You might try to just forward without modifying the Req-URI with the config file options:
t_relay_to_udp(ip, port);
OR
t_relay_to_tcp(ip, port
OR
forward(ip,port);
where the tuple ip,port corresponds to the SER instance acting as voicemail/conference server. The last option does not require the tm module and does not involve any transaction, Depending on your performance requirements, it should be better to use one or another...
Hope it helps,
Samuel.
Unclassified.
>>> Teemu Harju <teemu.harju(a)gmail.com> 09/16/05 03:11PM >>>
Hi,
I'm implementing following kind of system. I have two separate machines
where I've installed SER on. The other machine has also SEMS installed. Like
you probably already figured out other is used as a plain proxy/registrar
and the other as a sort of UAS serving voicemail, conference etc with the
help of SEMS.
Now my question is, what is the best way to forward messages from
proxy/registrar to the media server? I've done the forwarding according to
the examples I've found and it uses rewritehostport() to forward messages to
voicemail. Voicemail needs to get the email address from mysql database
located in proxy. This is no problem connecting to the database, but isn't
it so that when I've rewritten the Request-URI that the voicemail cannot
find the email from database beacause it's rewritten and has the media
servers host part in the URI.
How have you guys implemented this? I would also need other SEMS plug-ins
like conference, so I would also need to differentiate those requests also.
Could I rewerite the Request-URI like voicemail@mediaserver? Can the
voicemail still find out who was I actually trying to call to since the URI
is rewritten?
Is there difference in implementing the media server with asterisk? Wouldn't
I need then two different databases? One in asterisk and one in SER.
Thanks...
--
Teemu Harju
http://www.teemuharju.net
Hi
I am guessing the ser does nothing with dtmf coming from the pstn side,
since it's all in the rtp stream, if is there anyway of "seeing" what
ser is doing with it, or what is being sent, I can see the dtmf in
asterisk being passed from ser, but if I use another remote gateway, I
wanted to be able to see if possible, even using ethreal or whatever
Iqbal
confirm 83b7fb1612811c64f85f47a6046dd84dea286d15
dear,
I'm working on a project and i'm using ser as a sip server.
actually i have this problem: every time i send an instant message to a
subscriber i get this error message:
483 too many hops
i tried to fix it but i couldn't and i don't know what to do !
(My computer connect many computers. Is it reason?!)
if any one can help me i'll be very grateful
thank you in advance
i'm using ser 0.9.3 & here's my code:
# ----------- 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
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=yes # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://root:vlftmd@localhost/ser"
alias=sam.cse
# ------------------ 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/textops.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_url","mysql://root:vlftmd@localhost/ser")
#modparam("usrloc", "db_mode", 0)
modparam("registrar","use_domain",1)
# 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","123123123")
#modparam("auth_db","db_url","mysql://root:vlftmd@localhost/ser")
#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 (msg:len >= 2048 ) {
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 (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 (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("sam.cse", "subscriber")) {
www_challenge("sam.cse", "0");
break;
};
save("location");
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();
};
}
_________________________________________________________________
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http://groups.msn.com/?pgmarket=ko-kr
Hi all,
i have a big problem... i need to parser information in Header field From() and eventually add or delete part of info in Header field TO.
For istance: if i(n From field there is '55')
add ('55' in To field).
I don't have familiar to SER (i have started to study it last week (: ) and i likr e to know if it is possible to do it using SER's method without
change the source code ....
thanks for your help
module tm
-----Message d'origine-----
De : Colin Jordan [mailto:cjordan@econetwireless.com]
Envoyé : jeudi 15 septembre 2005 14:55
À : 'Olivier Taylor'
Cc : serusers(a)lists.iptel.org
Objet : RE: [Serusers] Handoff to PSTN
t_relay_to( 192.168.0.2, 5060);
_____
From: Olivier Taylor [mailto:olivier.taylor@gmail.com]
Sent: Thursday, September 15, 2005 2:50 PM
To: 'Colin Jordan'
Cc: serusers(a)lists.iptel.org
Subject: RE : [Serusers] Handoff to PSTN
what's on the line 143???
