Hi all,
After the work of around 24 hours totally, I finally make the H323 to SIP
to work finally, what the .........
First of all, I missed out the port range of H323 and asterisk. I set
50000-59999 for RTP in GNUGK, but I tell asterisk to go for 10000-20000,
which seems make problem, but I didn't try that, because once I erased the
h323 channel driver (bundled one) and re-install the oh_323 from
inaccessnetwork, i correct the rtp port setting for both rtp.conf and
oh323.conf.
Secondly, SIP do really have better NAT transval. It does not need any
changes of router or router setting, but it works very kool. So I have my
asterisk as SIP registar.
In oh323.conf, make it listen to local lan address (not just 127.0.0.1 is
ok, but becareful of any crash with other application such as T38modem. This
problem i will have to fix it, by changing the port of T38modem. Hope I can
make it works tonight.
As mentioned by Assen, codec problem is really a big problem. I got core
dump in codec conversaion. For me i fix everything to G711A as it works well
for me and really i didn't care too much about the bandwidth.
Cheers!
Yours,
Jason Chan, Hong Kong
----- Original Message -----
From: "Bulgaria Online - Assen Totin" <assen@xxxxxxxxx>
To: "openh323gk-users-request@xxxxxxxxxxxxxxxxxxxxx"
<openh323gk-users@xxxxxxxxxxxxxxxxxxxxx>
Sent: Wednesday, February 02, 2005 11:31 PM
Subject: Re: Re: H323 to SIP
> Hi all,
>
> I've been experimenting a bit with Asterisk/GnuGK interconnect for
> several months, but without significant success.
>
> Here are just a few observations (to know where you stand):
>
> 1. Both h323 driver (Asterisk native) and oh-323 only compile
> against Janus-Patch4 versions of Pwlib and OpenH323Lib. I've been
> running both GnuGK 2.0.8 and 2.2.0 with Pandora release, no problem,
> but the Asterisk channel drivers are stubborn enough (platforms
> tested: Fedora core2 and core3, i.e. gcc 3.3 and 3.4).
>
> 2. I'm not aware that oh-323 is better; whenever I tried to use the
> "stable" version, when the channel tried to "dial" to GnuGK, I got
> SIGSEGV. With the CVS this does not happen, however, you should be
> prepared to face another serious problem: codecs. In fact, Asterisk
> receives the SIP call through the SIP channel and then "dials" the
> H323 channel to the gatekeeper. Apparently, Asterisk tries to
> negotiate an appropriate codec with both the SIP phone and with the
endpoint
> that the H323 GK has routed the call to ("direct" endpoint-to-endpoint
> connection through Asterisk via the "reinvite" mechanism can only be
> done when making a SIP-to-SIP call). If all your endpoints speak G.711
> and you are happy with the bandwidth usage, it works fine.
> Unfortunately to me, most commercial endpoints prefer G.729, which is
> available from Digium, but not for free and through a weird
> registration procedure (their registration code is bound to the MAC
> addresses of all NICs on the machine and if you replace a NIC, you
> have to re-register, which can only be done once). Other problems with
> the Asterisk/GnuGK interconnect are that calls do not hang up
> properly (I suspect Asterisk's fault here), which results in billing
> issues (Asterisk has almost no billing capabilities; it doesn't even
> speak RADIUS except through a highly-specialized patch for
> prepaid-card-type only services; this patch is nsuitable for SIP/H323
billing).
>
> 3. I haven't played much with Vovida VOCAL - just know that it
> compiles cleanly, but be warned - it is a _huge_ projects consisting
> of several interconnected services and a normal setup requires a set
> of several *nix boxes to run on.
>
> Just my 2c.
>
> WWell,
>
> Assen Totin
> Development Manager
>
> ===============================
> BULGARIA ONLINE
> Your quality... Your price!
> ===============================
> tel. (+359 2) 973-3000 ext. 511
> http://home.online.bg
>
>
>
>
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>
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