On Tue, 13 May 2003 seb.peterson@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