Hi,
I need some documentation for SEMS, havent found none by now.
I have it running and working but I would like to explore more
possibilities, for instance:
Message playback via SIP client
or/and
Storing the voice message in a folder then sending an email with a
http link to it instead of using attach method
of/and
Compressing the message using MP3/Ogg
and so on ...
Also:
can SEMS be used to provide voice mail messages insted of result
codes for failed calls ?
Something like "the user is busy", "no such user".
I tested addaline.com service, that runs on SER, and they do this.
Any help will be appreciated!
TNX
--
Best regards,
Alessio mailto:alessiof@interconnessioni.it
Hi,
I have just installed SER on my Linux Debian
server. It seems to work just great for users on the
Internet calling each other ( 2 way Audio is
successful ). However if someone calls me ( I am
behind my SER server ) they can hear me but I can't
hear them ( 1 way Audio ONLY ). Based on all the docs
I've read it seems to have something to do with NAT
and RTP ports. My Phone connected to LAN --- SER
Server connected to my LAN on one interface and to DSL
Provider on the other ---- DSL Service Provider. I am
also doing NAT/PAT on SER server. Below is my Topology
and ser.cfg. Appreciate any feedback I get.
My Phone --- SER --- Internet --- Users that register
with my SER server.
SER.CFG File
#
# $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 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)
/* Uncomment these lines to enter debugging mode
debug=7
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/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/nathelper.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
#loadmodule "/usr/lib/ser/modules/auth.so"
#loadmodule "/usr/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)
# ------------------------- 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 (len_gt( 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") {
# 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; use stateful
forwarding; that
# works reliably even if we forward from TCP
to UDP
if (!t_relay()) {
sl_reply_error();
};
}
__________________________________
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Hi Peter!
Not every SIP proxy can deal with clients behind a firewall/NAT box.
Therefore it is useful to us an outbound proxy which is known to be
capable of NATed clients. This proxy will send keep alive messages to
your client to keep the pinhole in your firewall open and rewrites the
IP addresses in the SIP messages of the client behind the FW (or NAT
box).
yes. ser can be an outbound proxy. every proxy can be a outbound proxy.
An outbound proxy is not a special proxy type. Outbound proxy just means
that the SIP UA client always sends the SIP messages to this proxy
instead of contacting the other participant directly.
regards,
klaus
> -----Original Message-----
> From: Peter Hudec [mailto:phudec@postel.sk]
> Sent: Saturday, November 01, 2003 12:21 AM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] Outbound proxy
>
>
> hello,
>
> this is maybe a bit silly question, but.
>
> Outbound proxy is needed, when UA is behind FW.
> What is the function of this proxy (how can this help the UA to make
> calls) and what kind of software can do this.
>
> Can SER be also outbound proxy, or I have to use something else ?
>
> hudecof
>
> --
> mail: [phudec(a)postel.sk] www: [http://www.postel.sk]
> cellular: [+421 02 50203166] icq: [99518783]
> gpg: [http://hudecof.net/data/hudecof.gpg]
>
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
FYI -- there will be a SIP tutorial in the upcoming CCNC conference,
hold by Charles Baker.
-jiri
>I will be presenting a half-day tutorial on SIP at the IEEE Consumer
>Communications and Networking Conference in Las Vegas on Thursday, January
>8. This is a new conference which is positioned to overlap the Consumer
>Electronics Show. See < www.CCNC2004.org >
[...]
> Dr. H. Charles Baker, P.E.,
> Professional Web Site: http://engr.smu.edu/~hbaker
Hi,
I was trying to get click-to-dial working from SERweb and it was failing
for unclear reasons. I then experimented with the "ctd.sh" script,
which does the same thing as the SERweb PHP code. I found that it would
fail during the 'dummy invite' stage, returning an odd error if I used a
URI which used a SIP domain/realm, but would work fine if I used the IP
address of the SIP proxy as the domain. i.e.:
./ctd.sh sip:joebob@mydomain.com sip:billybob@mydomain.com <- fails
./ctd.sh sip:joebob@1.2.3.4 sip:billybob@mydomain.com <- succeeds
The reply from SER would be "500 fifo_uac: no mem for hf block" . The
log file would have these errors:
ERROR: mk_proxy: could not resolve hostname: "mydomain.com"
ERROR: uri2proxy: bad host name in URI <sip:joebob@mydomain.com>
ERROR: uri2sock: Can't create a dst proxy
ERROR: get_hf_block: send_sock failed
ERROR: fifo_uac: no mem for hf block
So it appears that SER didn't understand that the URI's domain was
itself, even though there's a host alias line in the ser.cfg file
listing "mydomain.com" as an alias for this host. It's as if it thinks
it needed to relay the transaction to another SIP router, and look up an
IP to do so. If "mydomain.com" were placed in /etc/hosts as an alias
for the system SER was running on, or if a DNS SRV entry for the SIP
domain were created pointing to the SER system, voila, the
"sip:user@domain" form would work just fine.
