[asterisk-users] Need help in communicating H323 and SIP
preeta.pandey at wipro.com
preeta.pandey at wipro.com
Fri Feb 8 02:23:06 CST 2008
Hi,
I am trying to communicate H323 and SIP users. I have configured h323.conf, sip.conf and ooh323.conf. If I am using gatekeeper (gnugk) then I am able to call successfully to h323 users using SJphone. And same for SIP users also.
But when I disabled gatekeeper and trying to call using gateway with sjphone then every time whatever number I dial the call goes to asterisk and some computerized information regarding asterisk is coming.
I am putting my h323.conf and ooh323.conf
h323.conf
; The NuFone Network's
; Open H.323 driver configuration
;
listenAddress=10.142.17.68
listenPort=1720
connectPort=1720
;TCP
tcpStart=10000
tcpEnd=20000
;UDP
udpStart=10000
udpEnd=20000
[general]
port = 1720
bindaddr = 0.0.0.0 ; this SHALL contain a single, valid IP address for this machine
;tos=lowdelay
;
; You may specify a global default AMA flag for iaxtel calls. It must be
; one of 'default', 'omit', 'billing', or 'documentation'. These flags
; are used in the generation of call detail records.
;
;amaflags = default
;
; You may specify a default account for Call Detail Records in addition
; to specifying on a per-user basis
;
;accountcode=lss0101
;
; You can fine tune codecs here using "allow" and "disallow" clauses
; with specific codecs. Use "all" to represent all formats.
;
;disallow=all
;allow=all ; turns on all installed codecs
;disallow=g723.1 ; Hm... Proprietary, don't use it...
;allow=gsm ; Always allow GSM, it's cool :)
;allow=ulaw ; see doc/rtp-packetization for framing options
;
; User-Input Mode (DTMF)
;
; valid entries are: rfc2833, inband
; default is rfc2833
;dtmfmode=rfc2833
;
; Default RTP Payload to send RFC2833 DTMF on. This is used to
; interoperate with broken gateways which cannot successfully
; negotiate a RFC2833 payload type in the TerminalCapabilitySet.
;
; You may also specify on either a per-peer or per-user basis below.
;dtmfcodec=101
;
; Set the gatekeeper
; DISCOVER - Find the Gk address using multicast
; DISABLE - Disable the use of a GK
; <IP address> or <Host name> - The acutal IP address or hostname of your GK
gatekeeper = DISABLE
;gatekeeper=10.142.17.68
;
;
; Tell As terisk 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
;
; When the channel works without gatekeeper, there is possible to
; reject calls from anonymous (not listed in users) callers.
; Default is to allow anonymous calls.
;
;AcceptAnonymous = yes
;
; Optionally you can determine a user by Source IP versus its H.323 alias.
; Default behavour is to determine user by H.323 alias.
;
;UserByAlias=no
;
; 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
;
; Use this option to help Cisco (or other) gateways to setup backward voice
; path to pass inband tones to calling user (see, for example,
; http://www.cisco.com/warp/public/788/voip/ringback.html <https://webmail.wipro.com/exchweb/bin/redir.asp?URL=http://www.cisco.com/warp/public/788/voip/ringback.html> )
;
; Add PROGRESS information element to SETUP message sent on outbound calls
; to notify about required backward voice path. Valid values are:
; 0 - don't add PROGRESS information element (default);
; 1 - call is not end-end ISDN, further call progress information can
; possibly be available in-band;
; 3 - origination address is non-ISDN (Cisco accepts this value only);
; 8 - in-band information or an appropriate pattern is now available;
;progress_setup = 3
;
; Add PROGRESS information element (IE) to ALERT message sent on incoming
; calls to notify about required backwared voice path. Valid values are:
; 0 - don't add PROGRESS IE ( default);
; 8 - in-band information or an appropriate pattern is now available;
;progress_alert = 8
;
; Generate PROGRESS message when H.323 audio path has established to create
; backward audio path at other end of a call.
;progress_audio = yes
;
; Specify how to inject non-standard information into H.323 messages. When
; the channel receives messages with tunneled information, it automatically
; enables the same option for all further outgoing messages independedly on
; options has been set by the configuration. This behavior is required, for
; example, for Cisco CallManager when Q.SIG tunneling is enabled for a
; gateway where Asterisk lives.
; The option can be used multiple times, one option per line.
;tunneling=none ; Totally disable tunneling (default)
;tunneling=cisco ; ; Enable Cisco-specific tunneling
;tunneling=qsig ; Enable tunneling via Q.SIG messages
;
;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
; H323 channel. Defaults to "no". An enabled jitterbuffer will
; be used only if the sending side can create and the receiving
&nbs p; ; side can not accept jitter. The H323 channel can accept jitter,
; thus an enabled jitterbuffer on the receive H323 side will only
; be used if the sending side can create jitter and jbforce is
; also set to yes.
; jbforce = no ; Forces the use of a ji tterbuffer on the receive side of a H323
; channel. Defaults to "no".
; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
; resynchronized. Useful to improve the quality of the voice, with
; big jumps in/broken timestamps, usualy sent from exotic devices
; and programs. Defaults to 1000.
; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a H323
; channel. Two implementations are currenlty available - "fixed"
; (with size always equals to jbmax-size) and "adaptive" (with
&nb sp; ; variable size, actually the new jb of IAX2). Defaults to fixed.
; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
;-----------------------------------------------------------------------------------
;
; 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]
; exten => time,1,Answer
; exten => time,2,Playback,current-time
;
; 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 termina ting.
