[asterisk-bugs] [Asterisk 0013291]: Not able to make outgoing call [Failed to authenticate on INV ITE]
Asterisk Bug Tracker
noreply at bugs.digium.com
Mon Aug 11 23:16:04 CDT 2008
The following issue has been SUBMITTED.
======================================================================
http://bugs.digium.com/view.php?id=13291
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Reported By: sujit
Assigned To:
======================================================================
Project: Asterisk
Issue ID: 13291
Category: Addons/General
Reproducibility: always
Severity: major
Priority: normal
Status: new
Asterisk Version: Older 1.4
SVN Branch (only for SVN checkouts, not tarball releases): N/A
SVN Revision (number only!):
Disclaimer on File?: N/A
Request Review:
======================================================================
Date Submitted: 2008-08-11 23:16 CDT
Last Modified: 2008-08-11 23:16 CDT
======================================================================
Summary: Not able to make outgoing call [Failed to
authenticate on INV ITE]
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 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 => 31045850:58503008 at 203.126.17.242:5060/31045850 ;added by
sujit to register to external sip server
register => 31045851:58513008 at 203.126.17.242:5060/31045851 ;added by
sujit to register to external sip server
[authentication]
auth=31045850:58503008 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=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()
======================================================================
Issue History
Date Modified Username Field Change
======================================================================
2008-08-11 23:16 sujit Asterisk Version => Older 1.4
2008-08-11 23:16 sujit SVN Branch (only for SVN checkouts, not tarball
releases) => N/A
======================================================================
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