[Asterisk-Users] sip over h323
Thomas Kuepper
tk at teldafax.de
Mon Aug 2 03:14:02 MST 2004
hi list,
i want to convert all none SIP calls to h323 and send them to our GnuGK
Gatekeeper.
with my setup (attached) i called the number 5678 and got the following
error msg:
Error msg:
Jul 29 10:19:45 WARNING[114696]: chan_sip.c:457 __sip_xmit: sip_xmit of
0x81210dc (len 635) to 0.0.22.46 returned -1: Invalid argument
here is my h323.conf:
[general]
port = 1720
bindaddr = 0.0.0.0
allow=all ; turns on all installed codecs
;disallow=g723.1 ; Hm... Proprietary, don't use it...
; User-Input Mode (DTMF)
;
; valid entries are: rfc2833, inband
; default is rfc2833
dtmfmode=rfc2833
;
; Set the gatekeeper
; DISCOVER - Find the Gk address using multicast
; DISABLE - Disable the use of a GK
;217.9.24.23 - The acutal IP address or hostname of your GK
;gatekeeper = DISABLE
;
;
; Tell Asterisk whether or not to accept Gatekeeper
; routed calls or not. Normally this should always
; be set to yes, unless you want to have finer control
; over which users are allowed access to Asterisk.
; Default: YES
;
;AllowGKRouted = yes
; Default context gets used in siutations where you are using
; the GK routed model or no type=user was found. This gives you
; the ability to either play an invalid message or to simply not
; use user authentication at all.
;
context=default
;
; H.323 Alias definitions
;
; Type 'h323' will register aliases to the endpoint
; and Gatekeeper, if there is one.
;
; Example: if someone calls time at your.asterisk.box.com
; Asterisk will send the call to the extension 'time'
; in the context default
;
[default]
type=h323
;
; Keyword's 'prefix' and 'e164' are only make sense when
; used with a gatekeeper. You can specify either a prefix
; or E.164 this endpoint is responsible for terminating.
;
this my sip.conf:
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP channel to
context = sip-phones ; Default context for incoming calls
allow=all ; Allow codecs in order of preference
[1236]
type=friend
username=1236
secret=111
host=dynamic
disallow=all
allow=ulaw
allow=alaw
dtmfmode=rfc2833
nat=yes
context=sip-phones
here extensions.conf:
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel
username/password
;IAXINFO=myuser:mypass
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
;TRUNK=IAX2/user:pass at provider
[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion
[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91NXXNXXXXXX,2,Congestion
[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9NXXXXXX,2,Congestion
[trunktollfree]
;
; Long distance context accessed through trunk interface
;
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91800NXXXXXX,2,Congestion
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,2,Congestion
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,2,Congestion
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,2,Congestion
[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
;
; Master context for long distance
;
ignorepat => 9
include => local
include => trunkld
[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
[sip-phones]
include => sip-endpoints
include => h323-gateway
[sip-endpoints]
exten => 5678,1,Dial(SIP/5678)
exten => 1234,1,Dial(SIP/${EXTEN},60)
exten => 1234,2,Congestion
exten => 1234,102,Busy
exten => 1235,1,Dial(SIP/${EXTEN},60)
exten => 1235,2,Congestion
exten => 1235,102,Busy
exten => 1236,1,Dial(SIP/${EXTEN},60)
exten => 1236,2,Congestion
exten => 1236,102,Busy
[h323-gateway]
exten => _X.,1,Dial(H323/${EXTEN}@217.9.24.23)
My h323 Gatekkeper accepts connections on 217.9.24.23.
any hints for me?
THX
--
Thomas Küpper
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