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Asterisk config h323.conf

Created by: oej,Last modification on Fri 08 of Sep, 2006 [08:06 UTC] by folke
Configuration for the Asterisk H323 channels



The NuFone Network's
Open H.323 driver configuration
;
[general]
port = 1720
bindaddr = 0.0.0.0
;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. (See Asterisk IAX channels for explanation of tos and amaflags)

;
;amaflags = default
;

You may specify a default account for Call Detail Records in addition to specifying on a per-user basis. See Asterisk billing

;
;accountcode=lss0101
;

You can fine tune codecs here using "allow" and "disallow" clauses with specific codecs. Use "all" to represent all formats.

;
;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
;
; 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
; <IP address> or <Host name>   - 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@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 terminating.

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' is omitted, Asterisk will not enforce a limit on concurrent calls.

;
;[BillyBob]
;type=user
;host=192.168.1.1
;context=incoming
;incominglimit=4

See also


Back to Asterisk H323 channels

Comments

Comments Filter
222

333CS1000 Hints

by folke, Friday 08 of September, 2006 [06:19:14 UTC]
Note on connecting NuFone H.323 to Nortel CS1000:
  • You'll get Codec-Problems with OOH323 and OH323, So NuFones is your friend.
  • telnet to SignallingServer and do "gkRegTrace ALL" if you need debugging

and:
In channels/h323/ast_h323.cpp you have to replace the single-quote at NuFone's, because this character is invalid for CS1000 (you'll get LOG0003 GKNPM: SOLID SQL Error 1: syntax error (line 1 near 'Network'S') at CS1000 console).

Then use SignallingServer as gatekeeper,
Asterisk's endpoint name configured at CS1000-NRS has to be used as H.323 alias in h323.conf, not the name of the CS1000!
Then you have to setup an CDP entry (type DSC) in overlay 87 with your prefix pointing to the RLB (overlay 86) which points to the route with the H323_IP_TRUNK's (but this isn't asterisk specific...)

222

333Lost and confused

by xpedition, Thursday 08 of June, 2006 [03:14:58 UTC]
we want to use * as a back end on a meridian cs1000. DID from the CS1000 to the *box with H.323 from the CS1000 and then sip and IAX off the * box. i have *1.2.7 and ooh323 installed and can ring a sip ext from netmeeting as in the only config docs i can find. we can also make and recieve calls from a sjphone (h.323) and a sip wifi phone. so we know the * box is configured to make internal calls. how do we set up trunking to the meridian cs1000? and which *.confs need to be changed. any input will be helpful.
lawrence.zablocki@xpedition.us
222

333help me about H323 and SIP communicating

by sodier, Wednesday 12 of April, 2006 [16:36:15 UTC]
i add the followings into H323.conf,but SJphone users cannot be regestered on the server with asterisk,that is ,the server cannot recognize the h323 user
600
type=friend
host=59.78.36.76
context=internal

while in the extensions.conf ,I add the followings:
internal
include => demo
exten => _6XX,1,Dial(H323/&{EXTEN})
exten => _6XX,3,Hangup()

anyone can help me?
what I want to do is to communicate between sip and h323 endpoints which both registered on the same server with asterisk.
thank u!

johnylhy@hotmail.com
222

333Using Cisco ATA with h323 firmware

by razametal, Wednesday 06 of April, 2005 [19:50:33 UTC]
I want to connect my cisco ata 186 with h323 firmware to asterisk, how can i do it ? please show a config or links to investigate it.

Regards
222

333How too H323 gtwy 2 gtwy

by mfmaduro, Monday 28 of March, 2005 [13:55:53 UTC]
(:cry:) I install Asterisk this weekend on my GENTOO box and she is up and running and she is off the hook yo all. I 'm using my deltha 3 acc for my calls but wanted to have her talk to my Cisco 3660 which is a IP2IP gateway how can i do this can any body help.

Michel@voipglobal.org
222

333Re: had success with Avaya multivantage

by , Tuesday 18 of January, 2005 [05:27:31 UTC]
Please help me.
I'm have communicate AVAYA Definity with Asterisk.
And I cannot make it.

andrews@mtelecom.chita.ru
222

333had success with Avaya multivantage

by meowmeow64, Tuesday 10 of August, 2004 [18:47:14 UTC]
used codec was g711
and dtmf mode as inband.
set up asterisk as ivr.
worked like a charm.

jonl@safi-sys.com