-----Message d'origine-----
De : Colin Jordan [mailto:cjordan@econetwireless.com]
Envoyé : jeudi 15 septembre 2005 14:46
À : 'Olivier Taylor'
Cc : serusers(a)lists.iptel.org
Objet : RE: [Serusers] Handoff to PSTN
The line 143 points to the script below. What modules will it be looking
for?
_____
From: Olivier Taylor [mailto:olivier.taylor@gmail.com]
Sent: Thursday, September 15, 2005 2:33 PM
To: 'Colin Jordan'
Cc: serusers(a)lists.iptel.org
Subject: RE : [Serusers] Handoff to PSTN
please reply to all
u have a module not loaded, the one involved on line 143
Olivier
-----Message d'origine-----
De : Colin Jordan [mailto:cjordan@econetwireless.com]
Envoyé : jeudi 15 septembre 2005 14:28
À : 'Olivier Taylor'
Objet : RE: [Serusers] Handoff to PSTN
It gives me the following message:
_______________________________________________________
Parse error (143,40-41): unknown command missing loadmodule?
_______________________________________________________
_____
From: Olivier Taylor [mailto:olivier.taylor@gmail.com]
Sent: Thursday, September 15, 2005 1:21 PM
To: 'Colin Jordan'; serusers-request(a)lists.iptel.org
Cc: serusers(a)lists.iptel.org
Subject: RE : [Serusers] Handoff to PSTN
just have a look at /var/log/messages u will find the line number where the
error occurs
Olivier
-----Message d'origine-----
De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la
part de Colin Jordan
Envoyé : jeudi 15 septembre 2005 12:33
À : serusers-request(a)lists.iptel.org
Cc : serusers(a)lists.iptel.org
Objet : [Serusers] Handoff to PSTN
I am not able to restart ser if I put he code below in the ser.cfg to
handoff calls to PSTN.
# attempt handoff to PSTN
if (uri=~^sip:9[0-9]*@mydomain.com) {
log(Forwarding to PSTN\n);
t_relay_to( 192.168.0.2, 5060);
break;
};
Please help!!!!
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prohibited
and maybe
unlawful.
DISCLAIMER: The information in this message is confidential and is legally
privileged. It is intended solely for the addressee. Access to this message
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anyone else is unauthorized.If receiving in error please accept our
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and notify the sender immediately. You must also delete the original
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and maybe
unlawful.
Hi,
I am testing the call forwarding features demonstrated in Issue 5 of the
onsip getting started document. I found that blind call transfer worked
perfectly but fwdbusy & fwdnoanswer gave me errors:
ERROR: t_forward_nonack: non branched for forwarding
ERROR: w_t_relay(failure mode): forwarding failed
ERROR: sl_reply_error used: I'm terribly sorry, server error occurred.
On the onsip site I noted that someone else had this problem and it was
solved by putting append_branch in the fwdbusy and fwdnoanswer sections in
the failure route.
Thankfully that fixed those errors. However when I went to test fwdbusy
again, it doesn't give any errors but still doesn't work. The call scenario
was as follows:
Windows Messenger client 3500 ring BT100 2092. 2092 is off the hook (thereby
sending a 486 busy message) and the call should be forwarded to xlite client
5000.
i.e. 3500 -> 2092(busy) -> 5000
The message sequence showed that everything was correct up to 2092 sending
the 486 busy to SER and then SER sending an ACK back to 2092. But then SER
sends a 407 proxy authentication required to 3500 which replies with an
ACK....and that's it...
Can someone explain why SER would send a 407 Proxy authentication to the
original caller?...I thought this should only be in response to a register
message?....
Any help appreciated,
Thanks,
Aisling.
