After Call Forward and all lines congestion

Hi All,

I have 2 PSTN line (35830001, 35830002) at my asterisk server and connected with 5 SIP phone.
All incoming calls rings at SIP001 only.

When I set a call forward function at my SIP001 to my cell phone number.
Then I test from SIP002 to SIP001 and all goes fine.

My problem is followings:

When I dial 35830001 (my staff’s cellphone), then asterisk will use 35830002 to call my cell phone, it works fine.
But after my cellphone or my staff’s cellphone hangup. All the PSTN line of asterisk become CONGESTION.
I have test for arround ten times, but everytime after Hangup. It cause the problem and need to restart the service of asterisk.

Here is my dialplan.


; OpenVox Channel 1,2
exten => 35830001,1,Goto(default,s,1)
exten => 35800002,1,Goto(default,s,1)
exten => s,1,Goto(default,s,1)
exten => t,1,Goto(default,s,1)
exten => i,1,Goto(default,s,1)

include => parkedcalls
exten => s,1,Wait(0.5)
exten => s,n,Dial(SIP/1001)
exten => s,n,Hangup()

; Office extension 1001 ~ 1005
exten => _100[1-5],1,Macro(dialexten,${EXTEN})

; Access VoiceMain at own device
exten => 800,1,Answer
exten => 800,2,VoiceMailMain(${CALLERID(num)},s)

; Direct Call Pickup
exten => _*8.,1,PickUpChan(SIP/${EXTEN:2})

exten => 911,1,Macro(dialout,911)

include => default
include => wtntcfwd
include => emergency
exten => _NXXXXXXX,1,Macro(dialout,${EXTEN})

include => local
exten => _00.,1,Macro(dialout,${EXTEN:2})

; Dialout Macro
; ${ARG1} - Exten Number
exten => s,1,Set(PATH=/var/spool/asterisk/monitor)
exten => s,n,Set(FILENAME=${UNIQUEID})
exten => s,n,Set(MONITOR_EXEC=/etc/asterisk/scripts/wavtomp3 ${PATH} ${FILENAME})
exten => s,n,Dial(DAHDI/g0/${ARG1})
exten => s,n,Congestion(5)
exten => s,n,Hangup()

; Dial SIP Extension Macro
; ${ARG1} - Exten Number
exten => s,1,Dial(SIP/${ARG1},20)
exten => s,n,Goto(s-${DIALSTATUS},1)
exten => s-NOANSWER,1,VoiceMail(${ARG1}@default,u)
exten => s-NOANSWER,n,Hangup()
exten => s-BUSY,1,VoiceMail(${ARG1}@default,b)
exten => s-BUSY,n,Hangup()
exten => s-CHANUNAVAIL,1,VoiceMail(${ARG1}@default,u)
exten => s-CHANUNAVAIL,n,Hangup()
;exten => s-.,Goto(s-NOANSWER,1)
exten => a,1,Hangup()

; Dial SIP Extension without VoiceMail
; ${ARG1} - Exten Number
exten => s,1,Dail(SIP/${ARG1},30)
exten => s,n,Hangup()[/color]

This sounds like some other issues that have been posted before. I don’t suspsect the problem is going to be your dialplan.

Its probably has to do with the configuration of how the PSTN lines are connected to the Asterisk box.

What is the connection? Hardware? Post the config for that hardware.