[Asterisk-Users] OH.323 Dialout Problem
Brian Wilkins
brian at hcc.net
Fri Aug 13 04:46:23 MST 2004
Hi,
I am using the Grandstream HandyTone 486 as a SIP Adapter with a regular
phone. Asterisk configuration is listed below. When I attempt to place a
H.323 call, I receive the following errors:
- Executing Dial("SIP/2000-3029", "OH323/##########@xxx.xxx.xxx.xx:1720|20")
in new stack
Aug 13 09:13:03 WARNING[20497]: channel.c:1806 ast_request: No translator path
exists for channel type OH323 (native 1) to 4
Aug 13 09:13:03 NOTICE[20497]: app_dial.c:705 dial_exec: Unable to create
channel of type 'OH323'
== Everyone is busy at this time
-- Executing Congestion("SIP/2000-3029", "") in new stack
== Spawn extension (default, ##########, 2) exited non-zero on
'SIP/2000-3029'
The Grandstream HandyTone is registered as SIP extension 2000. The Grandstream
HandyTone is configured to use the codec G723 6.3 with 32 frames. Asterisk
version is Asterisk CVS-HEAD-06/21/04-14:23:24. OpenH323 and PWLib are
Compile-time libraries OpenH323 v1.13.5, PWlib v1.6.6. The Asterisk server
also does not appear to register with the gatekeeper, denoted as the ip
address with x's below.
Asterisk configuration follows:
---[oh323.conf]---
;
; Configuration file of OpenH323 channel driver
;
;-----------------------------------------
; General configuration options
; (ports, jitter, GK, ...)
;-----------------------------------------
[general]
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Port to connect to.
; (Used only when we don't have a gatekeeper)
; Default value is 1720.
;
;connectPort=1720
;
; Configure TCP port range to be used by H.323
;
tcpStart=10000
tcpEnd=20000
;
; Configure UDP port range to be used by H.323
; Note: The port range used by RTP are configured from
; "rtp.conf"
;
udpStart=10000
udpEnd=20000
;
; Enable fast start (yes,no).
;
fastStart=no
;
; Enable H.245 tunnelling (yes,no).
;
h245Tunnelling=no
;
; Enable early H.245 messages in call SETUP message.
;
h245inSetup=no
;
; Enable in-band-DTMF detection.
; (Note: Netmeeting uses in-band DTMFs)
;
inBandDTMF=yes
;
; Enable silence suppression.
;
silenceSuppression=no
;
; Set jitter buffer (in milliseconds, 20...10000).
;
jitterMin=20
jitterMax=500
;
; Set IP Type-of-Service byte for RTP channels.
; Valid values for this option are:
; lowdelay, throughput, reliability, mincost, none
;
ipTos=reliability
;
; Set the maximum number of inbound/outbound/simultaneous
; H.323 connections.
;
outboundMax=10
inboundMax=10
simultaneousMax=10
;
; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=1
libTraceLevel=0
libTraceFile=stdout
;
; Disable gatekeeper or specify a gatekeeper.
; Valid values for this option are:
; DISABLE,
; DISCOVER,
; <gatekeeper's DNS name>,
; <gatekeeper's ip>,
; GKID:<gatekeeper's id>
;
;gatekeeper=DISCOVER
gatekeeper=DISCOVER,xxx.xxx.xxx.xx,GKID:HCCGK1
;gatekeeper=xxx.xxx.xxx.xx
;
; Set the gatekeeper password
;
;gatekeeperPassword=HCCGK1
;
; Set the gatekeeper registration timeout
;
gatekeeperTTL=600
;
; Set the mode for sending user-input
; Valid values for this option are:
; Q931 - Q.931 Keypad Information Element
; STRING - H.245 string
; TONE - H.245 tone
; RFC2833 - RFC2833
;
userInputMode=TONE
;
; AMA flags (default, omit, billing, documentation)
;
amaFlags=default
;
; Account code
;
;accountCode=HCCGK1
;
; Set the default context of H.323 calls.
;
context=default
[register]
context=default
alias=asterisk
;-----------------------------------------
; Specify and configure CODEC related
; options
;-----------------------------------------
[codecs]
;
; Define the codec list of the channel driver.
; Every "codec" option may have a "frames" option
; associated with it.
; Valid values for the "codec" option are:
; G711U - G.711 u-Law
; G711A - G.711 A-Law
; G7231 - G.723.1(6.3k)
; G72316K3 - G.723.1(6.3k)
; G72315K3 - G.723.1(5.3k)
; G7231A6K3 - G.723.1A(6.3k)
; G7231A6K3 - G.723.1A(6.3k)
; G726 - G.726(32k)
; G72616K - G.726(16k)
; G72624K - G.726(24k)
; G72632K - G.726(32k)
; G72640K - G.726(40k)
; G728 - G.728
; G729 - G.729
; G729A - G.729A
; G729B - G.729B
; G729AB - G.729AB
; GSM0610 - GSM 0610
; MSGSM - Microsoft GSM Audio Capability
; LPC10 - LPC-10
; Number of frames in RTP packet (if not specified) is 1.
