asterisk users - Jul 2004 - sip over h323

Hi List,

we are using openh323 gatekeeper for voip telefony. We also have a voip 
over ss7 TELES Switch for voip into POSTN Network. Know we want to use 
Asterisk for converting SIP to h323.

Now my question. Is Asterisk an full h323 gatekeeper like openh323? Do 
we need openh323 GK for astrerisk, too?.  And how can i tell asterisk 
to sent all none SIP-ip calls to the gatekeeper over h323?

thx in advanced.

--
Thomas K?pper

01063 Telecom GmbH & Co. KG
Mottmannstr. 2
53842 Troisdorf

Telefon: 02241-9434-506
Telefax: 02241-9434-846

E-Mail: thomas.kuepper@01063telecom.de
E-Mail: tk@teldafax.de
Homepage: http://www.01063telecom.de

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Hi!
> Now my question. Is Asterisk an full h323 gatekeeper like openh323?
No, Asterisk has no gatekeeper functionality, at least not yet.
> And how can i tell asterisk to 
> sent all none SIP-ip calls to the gatekeeper over h323?
One (standard) way to solve this is to place incoming H.323 into their 
own context (e.g. "from-h323"), and incoming SIP calls into another 
context (e.g. "from-sip"). Now define your Dial() statements as
needed.

Philipp

Am 27.07.2004 um 15:37 schrieb Philipp von Klitzing:
> Hi!
>
>> Now my question. Is Asterisk an full h323 gatekeeper like openh323?
>
> No, Asterisk has no gatekeeper functionality, at least not yet.
>
>> And how can i tell asterisk to sent all none SIP-ip calls to the 
>> gatekeeper over h323?
>
> One (standard) way to solve this is to place incoming H.323 into their 
> own context (e.g. "from-h323"), and incoming SIP calls into
another
> context (e.g. "from-sip"). Now define your Dial() statements as 
> needed.
>
ok, thx. what do you meen with context? wherre must i place this?
> Philipp
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
--
Thomas K?pper

01063 Telecom GmbH & Co. KG
Mottmannstr. 2
53842 Troisdorf

Telefon: 02241-9434-506
Telefax: 02241-9434-846

E-Mail: thomas.kuepper@01063telecom.de
E-Mail: tk@teldafax.de
Homepage: http://www.01063telecom.de

---------------------------------------
Diese Nachricht ist vertraulich. Sie ist ausschliesslich fuer den im
Adressfeld ausgewiesenen Adressaten bestimmt. Sollten Sie nicht der
vorgesehene Empfaenger sein, so bitten wir um eine kurze Nachricht. Jede
unbefugte Weiterleitung oder Fertigung einer Kopie ist unzulaessig. Da 
wir
nicht die Echtheit oder Vollstaendigkeit der in dieser Nachricht 
enthaltenen
Informationen garantieren koennen, schliessen wir die rechtliche
Verbindlichkeit der vorstehenden Erklaerungen und Aeusserungen aus. Wir
verweisen in diesem Zusammenhang auch auf die fuer uns geltenden 
Regelungen
ueber die Verbindlichkeit von Willenserklaerungen mit verpflichtendem
Inhalt, die in den bank- bzw. unternehmensueblichen
Unterschriftenverzeichnissen bekannt gemacht werden.
---------------------------------------
This message is confidential and may be privileged. It is intended 
solely
for the named addressee. If you are not the intended recipient please 
inform
us. Any unauthorised dissemination, distribution or copying hereof is
prohibited. As we cannot guarantee the genuineness or completeness of 
the
information contained in this message, the statements set forth above 
are
not legally binding. In connection therewith, we also refer to our 
governing
regulations of concerning signatory authority published in the standard 
bank
or company signature lists with regard to the legally binding effect of
statements made with the intent to obligate us.
---------------------------------------

This will explain the basics of * configuration:

http://www.digium.com/handbook-draft.pdf

After reading this document, you will understand contexts.  

-g




On Tue, 2004-07-27 at 10:53, Thomas Kuepper wrote:
> Am 27.07.2004 um 15:37 schrieb Philipp von Klitzing:
> 
> > Hi!
> >
> >> Now my question. Is Asterisk an full h323 gatekeeper like
openh323?
> >
> > No, Asterisk has no gatekeeper functionality, at least not yet.
> >
> >> And how can i tell asterisk to sent all none SIP-ip calls to the 
> >> gatekeeper over h323?
> >
> > One (standard) way to solve this is to place incoming H.323 into their
> > own context (e.g. "from-h323"), and incoming SIP calls into
another
> > context (e.g. "from-sip"). Now define your Dial() statements
as
> > needed.
> >
> 
> ok, thx. what do you meen with context? wherre must i place this?
> 
> > Philipp
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-users
> >
> >
> --
> Thomas Küpper
> 
> 01063 Telecom GmbH & Co. KG
> Mottmannstr. 2
> 53842 Troisdorf
> 
> Telefon: 02241-9434-506
> Telefax: 02241-9434-846
> 
> E-Mail: thomas.kuepper@01063telecom.de
> E-Mail: tk@teldafax.de
> Homepage: http://www.01063telecom.de
> 
> ---------------------------------------
> Diese Nachricht ist vertraulich. Sie ist ausschliesslich fuer den im
> Adressfeld ausgewiesenen Adressaten bestimmt. Sollten Sie nicht der
> vorgesehene Empfaenger sein, so bitten wir um eine kurze Nachricht. Jede
> unbefugte Weiterleitung oder Fertigung einer Kopie ist unzulaessig. Da 
> wir
> nicht die Echtheit oder Vollstaendigkeit der in dieser Nachricht 
> enthaltenen
> Informationen garantieren koennen, schliessen wir die rechtliche
> Verbindlichkeit der vorstehenden Erklaerungen und Aeusserungen aus. Wir
> verweisen in diesem Zusammenhang auch auf die fuer uns geltenden 
> Regelungen
> ueber die Verbindlichkeit von Willenserklaerungen mit verpflichtendem
> Inhalt, die in den bank- bzw. unternehmensueblichen
> Unterschriftenverzeichnissen bekannt gemacht werden.
> ---------------------------------------
> This message is confidential and may be privileged. It is intended 
> solely
> for the named addressee. If you are not the intended recipient please 
> inform
> us. Any unauthorised dissemination, distribution or copying hereof is
> prohibited. As we cannot guarantee the genuineness or completeness of 
> the
> information contained in this message, the statements set forth above 
> are
> not legally binding. In connection therewith, we also refer to our 
> governing
> regulations of concerning signatory authority published in the standard 
> bank
> or company signature lists with regard to the legally binding effect of
> statements made with the intent to obligate us.
> ---------------------------------------
> 
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

Hallo Thomas!
> ok, thx. what do you meen with context? wherre must i place this?
Du kommst nicht umhin, Dich etwas in Asterisk einzuarbeiten. Ein guter 
Start f?r Dein Problem ist:

http://www.voip-info.org/wiki-Asterisk+config+extensions.conf

Cheers, Philipp

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@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@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 K?pper