Hello all,
Im using ser, with esteras softphone, and a couple of cisco ata186's. I
have to say I really like ser, as it is small fast, and reliable without
being over complicated.
I would like to know what other ser users and developers use for pstn
gateways? I have a openline4 card from voictronix, but there is no
software support for sip (When I bought it I thought I would use h.323).
I have looked at cisco's pstn gateway modules, but was turned off them
because of price, and lack of clear straight forward information to name
but two reasons.
Ideally what I would like to have is a solution that would support at
least 4 pstn lines (expandable to maybe 8), that would run on FreeBSD
(or at worst linux), works well with sip & ser, with high quality audio,
and minimal delays when dialing in out.
TIA,
-Emil
Hi All,
I need to implement an used based dialplans module on SER, so that depending
on the user information stored in the database, a certain prefix should be
inserted in the called number. Has anyone had a similar issue?
Thanks
Fernando
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Folks,
I am currently playing with t_on_negative feature trying to implement
overflow routing (i.e. if original destination returns an error, then
request should be adjusted somehow and redirected to some, possibly
different, destination). I've started with the following config:
route {
[...]
rewriteFromRoute();
if (method == "INVITE") {
addRecordRoute();
};
if (method == "INVITE) {
t_on_negative("1");
};
t_relay_to("address1", "port1");
}
reply_route[1] {
rewritehostport("address2:port2");
append_branch();
}
But quickly found that after transaction was redirected to
address2:port2, ACKs, BYEs, 200K and CANCELS are still being forwarded
to address1:port1, despite containing valid Route fields pointing to
address2:port2. Then I've modified it as follows to let ser use
information from that field to route ACKs, 200OKs and BYEs:
route {
[...]
rewriteFromRoute();
if (method == "INVITE") {
addRecordRoute();
};
if (method == "INVITE" || method == "CANCEL") {
# INVITEs and CANCELs
if (method == "INVITE) {
t_on_negative("1");
};
t_relay_to("address1", "port1");
} else {
# ACKs, 200OKs, BYEs
t_relay();
};
}
reply_route[1] {
rewritehostport("address2:port2");
append_branch();
}
For the most of the time everything works like a charm - if
address1:port1 is unreachable or replies with an error the request is
being redirected to the second destination, BUT if the initiating UA
tries to cancel transaction when transaction is already redirected but
before receiving final "200 OK" from the second destination, the
CANCEL request is forwarded to address1:port1, not to address2:port2
as it should be. I've tried to modify setup as follows, thinking that
maybe in the case of CANCEL explicit specification of proxy address
confuses ser, but no avail - in this case ser forwards the CANCEL
request to its own address and eventually it dies with Too Many Hops.
route {
[...]
rewriteFromRoute();
if (method == "INVITE") {
addRecordRoute();
};
if (method == "INVITE") {
# INVITEs
t_on_negative("1");
t_relay_to("address1", "port1");
} else {
# CANCELs, ACKs, 200OKs, BYEs
t_relay();
};
}
reply_route[1] {
rewritehostport("address2:port2");
append_branch();
}
I think that such behaviour arises from the fact that ser after
branching a transaction doesn't keep an address this transaction was
forwarded to. In my opinion, it needs to be corrected, so that after
receiving CANCEL from the UA that initiated transaction ser probably
should CANCEL *all* branches of this transaction. To do this it needs
to be able to tell exactly where to send CANCELs.
What do you think?
-Maxim
Friends,
please, can you help me and share with me privately which SIP2PSTN
gateways you are using in your networks? I'ld like to get a better
feeling of what devices today have reputation of being usable and
with which of them SER has been tested. I'm interested in all-grade
devices, from single FXO to SS7 gateways. I'll keep it for myself.
Thank you a lot,
-Jiri
--
Jiri Kuthan http://iptel.org/~jiri/
Hello,
see dbtext for that. But notice that the module has not been thoroughly
tested. It is not included in v0.8.10 so you have to use a CVS snapshot
which is quite unstable these days due to lots of changes. If you want
to use it before the new release of ser is out you might get weird
situations, but you can provide us some help with testing ... It is your
choice ...
-.-
Best regards,
Daniel
David Beckemeyer wrote:
>I'm trying to get SER up and I need auth. I'm not a Mysql user
>and, frankly, I'd rather not need to learn it, just to get
>authentication support. I wish there were a middle-ground
>between no-auth at all and a full-blown Mysql installation.
>
>I tried the Cookbook approach, using Dan Austin's HOWTO for
>setting up Mysql for SER, but the commands do not work as shown
>so I'm left with learning Mysql through and through to debug
>why those cookbook commands don't work.
>
>I'm very familiar with SIP and I want to test SER, not a new Sql
>server DB engine. It seems like Mysql is a distraction to that
>effort.
>
>Is anyone working on something like a flat-file module to
>replace Mysql? Any pointers for the best place to start on
>developing such a module (i.e. shortcuts)?