;
;codec=G711A
;frames=20
;codec=G711U
;frames=20
;codec=GSM0610
;frames=4
;codec=G7231
;frames=2
;codec=G729A
;frames=2
;codec=G729B
;frames=2
;codec=G7231
;frames=40
codec=G72316K3
frames=32
---[extensions.conf]---
[general]
static=yes ; These two lines prevent the
writeprotect=yes ; CLI from overwriting the config file
[global]
MAIN_CONTEXT = default
[default]
include => heritage-sip
include => heritage-h323
[heritage-h323]
exten => 9000,1,Dial(OH323/9000)
exten => 9000,2,Hangup
exten => 9001,1,Dial(OH323/9001,20)
exten => 9001,2,Hangup
; Outbound to 1-nxx-nxx-xxxx goes via PSTN
exten => _1NXXNXXXXXX,1,Dial(OH323/${EXTEN},20)
exten => _1NXXNXXXXXX,2,Congestion
; Outbound to nxx-xxxx goes via PSTN
exten => _NXXXXXX,1,Dial(OH323/${EXTEN},20)
exten => _NXXXXXX,2,Congestion
[heritage-sip]
; If the number dialed by the calling party was
; 2000 then dial the user 2000 via the SIP channel
; driver. Let the number ring for 20 seconds, and if
; no answer, proceed to priority 2. If the number
; gives a busy result, then jump to priority 102
exten => 2000,1,Dial(SIP/2000,20)
; Priority 2, send the caller to voicemail
; and gives the "u"navailable of voicemail
; in this instance is to hang up, so
; we have reached the end of our
; priority list
exten => 2000,2,Voicemail(u2000)
;
; If the Dialed number in priority 1 above results in
; a "busy" code, then Dial will jump to 101 + (current priority)
; which in our case will be 101+1=102. This +101 jump is built
; into Asterisk and does not need to be defined.
;
exten => 2000,102,Voicemail(b2000)
exten => 2000,103,Hangup
; Now, what if the number dialed was 2001?
exten => 2001,1,Dial(SIP/2001,20)
exten => 2001,2,Voicemail(u2001)
exten => 2001,102,Voicemail(b2001)
exten => 2001,103,Hangup
exten => 2002,1,Dial(SIP/2002,20)
exten => 2002,2,Voicemail(u2002)
exten => 2002,102,Voicemail(b2002)
exten => 2002,103,Hangup
exten => 2003,1,Dial(SIP/2003,20)
exten => 2003,2,Voicemail(u2003)
exten => 2003,102,Voicemail(b2003)
exten => 2003,103,Hangup
exten => 2004,1,Dial(SIP/2004,20)
exten => 2004,2,Voicemail(u2004)
exten => 2004,102,Voicemail(b2004)
exten => 2004,103,Hangup
;
; Define a way so that users can dial a number to reach
; voicemail. Call the VoicemailMain application with the
; number of the caller already passed as a variable, so
; all the user needs to do is type in the password.
;
exten => 2999,1,VoicemailMain(${CALLERIDNUM})
exten => 2999,2,Hangup
---[sip.conf]---
[general]
port = 5060 ; Port to bind to (SIP is 5060)
bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine)
context = bogon-calls ; Callers we don't know about
[2000]
type=friend ; This device takes and makes calls
username=2000 ; Username for device
;secret=hccnet12 ; Password for device
host=dynamic
context=default
mailbox=100
nat=yes
[2001]
type=friend
username=2001
;secret=hccnet13
host=dynamic
context=default
mailbox=101
[2002]
type=friend
username=2002
;secret=hccnet14
host=dynamic
context=default
mailbox=102
[2003]
type=friend
username=2003
;secret=hccnet15
host=dynamic
context=default
mailbox=103
nat=yes
qualify=yes
[2004]
type=friend ; This device takes and makes calls
username=2004 ; Username for device
secret=hccnet14 ; Password for device
host=dynamic
context=default
mailbox=104
nat=yes
qualify=yes
[9000]
type=friend
username=9000
;secret=hccnet15
host=209.114.205.11
context=default
mailbox=105
[9001]
type=friend
username=9001
;secret=hccnet15
host=209.114.205.11
context=default
mailbox=106
--
--
Heritage Communications Corporation
Melbourne, FL USA 32935
http://www.hcc.net
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