asterisk sip failure to hagup

hi
i just installed asterisk and configured extensions to call out through a sip line.
now the calling out works fine but once the call is finished , and if i attempt to call another number or the same number then the call is not initialised and i get "unrecognised extension on the debug log " and the only way to make it work is restart asterisk.
is there something i am missing on the config .


thanks

Hi

Without seeing what you have done who knows.

it could be the localnet and externip settings

Ian

ianplain :

Hi Without seeing what you have done who knows. it could be the localnet and externip settings Ian

my config is below, all i need is the basic configuration to have all my extensions to dial out through my sip provider



[general]

static=yes
writeprotect=no

[default]
; Wait 10 seconds for an answer (pick up the local phone)
exten => s,1,Wait(10)

; Answer the phone
exten => s,n,Answer

; Playback generic voice mail message
exten => s,n(greeting),Background(vm/generic/greet)

; They must respond within 10 seconds
; else jump to priority t (terminate call)
exten => s,n,Set(TIMEOUT(digit)=3)
exten => s,n,WaitExten(10)

; 1 - Jeff's voicemail
exten => 1,1,Playback(vm/1/unavail)
exten => 1,2,Voicemail(1)
exten => 1,3,Goto(t,1)

; 2 - Tilghman's voicemail
exten => 2,1,Playback(vm/2/unavail)
exten => 2,2,Voicemail(2)
exten => 2,3,Goto(t,1)

; 8 - Voice mail administration
exten => 8,1,VoicemailMain
exten => *,1,Directory(default)
exten => *,2,Goto(s,greeting)
exten => #,1,Playback(vm/generic/goodbye)
exten => #,2,Hangup

; t - terminate call
exten => t,1,Playback(vm/generic/goodbye)
exten => t,2,Hangup

exten => _9.,1,Dial(SIP/${EXTEN:1}@mysipprovider-out,30,r)

[from-mysipprovider]
exten => 1,Answer ; 1234 is the contact extension, default contact extension is "s"
exten => 3,Hangup

exten => _9.,1,Dial(SIP/${EXTEN:1}@from-voip-sip,30,r)

[from-voip-sip]
exten => 1,Answer ; 1234 is the contact extension, default contact extension is "s"
exten => 2,Dial(SIP/600,25,Ttr) ; incoming calls are redirected to SIP telephone with number 111
exten => 3,Hangup


SIP.conf


[600]
type=peer
host=dynamic
secret=600
context=users
mailbox=600@default

[700]
type=peer
host=dynamic
secret=700
context=users
mailbox=700@default

[mysipprovider-out]
type=peer
secret=xxxxx
username=xxxxx
host=sip.voipuser.org
fromuser=mikej
fromdomain=sip.voipuser.org
canrenvite=no
insecure=very
qualify=yes
nat=yes
context=from-mysipprovider ; this section will be defined in extensions.conf

[voip-sip]
type=peer
secret=xxxxxx
username=xxxxx
host=sip.intervoip.com
fromuser=michael
fromdomain=sip.intervoip.com
canreinvite=no
insecure=very
qualify=yes
nat=yes
context=from-voip-sip ; this section will be defined in extensions.conf

Hi as mentioned befor what is the localnet set to and extern ip ?

and perhaps a bit of debug of the call would help as well

Ian