Hi Nir,
I have the following scenario:
--------------
| gnugk |
| GK |
--------------
/ \
--------------/ -------------
| | | GW |
| * | | H323 |
-------------- --------------
/\
/ \
SIP \
Phone \
SIP
Phone
Because, all my system's authentication and billing procedures are inside my
GK, I need that SIP calls to authenticate through the GK.
I hope this would clear out your doubts.
greetings,
Pablo Salinas
On Thursday 29 April 2004 12:15, Nir Simionovich wrote:
> Hi Pablo,
>
> I'm not really sure if you got an answer for this, but from what I can
> understand, you are trying to
> accomlish 2 issues currently not possible. The chan_h323 channel in
> asterisk doesn't support
> CLID forwarding correctly, so when a call goes out on H323 it will not have
> a CLID. You can forge
> a static CLID, with adding it to the end of h323.conf like this:
>
> [0123456789]
> type=h323
>
> Now, regarding SIP to SIP, I need to understand a bit better, please
> elaborate as it sounds somewhat
> impossible.
>
> Nir S
>
> -----Original Message-----
> From: openh323gk-users-admin@xxxxxxxxxxxxxxxxxxxxx
> [mailto:openh323gk-users-admin@xxxxxxxxxxxxxxxxxxxxx] On Behalf Of pesb
> Sent: Thursday, April 15, 2004 5:28 PM
> To: asterisk-dev@xxxxxxxxxxxxxxxx
> Cc: openh323gk-users@xxxxxxxxxxxxxxxxxxxxx;
> openh323gk-developer@xxxxxxxxxxxxxxxxxxxxx
> Subject: SIP to SIP calls authenticating through a
> GK(gnugk)
>
> Hi there,
> I am using asterisk as an H323 GW through the chan_h323 so
> that SIP terminals can interact with an H323 network managed by gnugk.
> When a SIP phone calls an H323 phone, everything works just fine. Except
> that the caller-id is not sent correctly.
> Now, what I need to do is to stablish a call from SIP to SIP authenticating
> through the H323 Gatekeeper.
> Is this possible?
> I have tried it, and it did not work. These are my config files:
>
> /***************************************************
> h323.conf
> /***************************************************
> [general]
> port = 1720
> bindaddr = 0.0.0.0
>
> disallow=all
> allow=g729
> allow=G723.1
> allow=ulaw ; Allow codecs in order of preference
> allow=alaw
>
> gatekeeper = 192.168.0.103
>
> context=h323
>
> [1005] ; When this line and the context [1004] lines
> are set
> type=h323 ; the caller id 1004 is always sent. I
> don't know why.
> e164=011005 ; In case, this lines are not set, the GS phones
> receives
> context=default ; "Error" as the caller id, and the H323 phone
> receives
> ; "asterisk" as the caller-id
>
> [1004]
> type=h323
> e164=011004
> context=default
>
> [asterisk]
> type=h323
> prefix=01
> context=h323
>
> /**************************************************
> extensions.conf (Just a few lines, the rest is the standar extensions.conf
> file)
> /**************************************************
> ;(...)
> [default]
> ;(...)
> exten => 021005,1,Dial(h323/011005)
> exten => 021004,1,Dial(h323/011004)
> ;exten => _03XXX,1,Dial(h323/${EXTEN:2})
>
> exten => 301,1,Dial(h323/301)
>
> [h323]
> exten => 011005,1,Dial(SIP/1005)
> exten => 011004,1,Dial(SIP/1004)
>
>
> /**************************************************
> sip.conf
> /**************************************************
> [general]
> port = 5060 ; Port to bind to
> bindaddr = 0.0.0.0 ; Address to bind SIP channel to
> context = default ; Default context for incoming calls
>
> disallow=all ; Disallow all codecs
> allow=g729
> allow=G723.1
> allow=ulaw ; Allow codecs in order of preference
> allow=alaw
>
> [1004]
> type=friend
> username=1004
> secret=123
> host=dynamic
> canreinvite=no
>
> [1005]
> type=friend
> username=1005
> secret=123
> host=dynamic
> canreinvite=no
>
> /*****************************************************/
> /*****************************************************/
>
> Also, find attached an ethereal file with the call flow for a call from SIP
> to SIP through my Gatekeeper
>
> My scenario is the following:
> SIP Phones: Grandstream phones
> H323 Phone: Planet IP phone
> Gatekeeper: gnugk 1.0.7
> Asterisk: version 0.9.0
>
> As you can see, asterisk doesn't inform the calling partie that the called
> partie has picked up the phone, so no RTP conexion is stablished between
> asterisk and the calling partie. This happens always, without the matter of
> who is the calling partie(1004 or 1005)
>
> I am really in a hurry here. Please, somebody help me.
>
> Pablo Salinas
>
>
>
>
>
>
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>
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