If you want to Sip REGISTER your phone to asterisk change the host=192.168.10.193 section of the [101] section to host=dynamic
Currently you are telling asterisk that sip user 101 is on host 192.168.0.193, which is you asterisk box, so when a call goes to 101, asterisk sends it to itself and then tries to connect the incoming sip call to 101, hence the loop :) Kind regards, E. Versaevel -----Oorspronkelijk bericht----- Van: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Namens Adnan Ahmed Verzonden: maandag 22 november 2004 21:34 Aan: [EMAIL PROTECTED] Onderwerp: [Asterisk-Users] SIP Problem! hi, I am not registered my SIP Phone with Asterisk i spend almost one day but find no luck.I know very well this is not kind a problem discussed in this group but i try my best and all in vein so finally i am here hoping you ppl helping me out.I discussed this problem in asterisk's-users group and adding feedback from asterisk-users group my configs are sip.conf [general] port=5060 bindaddr=192.168.10.193 allow=all [101] username=101 type=friend secret=12345678 host=192.168.10.193 context=from-sip callerid="101"<101> defaultip=192.168.10.176 extensions.conf [globals] 101=SIP/101 [incoming] exten => s,1,Dial(Zap/1,20) exten => s,2,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${announce}) exten => s-NOANSWER,2,Goto(incoming,s,1) exten => s,3,NoOp,$(CALLERID) include => outgoing include => from-sip callerid=yes [outgoing] exten => _NXXXXXX,1,Dial/Zap/4/${EXTEN:0} exten => _0NXXXXXXXX,1,Dial,Zap/4/${EXTEN:0} exten => _0NXXXXXXXXX,1,Dial,Zap/4/${EXTEN:0} exten => _0NXXXXXXXXXX,1,Dial,Zap/4/${EXTEN:0} exten => 101,1,Dial(101,20) include => from-sip include => incoming [sip] exten => 101,1,Dial(${101,20}) exten => 101,2,VoicemailMain exten => 101,3,Hangup include => outgoing include => from-sip here are the console output : :-X ). *cli> --Starting simple switch on 'Zap/1-1' Executing Dial(" "," ") in new stack Called 101 Got SIP Responce 482 "Loop Detected" back from 192.168.10.193 No one is available to answer qt this time Executing VoiceMailMain(" ","") in new stack Playing 'vm-login' (language 'en' ) Username not entered Executing Hangup(" ","") in new stack Spawn Extension (outgoing , 101, 3) exited non-zero on 'Zap/1-1' Hangup 'Zap/1-1' *cli>sip show registry Host Username Refresh State *cli>sip show users Username Secret Authen Def.Context A/C 101 12345678 md5,plaintext sip No *cli>sip show peers Name/Username Host Mask Port Status 101/101 192.168.10.195 255.255.255.255 5060 Unmonitored *cli>sip show channels Peer User/ANR Call ID Seq (Tx/Rx) Lag Jitter Buffer 0 active SIP channel(s) Kindly pointout my mistakes/errors and helping me out. Any Help Is Highly Appreciated. Thanks in Advance. Adnan Ahmed. _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users