[Serusers] SER forwarding to Asterisk

[Rafael J. Risco G.V.]

Hi 
I am using this condition that works very well for me:

        if(uri=~"^sip:6607.@(my\.domain\.com\.pe|a\.b\.c\.d)"){
                rewritehostport("w.x.y.z:5060");    ## --->asterisk IP:port
                log(1,"n digit expression match - SIP_to_H323 ");      
                route(2);
                break;    
        };

route[2]{       
        log(1,"route[2]:SIP-to-* call routed");
        if(!t_relay()){
                sl_reply_error();
        };

in sip.conf:

[general]
context=default
autocreatepeer=yes
canreinvite=no                 

[mic-inout]
type=friend
secret=******
username=asterisk                 ; Authentication user for outbound proxies
fromuser=asterisk               ; Many SIP providers require this!
host=my.domain.com.pe
dtmfmode=rfc2833
insecure=very  ; needed if we want to allow incoming SER calls to
bypass authentication

...you have to register Asterisk as a gateway in GNUGK, see oh323.conf...

in extensions use exten => _00NXX.,1,Dial,OH323/${EXTEN}

and if you are using oh323 from inaccessnetworks in asterisk try 
*CLI> oh323 debug toggle  ...maybe provide you some useful logs to
solve your problem...

rafael

PS: I recommend this versions:
- PWLIB  :       pwlib_1.5.2.tar.gz
- OpenH323  :       openh323_1.12.2.tar.gz
- Inaccessnetworks-asterisk-oh323  :       asterisk-oh323-0.7.0.tar.gz
(see readme file, there is a note about a patch)
- Asterisk 1.0+
- GNUGK (from www.gnugk.org) versions 2.2+ or 2.0.9, note: radius
billing for sip-to-h323 calls does not work properly with older
versions...


On Sat, 11 Dec 2004 03:38:21 -0800 (PST), Girish <gr_sh2003 at yahoo.com> wrote:
> Joao,
> 
> --- Joao Pereira <joao.pereira at fccn.pt> wrote:
> > Im trying to make SIP to H.323 calls, redirecting from SER to Asterisk and then to GNUGK
> > I tried hard to do it but the documentation is not very explicit.
> <snip>
> 
> >  rewritehostport("Asterisk_ip : Asterisk_port");
> <snip>
> 
> > Can anyone tell me how to redirect H.323 calls to Asterisk?
> 
> The rewritehostport should work for you. Have you installed the h323 channel driver for Asterisk?
> If yes, which one? Experience from people say that Asterisk's native h323 channel driver has
> issues bridging calls between SIP and H323. You need to use the channel driver from
> inaccessnetworks (chan_oh323).
> 
> Here's a caveat if you are installing oh323 driver: Read the documentation carefully before
> installing pwlib and openh323 libraries for the driver. Unless you choose the right versions and
> follow the readme, you'll never be able to install the driver. Just search through the * archives
> to see how many people have faced problems installing the driver.
> 
> Best,
> 
> =====
> Girish Gopinath  <gr_sh2003 at yahoo.com>
> 
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-- 

rrgv