[Asterisk-Users] Quintum A400 Call Establishment Prob

Rizwan Hisham rizwanhasham at gmail.com
Tue Jul 4 06:05:06 MST 2006


Hi,
I have a little problem related to quintum a400 gateway.
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.
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:

exten=>s,1,Dial(H323/192.168.0.23,20) ;23 is the ip address of pc using
softphone.

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:

H323 LOG
 == Starting H323/ip$192.168.0.22:24602/21 at default,15,1 failed so falling
back to exten 's'
    -- Executing Dial("H323/ip$192.168.0.22:24602/21",
"H323/192.168.0.23/20") in new stack
    -- Called 192.168.0.23/20
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)
    -- H323/192.168.0.23-2 is ringing
    -- H323/192.168.0.23-2 is ringing
    -- H323/192.168.0.23-2 answered H323/ip$192.168.0.22:24602/21
  == Spawn extension (default, s, 1) exited non-zero on
'H323/ip$192.168.0.22:24602/21'

I ALSO DONT KNOW THE REASON WHAT THIS WARNING IS ABOUT

SIP LOG
the same thing happens for sip account but without the warning.

H323 Channel Configuration
[laptopAsus]
type=friend
host=192.168.0.23
context=default

SIP Channel Configuration
[Ammad]
type=friend
secret=tu
qualify=4000
nat=yes
host=dynamic
canreinvite=no
context=default

I have no idea how to solve this problem. already tried to use different
codecs but no progress......plz help
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