Hi,
I currently have the following setup and working chris(a)sip.domain.com
incoming sip calls work fine. I want to be able to setup SER to forward
incoming packets for chris(a)domain.com to chris(a)sip.domain.com.
I've tried the following but it didn't work.
route{
if (uri=~"sip:chris@domain.com") {
forward(sip.domain.com, 5060);
}
Can anybody help?
Kind Regards,
Chris Bond
hello friends,
mines os is-Linux i686 i686 i386 GNU/Linux
ser is -ser 0.8.12-tcp_nonb (i386/linux)
iam trying to find out the least configuration for the
ser to run .
i.e i just want to establish a seesion between two sip
end points which are in public ip and the ser is also
in public ip .
so with my analysis i found this as the least,
can any body suggest that further decrease the
statements .
with regrards
rama kanth
ser.cfg is
---------------------------------------------------
---- global configuration parameters
------------------------
debug=8 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (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/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"
# ----------------- setting module-specific parameters
---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# ------------------------- request routing logic
-------------------
# main routing logic
route{
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;
};
record_route();
if (loose_route()) {
t_relay();
break;
};
if (uri==myself) {
if (method=="REGISTER") {
save("location");
break;
};
if (!lookup("location")) {
sl_send_reply("404", "Not
Found");
break;
};
};
if (!t_relay()) {
sl_reply_error();
};
}
__________________________________
Do you Yahoo!?
New and Improved Yahoo! Mail - Send 10MB messages!
http://promotions.yahoo.com/new_mail
hello,
I have install SER on my network.user can register it,but have any problem:
first,when we use NAT,private network users can't dial each other ,but can dial any public network users.
second,my pstn gateway is cisco5300(c5300-is-mz.123-9.bin),i config it ,but it don't work now.i don't know it is 5300 confiration error or ser.cfg error.
please look attachment files.
thank you!
luhaolong 路浩龙
CERNET CORPORATION
http://www.cernet.com/
Tel:+86 10 6522 6655 ext.2305
Fax:+86 10 8518 1010
MP:+86 139 1105 7110
Add:10th Floor,W1,the Towers
Oriental Plaza,Beijing 100738
How can you configure SER to process a 302 redirect, rather then sending it
back to the SIP Client?
When my SIP client places a call through SER to a destination, and the
destination replies with a 302 redirect, SER seems to proxy that redirect
back to my SIP client. Rather then have my client send a second invite with
the new "Contact" information, I would rather have SER directly send an
invite to the new destination without telling my client about the 302
redirect.
I thought about using the t_on_failure, as a response code >=300 will
trigger a failure_route, but how do you pull the content out of the
"Contact" field of the response to put into the request URI?
- Robert Liao
hello friends,
1)i want to know what is the extra facilities which
we get if we use the subscription to the ser
(i.e with serctl add or serweb giveng the access
to the users).
i am using the free radius authentication and its
working fine for me.
2)can we use the database to the free
radius server rather than just appending the username
,
password in the file of users in free radius server.
3)can we put the inter relation betwen the radius and
the subscribe table of the ser so that we need not
manually enter the details into the users file as and
when i need to add the users to the ser system.
__________________________________
Do you Yahoo!?
New and Improved Yahoo! Mail - Send 10MB messages!
http://promotions.yahoo.com/new_mail
I did not think so much about watching URIs, rather just keeping some
internal status associated with transaction in progress.
A possibility would be to cascade through failure_routes:
failure_route[1] {
t_on_failure(2); # if this fails, try 2
...
}
failure_route[2] {
t_on_failure(3); # if this fails, try 3
...
}
etc.
Other possibilty I thought of would be
route[1] {
...
setflag(1); # state == 1
t_relay();
...
failure_route[1] {
if (isflagset(1)) {
# try 2
resetflag(1); setflag(2);
break;
};
if (isflagset(2)) {
# try 3
resetflag(2); setflag(3);
break;
};
...
-jiri
At 03:21 AM 7/14/2004, Michael Przybylski wrote:
>I was thinking along the lines of using the textops.so search() function.
>
>Will that let me see what "Contact" headers have already been appended as
>branches?
>
>Best regards,
>Mike Przybylski
>
>On Wed, 14 Jul 2004, Jiri Kuthan wrote:
>
>> A possibility to keep status would be to use flags -- a poor variable
>> substitute till support for variables appears.
>>
>> -jiri
>>
>> At 10:14 PM 7/13/2004, Michael Przybylski wrote:
>> >Hello everyone,
>> >
>> > I've got a recursing failure_route block that I'm trying to use
>> >for voice over IP gateway failover.
>> > The recursion bit works without any problem, but I could really
>> >use some hints on examining the branches appended by previous calls to the
>> >failure route so that it can know when to stop.
>> >
>> > Here's the pseudo-code version of what I'm trying to do.
