Hi,<br>
I have a little problem related to quintum a400 gateway. <br>
I have installed asterisk-1.2.8. Have configured it with SIP and H323
channels to recieve and make calls over lan using softphone (shphone
for both SIP and H323). H323 driver version is openh323-v1.17.1 and
pwlib-v1.9.0 . pc to pc calls thru asterisk are established without any
problem. <br>
Recently i connected a quintum a400 gateway to the lan. quintum is
programmed like whenever it recieves a call request, it should forward
the request to asterisk server, and it does with no error. after call
being forwarded to asterisk, asterisk uses 's' extensions to handle the
call. initially i am using the following extension:<br>
<br>
exten=>s,1,Dial(H323/192.168.0.23,20) ;23 is the ip address of pc using softphone.<br>
<br>
a digital phone (simple one which we use for direct pstn comm) is
connected with the 1st pbx port of quintum. we dial quintum extension,
quintum(using h323) forwards the call to asterisk, asterisk dials the
ip .23, softphone rings, as we answere the phone the call gets
disconnected atuomatically. SIP account ends up with the same result.
here is the log info:<br>
<br>
<span style="font-weight: bold; text-decoration: underline;">H323 LOG</span><br>
== Starting H323/ip$192.168.0.22:24602/21 at default,15,1 failed so falling back to exten 's'<br>
-- Executing Dial("H323/ip$192.168.0.22:24602/21", "H323/192.168.0.23/20") in new stack<br>
-- Called <a href="http://192.168.0.23/20">192.168.0.23/20</a><br>
Jul 4 16:17:23 WARNING[2955]: channel.c:2693
ast_channel_make_compatible: No path to translate from
H323/192.168.0.23-2(-2033656) to H323/ip$192.168.0.22:24602/21(-2033656)<br>
-- H323/192.168.0.23-2 is ringing<br>
-- H323/192.168.0.23-2 is ringing<br>
-- H323/192.168.0.23-2 answered H323/ip$192.168.0.22:24602/21<br>
== Spawn extension (default, s, 1) exited non-zero on 'H323/ip$192.168.0.22:24602/21'<br>
<br>
I ALSO DONT KNOW THE REASON WHAT THIS WARNING IS ABOUT<br>
<br>
<span style="font-weight: bold; text-decoration: underline;">SIP LOG<br>
</span>the same thing happens for sip account but without the warning.<br>
<br>
<span style="font-weight: bold; text-decoration: underline;"><span style="text-decoration: underline;"><span style="font-weight: bold;">H323 </span></span>Channel Configuration</span><br>
[laptopAsus]<br>
type=friend<br>
host=<a href="http://192.168.0.23">192.168.0.23</a><br>
context=default<br>
<span style="font-weight: bold; text-decoration: underline;"><span style="text-decoration: underline;"><span style="font-weight: bold;"><br>
SIP Channel Configuration</span></span></span><br>
[Ammad]<br>
type=friend<br>
secret=tu<br>
qualify=4000 <br>
nat=yes <br>
host=dynamic <br>
canreinvite=no <br>
context=default <span style="font-weight: bold; text-decoration: underline;"><span style="text-decoration: underline;"><span style="font-weight: bold;"><br>
</span></span><br>
</span>I have no idea how to solve this problem. already tried to use different codecs but no progress......plz help<br>