13 May
2003
13 May
'03
1:25 p.m.
On Tue, 13 May 2003 seb.peterson(a)internet.lu wrote:
Hi,
I'm doing a few tests to try SER.
I use a simple configuration:
-two Cisco ip phones
-one Cisco 1760 Voice gateway with ISDN interfaces.
Calling between the ip phones works, but going through
the Voice Gateway is something else... I don't really
understand how to edit the config file, and I suppose
their's a simple way. Here's how I tried.
I added these lines to the default config:
if (uri=~"^sip:452564@10.1.1.22") {
t_relay_to("10.1.1.240","5060");
break;
};
And I call always teh same number 452564. 10.1.1.240 being
my gateway, and 10.1.1.22 the SIP proxy server.
An extract of the debbuging info:
...
6(13970) SIP Request:
6(13970) method: <ACK>
6(13970) uri: <sip:452564@10.1.1.22>
6(13970) version: <SIP/2.0>
6(13970) parse_headers: flags=1
...
6(13970) DEBUG: add_param: tag=d907c037823644515dfe0ede38ca9976.1dfc
6(13970) end of header reached, state=29
6(13970) DEBUG: get_hdr_field: <To> [66]; uri=[sip:452564@10.1.1.22]
6(13970) DEBUG: to body [<sip:452564@10.1.1.22>]
6(13970) DEBUG: sl_filter_ACK : local ACK found -> dropping it!
6(13970) error:...
..
If you see something big, please let me know.
THanks.
Sebastien Peterson
Luxembourg.
See http://lists.iptel.org/pipermail/serusers/2003-May/001315.html
I had this same problem but Jan solved this
Try it:
if (uri=~"sip:452564@10\.1\.1\.22") {
if (!t_relay_to("10.1.1.240", "5060")) {
sl_reply_error();
};
break;
t_relay();
};
If you have VoIP gateway configure properly
the PSTN phone 452564 will ring on your desk.
Andrzej Radke