[asterisk-dev] Not able to make outgoing call [Failed to authenticate on INVITE]

Tue Aug 12 22:30:06 CDT 2008

Hi friend,
 I am using 
----------------------------------------------------------
 Asterisk 1.2.7.1-1.0.0 built by mindspeed @ newubuntu
 two SIP accounts 31045850and 31045851
 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 at 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:XXXXXXXXX at 203.126.17.242:5060/31045850  ;added by
sujit to register to external sip server
register => 31045851:YYYYYYY at 203.126.17.242:5060/31045851  ;added by
sujit to register to external sip server

[authentication]
auth=31045850:XXXXXXXXX at 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=ZZZZZZZ
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()

====================================================================== 

Thanks and Regards,
    Sujit Das
    Aztech Systems, Singapore