[Asterisk-Users] SIP stage

[Sergio Serrano Revuelto]

Hi, 
I would like to configure a stage for SIP phones. This stage would be
the next:
 
two netergy SIP phones connected to Asterisk through chan_sip. 
one X100P or AVM FRITZ to outside lines.
 
I think that sip.conf would be the next:
 
;
; SIP Configuration for Asterisk
;
[general]
port = 5060                     ; Port to bind to
bindaddr = 192.168.0.207                ; Address to bind to
context = outgoing              ; Default for incoming calls
disallow=all
allow=alaw
maxexpirey=3600         ; Max length of incoming registration we allow
defaultexpirey=120              ; Default length of incoming/outoing
registration

[704]
type=friend
username=704
;secret=704
host=192.168.0.154
dtmfmode=rfc2833
mailbox=704
callerid=704
context=outgoing
reinvite=yes
canreinvite=yes
qualify=yes
nat=-1
 
[705]
type=friend
username=705
;secret=705
host=192.168.0.155
;defaultip=192.168.0.5
dtmfmode=rfc2833
mailbox=705
callerid=705
context=outgoing
reinvite=yes
canreinvite=yes
qualify=yes
nat=-1

And my extensions.conf would be the next:
 
[outgoing]
 
exten=>i,1,Playback(invalid)
exten=>t,1,Hungup()
 
exten=>_7XX,1,Goto(SIP|${EXTEN}|1)
exten=>_XXXXXXXXX,1,ChanIsAvail(CAPI/951014943&CAPI/951014944)
exten=>_XXXXXXXXX,2,SubString,CANAL=${AVAILCHAN}|12|9
exten=>_XXXXXXXXX,3,Dial(CAPI/@${CANAL}:B${EXTEN}|17)

[SIP]
exten=>704,1,Dial(SIP/704|tTm)
exten=>705,1,Dial(SIP/705|tTm)
 
 
are these files correct?
 
 
Why hwen I try call from one phone to other only rings once and then
hungup?
 
 
 
Any idea,
thanks,

 srsergio

 

 

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