For example, from serweb user_interface I can add new contacts with
their sip addresses
But when I open my SIP client (I'm using Nortel SIP Client and Windows
Messenger, not phones) I can't see my contacts.
I have been watching the network whit tcpdump, and I can see the
following messages:
REGISTER from the client
200 OK from the server
You are right, Jiri (and when not?). In 200 OK message, SIP Message
Header, I can see several Contact Fields with my contacts.
So, it should be that Windows Messenger can't understand this. I'll try
another clients.
Thanks very much.
Curro
----- Mensaje Original -----
De: Jiri Kuthan <jiri(a)iptel.org>
Fecha: Miercoles, Enero 14, 2004 4:27 pm
Asunto: Re: [Serusers] SIP client and serweb
> At 04:23 PM 1/14/2004, CURRO_DOMINGUEZ wrote:
> >Hello everybody
> >
> >Well, we have been able to make a Voice call between two Nortel
> SIP
> >clients using SER. It's great! Thanks for your help.
> >
> >I have a question about user's contacts. In one hand, I can use
> serweb
> >to login with my username and add sip contacts or add phonebook
> entries.>In other hand, I can use my client to add new contacts.
> Moreover, the
> >client can tell me if my contacts are online or offline.
> >
> >But the two ways don't display the same information, it means,
> from
> >serweb I add several sip address (I check that are recorded at ser
> >database) but this addresses are not imported when I start the SIP
> >client.
> >
> >I need my client import sip contacts ( in a very similar way like
> MSN
> >Messenger) from server. Is there any way to do this? Does the
> client
> >have to make a request to the server or is the server who has to
> send
> >this information when it receives a REGISTER?
>
> SER sends all registered contacts in replies to REGISTER. If you wish
> your phone to display them, ask your phone's vendor. Look otherwise
> in serweb.
>
> -jiri
>
>
> >Or maybe SER is not intended to do this?
> >
> >Thank you very much for your help.
> >
> >Curro
> >
> >
> >_______________________________________________
> >Serusers mailing list
> >serusers(a)lists.iptel.org
> >http://lists.iptel.org/mailman/listinfo/serusers
>
> --
> Jiri Kuthan http://iptel.org/~jiri/
>
>
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Thanks Jiri, Jesus
You were right, it was an issue about permissions.
----- Mensaje Original -----
De: Jiri Kuthan <jiri(a)iptel.org>
Fecha: Miercoles, Enero 14, 2004 12:41 pm
Asunto: Re: [Serusers] Error: sorry --cannot open write fifo
> make sure that your web server has write permissions to access the
> fifo.
> At 11:45 AM 1/14/2004, CURRO_DOMINGUEZ wrote:
> >Hello
> >
> >I continue with my test of SER. SER is running almost OK, but I
> have
> >some problems. Sorry if this is a newbie question, but is really
> >important to me configure SER correctly.
> >
> >When I try to enter to serweb user_interface, I have this error:
> >
> >sorry --cannot open write fifo
> >
> >I can add, delete contacts with serweb, but I can't send Instant
> >Messages(nor whit Windows Messenger). I have read fifo is the
> >communication way between ser and serweb, and responsible for
> instant
> >messages.
> >
> >In config.php, the configuration is
> >$this->fifo_server="/tmp/ser_fifo"
> >And in ser.cfg the configuration is
> >fifo="/tmp/ser_fifo"
> >
> >I think is OK, isn't it?
> >
> >What can be wrong?
> >
> >One more time, thank you very much for your time an knowledge.
> >
> >Curro
> >
> >
> >_______________________________________________
> >Serusers mailing list
> >serusers(a)lists.iptel.org
> >http://lists.iptel.org/mailman/listinfo/serusers
>
> --
> Jiri Kuthan http://iptel.org/~jiri/
>
>
Hi,
There is a SIP client from Siemens called SCS-Client:
http://mysip.ch/index.html
Has anybody ever tried to use the SCS-Client with SER?
According to its specification, it can provide audio,
video, presence and IM functions much similar to MS's
Windows Messenger 5.0.
Since the IM part of Messenger 5.0 can only work correctly
with MS's Live Communications server, I was just trying to
find another client with similar features that can work
with SER.
Thanks,
Kevin
Hello
I continue with my test of SER. SER is running almost OK, but I have
some problems. Sorry if this is a newbie question, but is really
important to me configure SER correctly.
When I try to enter to serweb user_interface, I have this error:
sorry --cannot open write fifo
I can add, delete contacts with serweb, but I can't send Instant
Messages(nor whit Windows Messenger). I have read fifo is the
communication way between ser and serweb, and responsible for instant
messages.