>> >
>> >if(examing previously appended branches shows
>> > IP of last-chance gateway already there){
>> > give up;
>> >}
>> >
>> > Any suggestions would be greatly appreciated.
>> >
>> >Best regards,
>> >Mike Przybylski
>> >
>> >_______________________________________________
>> >Serusers mailing list
>> >serusers(a)lists.iptel.org
>> >http://lists.iptel.org/mailman/listinfo/serusers
>>
>> --
>> Jiri Kuthan http://iptel.org/~jiri/
>>
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
I installed SER without authentication or my_sql
support a few dyas back. When I realized that I needed
to keep some call records and enable authentication, I
installed SER_MYSQL. But SER doesn't automatically use
the SER database for call accounting. What changes do
I need to make to my ser.cfg (or any other file) to
enable accounting support with database after
installing SER_MYSQL?
Thanks.
__________________________________
Do you Yahoo!?
Read only the mail you want - Yahoo! Mail SpamGuard.
http://promotions.yahoo.com/new_mail
Hello:
I have a SER proxy server, and 2 SIP-to-pstn gateways (GW1, GW2). I would
like to provide redundancy with SER using DNS SRV entries.
The scenario is: every call must be directed to GW1, and **only** if GW1
fails, these calls must be directed to GW2.
I've created these dns entries in Bind:
_sip._udp.my.service.com. IN SRV 10 0 5060 gw1.service.com.
_sip._udp.my.service.com. IN SRV 20 0 5060 gw2.service.com.
gw1.service.com. IN A 192.168.1.1
gw2.service.com. IN A 192.168.1.2
service.com. IN A 192.168.1.10
SER receives calls to "user1(a)service.com", and after a dns srv lookup it
sends the call to gw1.service.com correcly. SER configuration is:
if (uri=~"user1"){
rewritehost("my.service.com");
forward(uri:host, uri:port);
break;
};
But, if gw1.service.com is down, SER never tries to send the calls to
gw2.service.com.
SER never send INVITEs to the server with priority 20.
Is it a SER limitation, or is it a configuration problem?
Thank you,
Mark
I would like to provide failover redundancy with SER
Hi to all
I've a small problem. When I try to REGISTER with MSN i can see this message
U 157.27.242.227:4014 -> 217.57.90.117:5060
REGISTER sip:sip.quellidelpirana.it SIP/2.0..
Via: SIP/2.0/UDP 157.27.242.227:16757..
From:
<sip:giandrea@sip.quellidelpirana.it>;tag=3a965a5f-1b25-4144-b509-ed6fb6f60d6e..
To: <sip:giandrea@sipquellidelpirana.it>..
Call-ID: fd9e04be-a1c2-472b-b78d-da18f2dbf3a3(a)157.27.242.227..
CSeq: 1 REGISTER..
Contact: <sip:157.27.242.227:16757>;methods="INVITE, MESSAGE, INFO,
SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK"..
User-Agent: Windows RTC/1.0..
Expires: 1200..
Event: registration..
Allow-Events: presence..
Content-Length: 0....
but when I try to use Xlite I can see this
U 157.27.242.227:5060 -> 217.57.90.117:5060
REGISTER sip:sip.quellidelpirana.it SIP/2.0..
Via: SIP/2.0/UDP
157.27.242.227:5060;rport;branch=z9hG4bK43C50241B19C4F9BA5F82F0922D526C1..
From: Andrea <sip:giandrea@sip.quellidelpirana.it>..
To: Andrea <sip:giandrea@sip.quellidelpirana.it>..
Contact: "Andrea" <sip:giandrea@157.27.242.227:5060>..
Call-ID: CFBD07105A04447A86952238875CEC95(a)sip.quellidelpirana.it..
CSeq: 35983 REGISTER..
Expires: 1800..
Authorization: Digest
username="giandrea",realm="sip.quellidelpirana.it",nonce="40f3e286cc783dea71369a49455a559eb8a4d341",response="a733f8ddfbc5cf07ba23ca368e750cdb",uri="sip:sip
.quellidelpirana.it"..
Max-Forwards: 70..
User-Agent: X-Lite build 1101..
Content-Length: 0....
why MSN doesn't insert Authorization???
Why I can use iptel account but not my sip proxy account? Have I to change
my ser.cfg?
Can I read a SIP tag from my configuration ?
With MSN the www_autorize fail
if (uri==myself) {
xlog("L_INFO", "%rm / %mi --> uri==myself");
if (method=="REGISTER") {
xlog("L_INFO", "%mf / %ct");
if (!www_authorize("sip.quellidelpirana.it", "subscriber")) {
log(1, "Auth: NO - requesting authentification\n");
www_challenge("sip.quellidelpirana.it", "0");
break;
} else {
log(1, "Auth: YES");
};
save("location");
break;
};
};
Andrea