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Hi All,
Can I have some extensions use gateway 1 and some can use gateway 2?
I want to do this for clients that pay premium they have unlimited calls
for free on a certain onuntry and others will still go to the per minute
basis.
Can I set this up on groups?
Hope anyone can help me. Thank You
Regards,
Nhadie
Hi,
I've read the SER Getting Started form http://www.onsip.org .
When I configure the ser with PSTN Gateway (subject 8 of the guide) I can't register any client (403, Forbidden). I think, the problem is the next line of ser.cfg:
if (!is_uri_host_local()) {
if (is_from_local() || allow_trusted()) {
route(4);
route(1);
} else {
sl_send_reply("403","Forbidden");
};
break;
};
I'd copied the permissions.allow and permissions.deny to /usr/local/etc/ser and in the .allow file I'd uncomment the last line (ALL:ALL)
Any idea? Thanks.
--
Alberto
Hi Jan,
I am following up on the conversation that you and Vikrant had a short while
ago. The OSP interface was implemented as a SER module. It does not
require any changes to the core SER or any of its modules. We have been
hosting the project @ http://developer.berlios.de/projects/osp-module We
would like to move the project to the experimental branch in your CVS. How
should we proceed?
> > In what form does the come ? Did you implement it as a SER module
> > ? If so then we can put it in the experimental branch in CVS. If you
> > made some changes to other modules or the core then it would be great if
> > you could generate patches against the latest development version.
Best Regards,
Dmitry
> > ----- Original Message -----
> > From: "Jan Janak" <jan(a)iptel.org>
> > To: <vmathur(a)transnexus.com>
> > Cc: <serusers(a)lists.iptel.org>
> > Sent: Friday, August 12, 2005 2:35 PM
> > Subject: Re: [Serusers] Distributing "OSP enable SER" binary file w/
> > source
> > code
> >
> >
> >> On 12-08-2005 18:18, vmathur(a)transnexus.com wrote:
> >> > Hi Jan,
> >> >
> >> > The code has been written as a SER module. The details of the code
are
> > available at:
> >> > http://developer.berlios.de/projects/osp-module
> >> >
> >> > Right now the process of compilation requires that you download the
OSP
> > toolkit source code and complie it. This would build the OSP client
> > library
> > that the SER would use. You should then download the SER and the OSP
> > modules
> > and build them. The process is rather lengthy and we were wondering if
we
> > come up with an easier way of doing this.
> >> >
> >> > What exactly is the experimental branch in CVS ?
> >
> >> SER CVS repository contains experimental directory. This is the
> >> directory where anyone can put their contribution. It stays there for
> >> a while and if it proves to be useful and supported by the author
then
> >> we move it to the main source tree, giving the author write access to
> >> the main tree.
> >>
> >> Jan.
> >>
> >>
> >
> > In what form does the come come ? Did you implement it as a SER module
> > ? If so then we can put it in the experimental branch in CVS. If you
> > made some changes to other modules or the core then it would be great if
> > you could generate patches against the latest development version.
> >
> > Jan.
Hello Everyone,
Can anyone tell me what I need to do next in order for SER version 0.9.3
to use the Jabber Module correctly?
This is what I have done so far (on a Red Hat Linux 9 machine):
Installed Jabberd 1.4.4
Installed GNU pth-2.0.4
Installed Expat 1.95.8
Installed Libidn ?
I have Modified the config file for SER to load the Jabber.so module
after the pa.so and tm.so
(I'm sure this is where I am stuck, what to do with parameters in config
file for SER)
This is what I can/cannot do:
I can run the jabber server
I cannot run the SER with the config file which includes the Jabber.so
module load command but I can run it without this command in the config
file.
I am using MySQL for SER
What other files do I need to modify? What information do I need? I'm
not very familiar with linux, I am a beginner...
Any comments/suggestions/websites/help is GREATLY appreciated!
Thanks,
Saleh