Summary: | ASTERISK-12568: Not able to make outgoing call [Failed to authenticate on INV ITE] | ||
Reporter: | sujit (sujit) | Labels: | |
Date Opened: | 2008-08-11 23:16:02 | Date Closed: | 2011-06-07 14:02:44 |
Priority: | Major | Regression? | No |
Status: | Closed/Complete | Components: | Addons/General |
Versions: | Frequency of Occurrence | ||
Related Issues: | |||
Environment: | Attachments: | ||
Description: | Hi friend, I am using ---------------------------------------------------------- Asterisk 1.2.7.1-1.0.0 built by mindspeed @ newubuntu two SIP accounts 31045850/58503008 and 31045851/58513008 SIP server IP: 203.126.17.242:5060. ---------------------------------------------------------- This SIP server is live server. After registering the two accounts in the SIP server, I can make incoming call from my mobile to the 31045850 but while making outgoing call it is failing and showing ================================================================================== Jan 1 00:16:30 NOTICE[11800]: chan_sip.c:9776 handle_response_invite: Failed to authenticate on INVITE to '"31045850" sip:31045850@203.126.17.242>;tag=as6651c09d' ================================================================================ After sending ACK for "401 Unauthorized" (sent by Server). Please help to resolve this issue. Thanks ! Sujit Das // Here is complete sip.conf ======================================================================== sip.conf ======================================================================== ; ;DO NOT MODIFY, FILE AUTOMATICALLY GENERATED BY SCRIPT configure_asterisk ; [general] CONTEXt=default ; Default context for incoming calls bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all allow=ulaw,alaw,g729 ;also defines preference dtmfmode=rfc2833 tos=0x10 defaultexpiry=3600 ;added by Sujit default registration expiry timer fromdomain=203.126.17.242 ;sujit register => 31045850:58503008@203.126.17.242:5060/31045850 ;added by sujit to register to external sip server register => 31045851:58513008@203.126.17.242:5060/31045851 ;added by sujit to register to external sip server [authentication] auth=31045850:58503008@203.126.17.24:5060 [my-singtel-server] type=peer fromuser=31045850 secret=58503008 host=203.126.17.242 canreinvite=no insecure=invite qualify=yes nat=no disallow=all allow=gsm allow=ulaw allow=alaw [2345] type=peer context=default ; Where to start in the dialplan when this phone calls username=2345; SIP username for registration secret=2345; SIP password for registration host=dynamic ; Sip phone has a dynamic IP address canreinvite=yes ; allow RTP voice traffic to bypass Asterisk insecure=invite [31045849] type=friend context=default username=31045849 secret=58493008 callerid=31045849 host=dynamic canreinvite=no [202] type=friend context=default username=202 secret=202 callerid=202 host=dynamic canreinvite=no [203] type=friend context=default username=203 secret=203 callerid=203 host=dynamic canreinvite=no [204] type=friend context=default username=204 secret=204 callerid=204 host=dynamic canreinvite=no // Here is complete extensions.conf ======================================================================== extensions.conf ======================================================================== ; ;DO NOT MODIFY, FILE AUTOMATICALLY GENERATED BY SCRIPT configure_asterisk ; [globals] ;Sujit [general] static=yes writeprotect=no autofallthrough=yes ;Sujit [default] include => phones include => parkedcalls ;sujit - start [phones] ; sujit include => internal ; sujit include => remote ; sujit [internal] ;exten => _1XX,1,NoOp() ;exten => _1XX,n,Macro(stdexten, SIP/${EXTEN},30) ;exten => _1XX,n,Playback(the-number-is-unavail) ;exten => _1XX,n,Hangup() [remote] exten => _9XXXXXXX,1,NoOp() exten => _9XXXXXXX,n,Macro(stdexten,SIP/${EXTEN}@my-singtel-server,30) ;connecting to other network which has 1XX numbers thru SIP protocol ;A.B.C.D is IP-Address of other board ;replace A.B.C.D and reload configuration files ;exten => _4xx,1,Macro(stdexten,SIP/${EXTEN}@A.B.C.D,,401) ;connecting to other network which has 2XX numbers thru SIP protocol exten => _5xx,1,Macro(stdexten,SIP/${EXTEN}@A.B.C.D,,401) exten => 101,1,Macro(stdexten,MSPD/phone0,,101) exten => 31045850,1,Macro(stdexten,MSPD/phone1,,31045850) exten => 31045851,1,Macro(stdexten,MSPD/phone2,,31045851) exten => 104,1,Macro(stdexten,MSPD/phone3,,104) exten => 31045849,1,Macro(stdexten,SIP/31045849,,31045849) exten => 202,1,Macro(stdexten,SIP/202,,202) exten => 203,1,Macro(stdexten,SIP/203,,203) exten => 204,1,Macro(stdexten,SIP/204,,204) exten => s,1,GotoIf($[${LEN(${ARG3})} > 0]?4) exten => s,2,SetVar(VMBOX=${MACRO_EXTEN}) exten => s,3,Goto(5) exten => s,4,SetVar(VMBOX=${ARG3}) exten => s,5,Dial(${ARG1},20,t${ARG2}) exten => s,6,Goto(s-${DIALSTATUS},1) exten => s-ANSWER,1,Hangup exten => s-BUSY,1,Voicemail(b${VMBOX}) exten => s-BUSY,2,Hangup exten => _s-.,1,Voicemail(u${VMBOX}) exten => _s-.,2,Hangup exten => 1234,1,VoiceMailMain() exten => 1234,1,NoOp(${EXTEN}) exten => 1234,2,NoOp(${MACRO_EXTEN}) exten => 1234,3,Hangup() [macro-stdexten] exten => s, 1, Dial(${ARG1}, 25, tT) exten => s, 2, SetVar(VMBOX=${MACRO_EXTEN}) exten => s, 3, NoOp(${MACRO_EXTEN}) exten => s, 4, NoOp(${VMBOX}) exten => s, 5, Goto(s-${DIALSTATUS},1) ;exten => s-ANSWER,1,Hangup ; sujit exten => s-ANSWER,1,Goto(1) ; sujit exten => s-BUSY,1,Voicemail(b${VMBOX}) ;exten => s-BUSY,2,Hangup ; sujit exten => s-BUSY,2,Goto(1) ; sujit ;exten => _s-.,1,Voicemail(u${VMBOX}) exten => _s-.,1,Goto(1) ; sujit ;exten => _s-.,2,Hangup ; sujit exten => _s-.,2,Goto(1) ; sujit #exten => s, 2, Goto(s, 102) #exten => s, 102, Playback(vm-nobodyavail) #exten => s, 103, Hangup() | ||
Comments: | By: Leif Madsen (lmadsen) 2008-08-12 12:42:48 This appears to be an issue with your configuration, and not a bug in Asterisk. The appropriate place to get general debugging help is either on the #asterisk IRC channel on the Freenode IRC network, or on the asterisk-users mailing list available at http://lists.digium.com (you can sign up there and then follow the instructions for posting to the mailing list). Good luck! Leif. |