Hi,
As explained in my original post on June 30. When I used CVS 2004-06-16 with
oh323-0.6.3a. I can compile and install without problem but when I am in
the asterisk console whenever I issue stop now or restart now or
extension reload I got stuck on the console and asterisk did not response
to
I too tried 0.6.3 and it is behaving the same. I have now downloaded oh323
to 0.6.2a and it seems fine.
Regards,
Anthony
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Hi,
Here it is when I start it with /etc/rc.d/init.d/asteriskd found in asterisk
source contrib/init.d/rc.redhat.asterisk
It started without problem and when i issue stop now It freezes, please
see below,
tai*CLI
add debug dontdumpextensions
Hi,
I am using asterisk CVS 2004-06-16 with oh323-0.6.3a
I have a strange problem if I start asterisk with oh323 loaded
/usr/sbin/asterisk -vc
once I am in the console and issue restart now or reload asterisk hangs
and it not stoping or restarting at all, below is the console logging when
According to voip-info.org,
3 way calling: Normally implemented by the phone
I am using a Grand Stream 100 and not able to make this work. I can dial out
to 1st number then with the flash button I am able to dial out again to a
2nd number. I am not able to bind them together into 1 conversation.
Hi,
I am having problem compiling radius support for asterisk and am wondering
if anyone would point out whats wrong, here is what I get when complie
cc -DOPENSSL_NO_KRB5 -fPIC -c -o res_radius.o res_radius.c
res_radius.c: In function `RDial_exec':
res_radius.c:1047: parse error before
Hi all,
I am able to track incoming h323 calls with phone number by using
amaFlags=billing or amaFlags=documentation. But is it possible to tracking
the incoming IP at the same time?
If I would like to restrict incoming h323 access to certain IP, should it be
done on asterisk or oh323 level?
Hi,
I am having the same issue and am wondering if the problem is with the sip
phone or oh323.
I have upgraded my oh323 to 0.5.10 since Michael had mentioned that it was
with the answer supervision but I am still encountering the same issue. Any
help is appreciated.
-- Called [EMAIL
I just purchased a G729 codec yesterday and am having the same issue. Does
anyone have a solution to this? Thanks for your input.
Regards,
Anthony
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Hi,
What I mean is when I start asterisk with -vvvc. I got
[codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec Translator)
== Detected 1 licensed G.729 transcoders
Mar 23 13:46:12 WARNING[1024]: translate.c:219 calc_cost: Translator
'g729tolinb' does not produce sample frames.
==
Hi,
exten = _1613XXX,1,Dial,OH323/[EMAIL PROTECTED]
This should work. Provide a more detailed Asterisk log to trace the
problem.
Sorry. It was my mistake. I believe it works now
Could * at the same time take h323 packets from let say a cisco gateway and
pass it through another softswitch?
Hi
I have tired using the below syntax but could't go through, I wonder if my
syntax is wrong. Please kindly comment.
Btw I am using asterisk-0.7.1 with oh323-0.5.7
exten = _1613XXX,1,Dial,OH323/[EMAIL PROTECTED]
Error Below
Mar 12 09:30:01 WARNING[5126]: chan_sip.c:2365
HI,
I have successfully configured asterisk to accept SIP session from sip
phones and use oh323 to forward calls to our gateway using H323 and
eventually PSTN termination. But since some of the gateways are not in our
control, we need to send tech-prefix + phone number to third parties. I am
Sorry for my ignorance but what is the difference between using the G.729
codec and using G.729 pass thru. In my scenario below does it consider to be
using the G.729 or using it as pass through?? Do I still need licence for
the G.729?
SIP(if using g.729) ---asterisk-h323
Thanks very much Michael.
It worked but only if I configure my cisco to use g711alaw.
If I config my cisco to use default g729r8 it created the below
Feb 9 15:37:59 WARNING[32788]: channel.c:1856 ast_channel_make_compatible:
No path to translate from H323:9242(256) to H323:28967(8)
Feb 9
Thanks very much Michael.
It worked but only if I configure my cisco to use g711alaw.
If I config my cisco to use default g729r8 it created the below
Feb 9 15:37:59 WARNING[32788]: channel.c:1856 ast_channel_make_compatible:
No path to translate from H323:9242(256) to H323:28967(8)
Feb 9
connections to it from 192.168.1.2
Tomica
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anthony Law
Sent: Thursday, February 05, 2004 10:41 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] question for oh323 users
Hi,
I am trying to forward calls
Hi Gus,
Thanks for your reply. I have tried below and still didn't work.
exten = _1905XXX,1,Dial,OH323/h323:[EMAIL PROTECTED]
or
exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED]
and now asterisk gives out below error
Feb 6 16:12:41 WARNING[30740]: pbx.c:1773 ast_pbx_run: Channel
Hi,
I am trying to forward calls from one cisco gateway to another cisco gateway
using asterisk
cisco(5300)A 192.168.1.1
asterisk 192.168.1.2
cisco(5300)B 192.168.1.3
pstn --ciscoA-asterisk --ciscoB--pstn
I have the below in my extension.conf
[default]
exten =
Hi,
Please excuse me if my question seems too simplistic. I have been reading
the mailing list for some time and I am still a bit confused. Here is the
scenario that I would need to achieve and am wondering if asterisk is the
correct software to use.
(h323) (h323/SIP)
Dear all,
Is Asterisk able to function straightly as a Voip transit softswitch (class
4) ?
Regards,
Anthony
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