search for: 41150

...168.12.160> From: "1002"<sip:1002 at 192.168.12.160>;tag=5f799517 Call-ID: Yjk5ZTFjZjI1Mzc5N2RiMWFhYWRiMjhhMjc0NTU4ZDI. CSeq: 1 REGISTER Expires: 3600 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INF O User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 192.168.25.116 : 52166 (NAT) localhost*CLI> <--- Transmitting (NAT) to 192.168.25.116:52166 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.25.116:52166;branch=...

Hi My asterisk output is: chan_sip.so => (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 201 -- Executing [907768385144 at default:1] Dial("SIP/201-083e75c0", "ZAP/g1/907768385144|60") in new stack [Oct 4 11:54:27] WARNING[6255]: channel.c:3388 ast_request: No channel type registered for 'ZAP' [Oct 4 11:54:27] WARNING[6255]: app_dial.c:1...

Hi, I am trying to get a SIP extension's status without actually making a call. I am using sofia-sip's "options" example utility and the sip clients are SJphone softphones.

...t;sip:poczta at voip.rd.touk.pl:7170> To: <sip:hellboy at voip.rd.touk.pl:7060>;tag=12711d6c From: "IPFon"<sip:0225761853 at 192.168.0.165:7070>;tag=as66773d49 Call-ID: 08027b6d45abc7a00c2c6a8630d2de47 at 192.168.0.165 CSeq: 102 INVITE User-Agent: X-Lite release 1011s stamp 41150 Content-Length: 0 <-------------> [Jan 4 10:43:27] --- (9 headers 0 lines) --- [Jan 4 10:43:27] -- Got SIP response 302 "Moved Temporarily" back from 192.168.0.165 [Jan 4 10:43:27] Transmitting (no NAT) to 192.168.0.165:7060: ACK sip:hellboy at voip.rd.touk.pl:7060 SIP/2.0...