>
>Thanks,
>
> - david
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
>
>
>
Hi Michael,
This is exactly what I would like to achieve.
I understand the meaning of your pseudo code below, but alas I have not enough
knowledge and skills to transform your pseudo code below into a working
script.
Could you please provide a real example ?
Lasse
Michael_Graff(a)isc.org wrote:
> Nils Ohlmeier <nils(a)ohlmeier.de> writes:
> > A proxy can challenge Invites and Byes, but should not do this with
> > external Invites to your local user. Otherwise your your user
> > wouldn't be reachable from outside.
>
> I implemented something much like this:
>
> if (to me):
> if register
>
> www_authorize or fail if not a valid register
>
> done
>
> if claiming to be "From" one of the domains I accept
> registrations for
>
> proxy_authorize
>
> done
>
> if not to me (I'm relaying for a local phone to an external address)
>
> proxy_authorize (once again, based on from address)
>
> done
>
> --Michael
On ATA side, you just need to enable the 'Receiver-tagged VIA header'
feature by setting bitt 22 of ConnectMode to 1. On SER side, nothing you
need to configure specifically.
----- Original Message -----
From: "Michael Vasilenko" <acid(a)dg.net.ua>
To: "Bo" <boman122(a)yahoo.com>
Sent: Friday, March 28, 2003 9:52 AM
Subject: Re: [Serusers] ATA 186 Behind NAT
> Bo (boman122(a)yahoo.com) wrote:
> > Yes. I already tested that.
>
> Can you please explain, what I must configure in ATA and SER
> to make a call from ATA with private IP
> >
> > ----- Original Message -----
> > From: "Ricardo Villa" <ricvil(a)epm.net.co>
> > To: <serusers(a)lists.iptel.org>
> > Sent: Friday, March 21, 2003 10:24 AM
> > Subject: [Serusers] ATA 186 Behind NAT
> >
> >
> > > Hi,
> > >
> > > I was reading about the ATA186's ability to detect that it is behind a
> > NAT.
> > > It requires that:
> > >
> > > "For the Cisco ATA to automatically detect its presence behind a NAT,
the
> > > SIP proxy server
> > > or remote user agent server must include the "received=" parameter in
the
> > > Via header in the
> > > responses to the Cisco ATA if the proxy detects that the source
address
> > and
> > > port do not match
> > > those in the Via header."
> > >
> > > Does SER support it?
> > >
> > >
> > > Thanks,
> > > Ricardo
> > >
> > > _______________________________________________
> > > 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
>
> --
> Michael Vasilenko
Hi
Where is the voicemail for ser at? Is it presently available? I glanced
through the cvs and couldn't find any.
Asterisk has decent voicemail.. however its not yet ported to anything else
except Linux
check out http://www.asteriskpbx.org
Sean
> -----Original Message-----
> From: Jiri Kuthan [mailto:jiri@iptel.org]
> Sent: Monday, March 31, 2003 7:59 AM
> To: ahmed.abdoulla(a)tedata.net; serusers(a)lists.iptel.org
> Subject: Re: [Serusers] Voicemail Feature
>
>
> At 06:11 PM 3/30/2003, Ahmed Abdoulla wrote:
>
> >Dear Madam/Sir,
> >
> >
> >
> >We are a data communication provider in Egypt, and we are
> very much interested in deploying SER.
> >
> >We are particularly interested in having a voicemail feature
> as one of our offerings, and we have tried the CVS (8.11)
> version but we have noticed (naturally) that it's not stable yet.
> >
> >
> >
> >Could you please give us a rough estimate on when you
> believe a beta or alpha release will be available?
>
> I'm little a bit reluctant to do so -- it has always taken
> longer than we anticipated
> and I would not like to set unrealistic expactations. If you
> insist -- without any
> guarantee, a *testing* version may be ready to go in two weeks.
>
> -Jiri
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
Hi,
Try to use mysql and auth. But, got the msg "ERROR:
bad config file (1 eorros)", when I type
"/usr/local/ser". Is anything wrong in my ser.cfg?
Thanks,
Alan
[root@localhost ser]# more ser.cfg
#
# $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters
------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=no
log_stderror=no # (cmd line: -E)
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"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.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", "secret", "alsdkhglaksdhfkloiwr")
modparam("auth", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which
true in this config),
# uncomment also the following parameter)
#
modparam("auth", "password_column", "password")
# ------------------------- request routing logic
-------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwars==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (len_gt( max_len )) {
sl_send_reply("513", "Message too
big");
break;
};
# Do strict routing if pre-loaded route
headers present
rewriteFromRoute();
# 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") {
# Uncomment this if you want to use digest
authentication
if
(!www_authorize("iptel.org", "subscriber")) {
www_challenge("iptel.org", "0");
break;
};
save("location");
break;
};
# native SIP destinations are handled
using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not
Found");
break;
};
};
# forward to current uri now
if (!t_relay()) {
sl_reply_error();
};
}
=====
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