[SIPForum-discussion] line jumping after restarting

aamir chougule aamir_ryu at yahoo.com
Thu Oct 6 14:04:19 UTC 2011

Hi Kias,

Thanks for your valuable suggestion. But the thing is I don't have the hangup line after the 20th Line in the Macro, there are just other lines for say if no answer what to do, if busy what to do...

The problem is whenever I restart the asterisk, the problem arises and jumps directly to the 20th line and to resolve this problem I need to reload the dialplan and then only it starts working.



--- On Thu, 6/10/11, HM Kias <hmkias at gmail.com> wrote:

From: HM Kias <hmkias at gmail.com>
Subject: Re: [SIPForum-discussion] line jumping after restarting
To: "aamir chougule" <aamir_ryu at yahoo.com>
Cc: discussion at sipforum.org
Date: Thursday, 6 October, 2011, 11:48 AM

Is there exten => s,n(dial),hangup() after 20th line, as I feel the value of n is not reset after it hits 17 and hence only after restart it works. 



On Wed, Oct 5, 2011 at 11:27 AM, aamir chougule <aamir_ryu at yahoo.com> wrote:

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

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 =>

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

exten => s,n(noim),GotoIf($[${LEN(${DB(CFIM/${ARG1})})} =

exten =>

exten =>

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

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


exten =>

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

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

=> 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







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