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
|
Title: Mensaje
- [Asterisk-Users] Meetme Admin menu Chee Foong
- Re: [Asterisk-Users] Meetme Admin menu Sergio Serrano Revuelto
- Re: [Asterisk-Users] Meetme Admin menu Brian West