbacks I still haven't found?
> What are main
> >advantages of rtpproxy-mediaproxy solutions?
> >I'm really curious to know serusers opinions about this issue.
> >
> >thank you all for your two cents ;)
> >
> >
> >
> STUN does not work if your NAT is Symmetric. For example all
> Linux NATs
> or routers with Linux OS like the Linksys ones. Unless you have full
> control on what type of NAT your customer will deploy, it
> will be very
> hard to stick to an all STUN solution.
>
> --
>
> Andres
> Network Admin
> http://www.telesip.net
>
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
I'm using Ser 0.9.3 with Microsoft Messenger 5.1. All the contacts that are logged in appear as disconnected, nobody sees the others are online. Has someone the same problem?
Thanks
____________________________________________________________
Navighi a 4 MEGA e i primi 3 mesi sono GRATIS.
Scegli Libero Adsl Flat senza limiti su http://www.libero.it
Hi
I am mapping external numbers to internal ones in a db, eg 08450040600
maps to 040600
So when a call comes from pstn to 08450040600 it hits the ip phone on
040600, all well and good.
Now what I am testing is deploying asterisk for other features like call
pickup and all. I have managed to create separate contexts in asterisk
based on company ID (i.e virtual PBX) assigned from a DB.
And when a call from IP phone A (company id =cid=500) calls out the call
goes via ser (thats where all IP phones are registered) to asterisk, and
then out to ser --->pstn (was gonna go out via asterisk - pstn, but
decided to keep it all within ser). This is fine, because asterisk is in
the loop.
Now what I need is for inbound calls to that IP phone to also be routed
into asterisk, and then to ser and to the ipphone, that way I pull all
calls for each company into a asterisk context for all its pbx
functionality.
The problem is that when 0845 is called from pstn, I cant setflag on it,
using is_user_in, since From = pstn dialing number, and To =
4408450040600, and not 040600, any ideas on what I can match on so that
I can set the correct flag and divert to asterisk.
Iqbal