;
; Example: The H.323 alias 'det-gw' will tell the gatekeeper
; to route any call with the prefix 1248 to this alias. Keyword
; e164 is used when you want to specifiy a full telephone
; number. So a call to the number 18102341212 would be
; routed to the H.323 alias 'time'.
;
;[time]
;type=h323
;e164=18102341212
;context=default
;
;[det-gw]
;type=h323
;prefix=1248,1313
;context=detroit
;
;
; Inbound H.323 calls from BillyBob would land in the incoming
; context with a maximum of 4 concurrent incoming calls
;
;
; Note: If keyword 'incominglimit' are omitted Asterisk will not
; enforce any maximum number of concurrent calls.
;
;[BillyBob]
;type=user
;host=192.168.1.1
;context=incoming
;incominglimit=4
;h245Tunneling=no
;
;
; Outbound H.323 call to Larry using SlowStart
;
;[Larry]
;type=peer
;host=192.168.2.1
;fastStart=no
[preeti]
type=peer
c ontext=testing
host=10.142.17.67
[radhamani]
type=peer
context=testing
host=10.142.17.68
ooh323.conf
; Objective System's H323 Configuration example for Asterisk
; ooh323c driver configuration
;
; [general] section defines global parameters
;
; This is followed by profiles which can be of three types - user/peer/friend
; Name of the user profile should match with the h323id of the user device.
; For peer/friend profiles, host ip address must be provided as "dynamic" is
; not supported as of now.
;
; Syntax for specifying a H323 device in extensions.conf is
; For Registered peers/friends profiles:
; OOH323/name where name is the name of the peer/friend profile.
;
; For unregistered H.323 phones:
; OOH323/ip[:port] OR if gk is used OOH323/alias where alias can be any H323
; &nbs p; alias
;
; For dialing into another asterisk peer at a specific exten
; OOH323/exten/peer OR OOH323/exten at ip
;
; Domain name resolution is not yet supported.
;
; When a H.323 user calls into asterisk, his H323ID is matched with the profile
; name and context is determined to route the call
;
; The channel driver will register all global aliases and aliases defined in
; peer profiles with the gatekeeper, if one exists. So, that when someone
; outside our pbx (non-user) calls an extension, gatekeeper will route that
; call to our asterisk box, from where it will be routed as per dial plan.
[general]
;Define the asterisk server h323 endpoint
listenAddress=0.0.0.0
listenPort=1720
connectPort=1720;wich port to sent traffic OUT
;Configure T CP 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
;gatekeeper=10.142.17.67
;The port asterisk should listen for incoming H323 connections.
;Default - 1720
;port=1720
;The IP address, asterisk should listen on for incoming H323
;connections
;Default - 0.0.0.0: tries to find out local ip address on it's own
bindaddr=0.0.0.0
;Alias address for for asterisk server
;Default - "Asterisk PBX"
h323id=ObjSysAsterisk
e164=100
;CallerID for the asterisk originated calls
;Default - Same as h323id
callerid=asterisk
;This parameter indicates whether channel driver should register with
;gatekeeper as a gateway or an endpoint.
;Default - no
;gateway=no
;Whether this asterisk server wi ll use gatekeeper.
;Default - DISABLE
;gatekeeper = DISCOVER
;gatekeeper = a.b.c.d
gatekeeper = DISABLE
;Whether asterisk should use fast-start and tunneling for H323 connections.
;Default - yes
;faststart=yes
;h245tunneling=yes
;Whether media wait for connect for fast start call
;Default - no
;mediawaitforconnect=no
;Location for H323 log file
;Default - /var/log/asterisk/h323_log
;logfile=/var/log/asterisk/h323_log
;Following values apply to all users/peers/friends defined below, unless
;overridden within their client definition
;Sets default context all clients will be placed in.
;Default - default
context=default
;Sets rtptimeout for all clients, unless overridden
;Default - 60 seconds
;rtptimeout=60 ; Terminate call if 60 seconds of no RTP activity
&nb sp; ; when we're not on hold
;Type of Service
;Default - none (lowdelay, thoughput, reliability, mincost, none)
;tos=lowdelay
;amaflags = default
;The account code used by default for all clients.
;accountcode=h3230101
;The codecs to be used for all clients.Only ulaw and gsm supported as of now.
;Default - ulaw
; ONLY ulaw, gsm, g729 and g7231 supported as of now
disallow=all ;Note order of disallow/allow is important.
allow=gsm
allow=ulaw
; dtmf mode to be used by default for all clients. Supports rfc2833, q931keypad
; h245alphanumeric, h245signal.
;Default - rfc 2833
dtmfmode=rfc2833
; User/peer/friend definitions:
; User config options Peer config options
; ------------------ & nbsp; -------------------
; context
; disallow disallow
; allow allow
; accountcode accountcode
; amaflags &nbs p; amaflags
; dtmfmode dtmfmode
; rtptimeout rtptimeout
; ip
; port
; ; h323id
; email
; url
; e164
; &nbs p;
;
;Define users here
;Section header is extension
[myuser1]
type=user
context=context1
disallow=all
allow=gsm
allow=ulaw
[preeti]
type=peer
context=testing
ip=10.142.17.67
port=1720
e164=101
[radhamani]
type=peer
context=testing
ip=10.142.17.67
port=1720
e164=101
[mypeer1]
type=peer
context=context2
ip=10.142.17.67 ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
e164=101
[myfriend1]
type=friend
context=default
ip=10.142.17.67 ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
disallow=all
allow=ulaw
e164=12345
rtptimeout=60
dtmfmode=rfc2833
Please help me understanding the problem and what error I am doing.
Thanking you,
With regards,
Preeta Pandey
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