Thanks for the response. I have been able to now receive calls over
h.323 using sjphone through the built in ooh323 channel driver. It
seems to work ok for a bit but then asterisk seems to stop accepting
connections and the server needs to be rebooted.
On a slightly different note, I have 3 sip trunks set up in asterisk as
Trunk1, Trunk2, and Trunk3 however asterisk only ever uses trunk3. This
makes it impossible to receive or make more than one call at a time.
An odd situation, but something I need to resolve. Also, when receiving
calls from the stateside client, it seems that the call picks up, and
then hangs up after about 3 seconds of connectivity:
-- Executing GotoIf("OOH323/2128506-5b99", "0?16") in new stack
-- Executing Dial("OOH323/2128506-5b99", "SIP/trunk3/09068601194") in
new stack
-- Called trunk3/09068601194
-- SIP/trunk3-7e99 is making progress passing it to OOH323/2128506-5b99
-- SIP/trunk3-7e99 is ringing
-- SIP/trunk3-7e99 is making progress passing it to OOH323/2128506-5b99
-- SIP/trunk3-7e99 answered OOH323/2128506-5b99
-- Attempting native bridge of OOH323/2128506-5b99 and SIP/trunk3-7e99
= Spawn extension (macro-dialout-trunk, s, 14) exited non-zero on
'OOH323/2128506-5b99' in macro 'dialout-trunk'
== Spawn extension (from-internal, 909068601194, 1) exited non-zero on
'OOH323/2128506-5b99'
-- Executing Macro("OOH323/2128506-5b99", "hangupcall") in new stack
-- Executing ResetCDR("OOH323/2128506-5b99", "w") in new stack
-- Executing NoCDR("OOH323/2128506-5b99", "") in new stack
-- Executing Wait("OOH323/2128506-5b99", "5") in new stack
== Spawn extension (macro-hangupcall, s, 3) exited non-zero on
'OOH323/2128506-5b99' in macro 'hangupcall'
== Spawn extension (from-internal, h, 1) exited non-zero on
'OOH323/2128506-5b99'
Any ideas? It works fine from a soft phone. Could it be something to
do with the codec they are using?
Thanks in advance
Aaron
Cesc wrote:
On 7/24/06, Aaron Anderson <[EMAIL PROTECTED]> wrote:
I have been messing with both all day. I think what might be
tripping me
up is the extensions.conf.
i do think so too :)
I was able to receive an incoming connection from the client, but
all the
system returned was a busy signal. This call was to a known good
number (my
phone) so I'm not sure what's wrong.
if you got the incoming connection but then got a busy tone, it is
because the "peer info" in your asterisk conf is not good ...
try the "standard" demo setup in asterisk ... there you can call
extension 600 (i think) and you get an echo test ...
later, extend this to call whoever you want ... but you'll need to
enter the "location" info somehow (i never did it ... ). I used
sjphone for testing ... try it ... you can add an extension to dial
directly to the IP address of the sjphone. So, you dial the MMM
extension on your phone, get to asterisk and then asterisk dials the
IP address of sjphone ... nice and easy for testing.
Will gnugk do a "translation" from h.323 to sip so I don't have to
make any
major modifications? Is there an example gnugk.ini and h323.conf
file I can
look at to get this all running?
gnugk has no clue of sip.
gnugk is a gatekeeper for h323 ... you can do stuff with the dialled
numbers ... forwarding the call to various [asterisk] h323 gateways
and the like ... but the conversion of h323 to sip is done in
asterisk.
Cesc
Thank you in advance
Aaron
[EMAIL PROTECTED] wrote:
Hello,
Try both chan_oh323 and gnugk .
Harry
--- Aaron Anderson <[EMAIL PROTECTED]> a écrit :
I have been scouring the net the last couple of days
looking for some
kind of tutorial or walkthrough on setting up a
h.323 channel in asterisk.
What I need to do is basically this:
I have a client who wants to be able to connect to
me via h.323 and make
a local phone call (local to me, he is in a
different country). The
call is an automated process and no callee
interaction is required. My
client simply wants to be able to call a user and
give them a
verification number and then hang up. He's using
some in-house software
so unfortunately, h.323 is his only option.
Can someone point me to a doc or perhaps give me a
simple breakdown of
what I need to add to asterisk in order to be able
to do this? I am on
a tight deadline and my searches have not revealed
the information I am
looking for. I have built chan_h323 and it is
loaded but I'm not sure
how to set it up beyond that.
Any help would be much appreciated.
Thank you
Aaron
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