At 12:09 PM 12/1/2003, Klaus Darilion wrote:
>Hi!
>
>How do I reach an ENUM subscriber when I'm registered to iptel.org? It works
>using the '+' sign and the number (+43699...). But how can I call such a
>number using a SIP-phone which doesn't support the '+' sign (like
>BudgeTone)? I tried using double zeros (0043699...) but that doesn't work.
Its changed to allow * as the lead character too.
-jiri
Would anyone know the whereabouts for the SER config file that goes with the RTPProxy utility that available from the SER site?
Thanks,
John Hermanski
John Hermanski
Sr. Technical Marketing Engineer
Intel Americas, Inc.
Worldwide Technical Sales and Support - Telecom
phone: 978-744-9098
cell: 978-836-8028
mailto:john.hermanski@intel.com
http://www.dialogic.com
I have a problem CANCELING requests when using t_on_failure. Example:
I forward the call to the PSTN gateway. Then I CANCEL the request. I am
using a second PSTN gateway for backup and I call the function
t_on_failure before t_relay:
t_on_failure("2");
if (!t_relay()) {
sl_reply_error();
};
As a result of the CANCEL request, a 487 code is received and
t_on_failure redirects the requests to the second PSTN gateway. This is
not what I expect after sending a CANCEL. The manual says all responses
> 300 are considered failures but an intended CANCEL is not a failure,
is it?
Do I miss something in the configuration to stop this from happening?
Regards,
Adrian Georgescu
ag(a)ag-projects.com
http://ag-projects.com
Tel: +31-23-5458104
IP phone: sip:ag@ag-projects.com
------------------------------------------------
DNS, ENUM & IP telephony http://managed-dns.org/
What kind of gateways are you using? 213 or 613?
I'm using both without any problems. Please post your rg config..
Tor.
>Date: Sun, 30 Nov 2003 19:59:37 -0500
>From: "Andy Vander Woude" <andyv(a)sympatico.ca>
>Subject: [Serusers] Residential Gateways
>To: <serusers(a)lists.iptel.org>
>Message-ID: <000001c3b7a6$68560d70$8adaacce@generator>
>Content-Type: text/plain; charset="us-ascii"
>
>
>Running Redhat V9 and have installed the SIP Express Router (ser)
>packages. Have two Allied Telesyn Residential Gateways (192.168.20.2 and
>192.168.20.3) and laptop (192.168.20.4) running ser connecting to a
>managed switch IP address (192.168.20.1 (gateway))
>The RG's are configured to access the Proxy server, & Domain server at
>192.168.20.4 and the Gateway as 192.168.20.1 The Domain is called
>ati.com.
>
>When connected all together I get a dial tone on the phones, however
>when I dial the # configured on the RG's I do not get a ring and they do
>not register with the ser server. When I type in the "ser start" command
>I see the alias entries as "127.0.0.1 localhost localdomain localhost"
>and "192.168.20.4 localhost.ati.com localhost"
>
>The entries in the ser.cfg relating to the localdomain have been changed
>to ati.com
>I have added the 192.168.20.4 to the hosts.cfg relating to the ati.com
>domain.
>
>Any idea as to why this configuration is not working. This is a lab
>situation with no outside connections.
>Thank you.
>
>
---
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Hi all,
I use function "save" to save use information when
user register, like this:
if (method=="REGISTER") {
if (!www_authorize("novsky.com",
"subscriber")) {
www_challenge("novsky.com",
"1");
break;
};
save("location");
break;
};
But after a SIP UA registed, I sometimes can't see the
use in mysql table ser.location. Is it right?
And if a user registerd from many different IP
address, there are many log about the user in mysql
table ser.location.
wangji
12/1/03
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Hi!
How do I reach an ENUM subscriber when I'm registered to iptel.org? It works
using the '+' sign and the number (+43699...). But how can I call such a
number using a SIP-phone which doesn't support the '+' sign (like
BudgeTone)? I tried using double zeros (0043699...) but that doesn't work.
regards,
Klaus
Hi all,
The problem is I can't register two SIP UAs to SER
server.
This is my environment:
Internet
|
router (X.X.X.163)
|
|Fiber |---- Ser server (X.X.X.166)
| |
Switch------|---- Trunk Gateway (X.X.X.167)
|
|---- Other servers
|
|---- Router( cisco 26xx) (X.X.X.163)
|
Switch
|
_____________________|_________________
Users SIP UA1(172.16.2.11) SIP UA2(172.16.2.23)
My ser.cfg like:
route{
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too
big");
break;
};
record_route();
if (loose_route()) {
t_relay();
break;
};
if (!method=="BYE") {
force_rport();
fix_nated_contact();
};
if (method=="REGISTER") {
if (!www_authorize("novsky.com",
"subscriber")) {
www_challenge("novsky.com",
"1");
break;
};
save("location");
break;
};
if (method=="INVITE") {
if(!proxy_authorize("novsky.com","subscriber")) {
proxy_challenge("novsky.com","1");
break;
};
};
if(uri=~"^sip:6169[0-9]*@[0-9|\.]*") {
if(!lookup("location")) {
sl_send_reply("404", "Not
Found");
break;
};
}else {
forward(172.16.255.2,6050);
break;
};
if (!t_relay()) {
sl_reply_error();
};
}
I use ethereal to look the packet, first UA1 send
REGISTER to SER server, on NAT it use 5060 port, and
SER return to corrent address/port.
When UA2 send REGISTER to SER server, on NAT it use
1027 port, but SER return to NAT:5060 port and the
packet transfered to UA1.
So UA2 can't register.
I can see the NAT table on route, like
udp x.x.x.165:5060 172.16.2.11:5060 x.x.x.165:5060
x.x.x.x.165:5060
udp x.x.x.165:5060 172.16.2.23:5060 x.x.x.165:1025
x.x.x.x.165:1025
Wangji
12/1/03
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