In config.php, the configuration is
$this->fifo_server="/tmp/ser_fifo"
And in ser.cfg the configuration is
fifo="/tmp/ser_fifo"
I think is OK, isn't it?
What can be wrong?
One more time, thank you very much for your time an knowledge.
Curro
Hi.
Can I forward SER calls to a h323 gatekeeper, or how to forward SER calls to
vocal sip server?
--
|o
|o
|o Fabio Silvestri
|o fabio(a)informatec.com.br
|o ICQ: 1667351
|o
--- Jiri Kuthan <jiri(a)iptel.org> wrote:
> in front of the t_relay which forwards the
> transaction you wish to report on. -jiri
>
The only place I have t_relay is the loose forward
section.
if (loose_route()) {
t_relay();
break;
};
Should I place setflag(3) before t_relay() in this
section? Or should it be in the section in which the
call is made?
In my ser.cfg (entire file is at the bottom of this
page) the call is made with the following lines:
if (uri=~"sip:1[0-9]+@.*") {
rewritehostport("sip.provider.net:5060");
setflag(2);
if(!t_relay()) {
sl_reply_error();
};
break;
};
::My ser.cfg ::
# ----------- 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"
alias=mysip.mydomain.com
# ------------------ module loading
----------------------------------
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.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/acc.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters
---------------
modparam("usrloc", "db_mode", 2)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("rr", "enable_full_lr", 1)
modparam("acc", "log_level", 2)
modparam("acc", "db_url",
"sql://ser:heslo@localhost/ser")
modparam("acc", "db_flag", 2)
# ------------------------- 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;
};
# 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)
# 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 (uri=~"sip:1[0-9]+@.*") {
rewritehostport("switch.myprovider.net:5060");
setflag(2);
if(!t_relay()) {
sl_reply_error();
};
break;
};
if (uri=~"sip:011[0-9]+@.*") {
rewritehostport("switch.myprovider.net:5060");
setflag(2);
if(!t_relay()) {
sl_reply_error();
};
break;
};
if (uri==myself) {
if (method=="REGISTER") {
if
(!www_authorize("mysip.mydomain.com", "subscriber")) {
www_challenge("mysip.mydomain.com", "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();
};
lookup("aliases");
save("aliases");
}
=====
Asterisk is my lover, and IAX2 is our scented lubricant
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Hi to all,
I have on my ser.cfg many dial plans forwarding calls to several Cisco PSTN, now
I need to configure, if the call could not find a specific dial plan the forward
to a second SIP proxy!
How can I do this?
Regards.
--
|o
|o
|o Fabio Silvestri
|o fabio(a)informatec.com.br
|o ICQ: 1667351
|o
Hi guys,
I need to setup a system, based on SER that can handle, for a particular
class of users, a prepaid call scenario.
I'm aware that I do need a B2BUA for this particular case.
I've been trying to mount a scheme like this, routing some calls from SER to
Vovida's B2BUA, but I'm stuck in the authentication process. I can't arrange
a way of authenticating SER users on B2BUA...
Has anyone tried a setup like this?
Has anyone other suggestions of accomplishing this?
I believe Maxim is developing a B2BUA, but I need a short term solution...
Thanks,
Edgar
I recently set up database accounting with SER and I
was testing it out to see if there were any situations
in which the logging might fail. I found that if I
unplug my SIP device during a call, then a SIP BYE
message will not be created in my database. Is there
any configuration change I can make to fix this
problem, or is this due to the design of the SIP
protocol?
Thank you for your time.
Just in case it is a configuration problem, I have
included my ser.cfg
#ser.cfg ###
# ----------- 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"
alias=sip.mydomain.com
# ------------------ module loading
----------------------------------
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.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/acc.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters
---------------
modparam("usrloc", "db_mode", 2)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("rr", "enable_full_lr", 1)
modparam("acc", "log_level", 2)
modparam("acc", "db_url",
"sql://ser:heslo@localhost/ser")
modparam("acc", "db_flag", 2)
# ------------------------- 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;
};
# 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)
# 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()) {
if (method == "BYE") {
setflag(2);
};
t_relay();
break;
};
if (uri=~"sip:1[0-9]+@.*") {
rewritehostport("sip.provider.net:5060");
setflag(2);
if(!t_relay()) {
sl_reply_error();
};
break;
};
if (uri=~"sip:011[0-9]+@.*") {
rewritehostport("sip.provider.net:5060");
setflag(2);
if(!t_relay()) {
sl_reply_error();
};
break;
};
if (uri==myself) {
if (method=="REGISTER") {
if
(!www_authorize("sip.mydomain.com", "subscriber")) {
www_challenge("sip.mydomain.com", "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();
};
lookup("aliases");
save("aliases");
}
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