[SIPForum-discussion] line jumping after restarting

[aamir chougule]

Hi Team,

 

 

I have been facing some problem after restarting the
asterisk; the problem is given below:

 

Lines configured in the extensions.conf

 

[macro-stdexten-abcd];;--Standard flow for all the
extensions--

exten => s,1,Set(DYNAMIC_FEATURES=automon-abcd)

exten => s,n,gotoif(${DB_EXISTS(DND/${ARG1})}?dnd)

exten => s,n,gotoif(${DB_EXISTS(CFIM/${ARG1})}?fim:dial)

 

exten => s,n(dnd),Playback(do-not-disturb)

exten => s,n(dnd),Playback(please-try-again-later)

exten => s,n(dnd),wait(2)

exten => s,n(dnd),hangup()

 

exten => s,n(fim),GotoIf($[${LEN(${CALLERID(num)})} = 4]?yesim:noim)

exten => s,n(yesim),GotoIf($[${LEN(${DB(CFIM/${ARG1})})}
= 4]?tim:nim)

exten =>
s,n(tim),Macro(stdexten-abcd,${DB(CFIM/${ARG1})},SIP/${DB(CFIM/${ARG1})})

exten => s,n(nim),Macro(cidcfd-abcd)

exten => s,n(noim),GotoIf($[${LEN(${DB(CFIM/${ARG1})})} =
4]?im:om)

exten =>
s,n(im),Macro(stdexten-abcd,${DB(CFIM/${ARG1})},SIP/${DB(CFIM/${ARG1})})

exten =>
s,n(om),Dial(SIP/${DB(CFIM/${ARG1})}@SOFTSWITCH_NOVA,40,r)

exten => s,n(om),VoiceMail(${ARG1},u)

exten => s,n(om),hangup()

 

exten =>
s,n(dial),Set(CHANNEL(musicclass)=moh-abcd)

exten => s,n(dial),Set(CALLERID(num)=${CALLERID(num)})

exten => s,n(dial),Dial(${ARG2},12,tT)

exten
=> s,n(dial),Goto(s-${DIALSTATUS},1)

 

CLI lines:

 

  == Using SIP RTP CoS mark 5

    -- Executing [1002 at international-abcd:1]
Macro("SIP/1001-00000006", "stdexten-abcd,1002,SIP1002") in
new stack

    -- Executing [s at macro-stdexten-abcd:1]
Set("SIP/1001-00000006", "DYNAMIC_FEATURES=automon-abcd")
in new stack

    -- Executing [s at macro-stdexten-abcd:2]
GotoIf("SIP/1001-00000006", "0?dnd") in new stack

    -- Executing [s at macro-stdexten-abcd:3]
GotoIf("SIP/1001-00000006", "0?fim:dial") in new stack

    -- Goto (macro-stdexten-abcd,s,20)

    -- Executing [s at macro-stdexten-abcd:20]
Goto("SIP/1001-00000006", "s-,1") in new stack

    -- Goto (macro-stdexten-abcd,s-,1)

[Oct  5 04:37:16] WARNING[5468]: pbx.c:4893
__ast_pbx_run: Don't know what to do with 'SIP/1001-00000006'

 

When I restart the asterisk server; and when I call my
internal extension say from ext: 1001 to ext: 1002, it directly jumps to the 20th
line in the macro which is been highlighted in yellow.

 

The problem gets resolved only when I reload the asterisk
server and then only it jumps to the 17th line in the macro which is
been highlighted in green.

 

Asterisk version: 1.8.4

 

 

Regards,

 

Aamir

-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20111005/6aa211a5/attachment-0002.html>