Is this a bug ? I think SER isn't checking SER host aliases against the
SIP domain if the SIP request comes in via the fifo t_uac_dlg() command.
I'm using the SER release 0.8.11-r1, aka the "stable CVS" release.
TIA,
Jim
--
+---------------------------------------------------------------------------+
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I have Ser installed and running on debian, I have mysql installed with
the ser module. I can see the table the module created. I am getting
this error message and not sure what it is,
When trying the link..127.0.0.1/html/admin/index.php
*Warning*: Unable to include '/prolog.html' - request execution failed
in */var/www/html/page.php* on line *46*
iptel.org User Management
*Warning*: Unable to include '/separator.html' - request execution
failed in */var/www/html/page.php* on line *48*
*Warning*: Unable to include '/epilog.html' - request execution failed
in */var/www/html/page.php* on line *82
Any ideas?
*
Hello All,
I am trying to register a user from a Messenger Application. I am able to register that user successfully and I am recieving a SIP/2.0 200 OK message. And I am able to see the registered user using the serctl utility.
But I am getting the following warning message:
Warning: 392 12.213.181.254:5060 "Noisy feedback tells: pid=2489 req_src_ip=67.161.15.238 req_src_port=2681 in_uri=sip:rajesh@12.213.181.254 out_uri=sip:rajesh@12.213.181.254 via_cnt==1"
Can some one explain the reason for this warning and how to remove it?
Find the complete send and recieve message trace below:
---- Sending -------
REGISTER sip:rajesh@12.213.181.254 SIP/2.0
Via: SIP/2.0/UDP 67.161.15.238
From: madhu <sip:madhu@67.161.15.238>
To: rajesh <sip:rajesh@12.213.181.254>
Call-ID: 522193516(a)67.161.15.238
Expires:10000
m: madhu <sip:madhu@67.161.15.238>
CSeq: 1 REGISTER
---- Receiving -----
SIP/2.0 200 OK
Via: SIP/2.0/UDP 67.161.15.238
From: madhu <sip:madhu@67.161.15.238>
To: rajesh <sip:rajesh@12.213.181.254>;tag=b27e1a1d33761e85846fc98f5f3a7e58.d4f5
Call-ID: 522193516(a)67.161.15.238
CSeq: 1 REGISTER
Contact: <sip:madhu@67.161.15.238>;q=0.00;expires=10000
Server: Sip EXpress router (0.8.11 (i386/linux))
Content-Length: 0
Warning: 392 12.213.181.254:5060 "Noisy feedback tells: pid=2489 req_src_ip=67.161.15.238 req_src_port=2681 in_uri=sip:rajesh@12.213.181.254 out_uri=sip:rajesh@12.213.181.254 via_cnt==1"
Thanks,
Madhuri.
---------------------------------
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Exclusive Video Premiere - Britney Spears
Hi,
Im currently trying to set SER 0.8.11 up and am having a problem with
the pdt module which translates addresses based on a numeric prefix.
When the accounting module tries to write its record to the database
(mysql), the INSERT query is truncated just after the address that has
been translated by the pdt module. This causes mysql to fail to insert
the record and give an error about query syntax. Other accounting
records for untranslated addresses are inserted properly with no error.
Is this a known issue? I suspect the translated address is being null
terminated by the pdt module but Im not enough of a code hacker to tell
for sure.
-- _--_|\ --------- Todd Wright -- wylie(a)geekasylum.org --------
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Hi...
Thanks a lot for your help
I have a question...
I have tens of telephony lines in a university that
all they work with PSTN... so i wish i could do something with SER to manage the phone calls that they do, and this way save money...is it possible, what i need(Hardware, Software).
Thankyou so much..
IVAN DARIO
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