When I use Asterisk1.4, could not compile oh323 with it, pwlib and openh323 are successfully compiled. Did somebody try that successfully.

Give some tips.

Thanks

Is there any way of sending an incoming call to an specific context depending on the ip which it came in?
This could be done with h323 module, isn't oh323 capable of doing this?

I just installed ooh323 0.73 in my asterisk box. oh323 commands can be found in asterisk CLI. But no calls can get through. Any one's help?

First problem, oh323.conf in etc/asterisk/ cannot be loaded with phpconfig. I need to restart amportal all the time. Does it mean I missed something during installation? What should I do?

Then my situation, what I need to do is:

SIP phones--->asterisk------>H323 soft switch for termination

The provider told me I can treat the soft switch as a gateway, gave me the IP, prefix and nothing else.

Then my oh323.conf

;
; Configuration file of OpenH323 channel driver
;

;-----------------------------------------
; General configuration options
; (ports, jitter, GK, ...)
;-----------------------------------------
general
;
; Address to bind to for incoming connections.
; Default is ALL.
;
listenAddress=0.0.0.0
;
; Port to listen to.
; Default value is 1720.
;
listenPort=1720
;
; Configure the TCP port range to be used by H.323
;
tcpStart=10000
tcpEnd=20000
;
; Configure the UDP port range to be used by H.323

Note

The port range used by RTP are configured from

; "rtp.conf"
;
udpStart=10000
udpEnd=20000
;
; Enable fast start (yes,no).
;
;fastStart=yes
fastStart=no
;
; Enable H.245 tunnelling (yes,no).
;
;h245Tunnelling=yes
h245Tunnelling=no

;
; Enable early H.245 messages in call SETUP message.
;
;h245inSetup=yes
h425inSetup=no
;
; Set jitter buffer (in milliseconds, 20...10000).
;
jitterMin=20
jitterMax=100
;

ipTos=none


outboundMax=100
inboundMax=100
simultaneousMax=100


; Set the bandwidth limit for H.323 connections.
; The value is in Kbps.
;
;bandwidthLimit=1024
;
; Set tracing options for the wrapper library and for the
; OpenH323 library.
; libTraceFile can be 'stdout' or a full path name to the tracefile.
; Only the trace info for OpenH323 is logged in libTraceFile.
;
wrapLibTraceLevel=3
libTraceLevel=3
libTraceFile=stdout
;

gatekeeper=DISABLE


gatekeeperTTL=600
;
; Set the mode for sending user-input (DTMF)

Valid values for this option are

; Q931 - Q.931 Keypad Information Element
; STRING - H.245 string
; TONE - H.245 tone
; RFC2833 - RFC2833
; INBAND -
;
userInputMode=TONE

amaFlags=default

accountCode=H323

language=en

musiconhold=default

context=voip-h323

;-----------------------------------------
; Configure H.323 aliases, prefixes and
; related ASTERISK's contexts
;-----------------------------------------
register
;
; Aliases/prefixes associated with the default context
; defined in section general.
;
;alias=asterisk
;alias=123
;
; Aliases/prefixes routed in "all-aliases" context.
;
context=all-aliases
;alias=ASTERISK
;alias=666
;
; Aliases/prefixes routed in "more-aliases" context.
;
;context=more-aliases
;alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
;context=all-prefixes
;gwprefix=00
;gwprefix=01
;
; Aliases/prefixes routed in "more-stuff" context.
;
;context=more-stuff
;alias=664
;gwprefix=02

codecs
codec=G711A
frames=20

Then I add a line to extensions.conf from-internal for testing

exten => _9999X.,1,Dial(OH323/${EXTEN}@xxx.xxx.xxx.xxx:1720)

Then the debug:

   — Executing Dial("SIP/9091-094537a8", "OH323/99998xxxxxx@xxx.xxx.xxx.xxx:1720") in new stack
   — H.323 call to 99998xxxxxx@xxx.xxx.xxx.xxx:1720 with codec(s) alaw
   — Outbound H.323 call to destination 99998xxxxxx@xxx.xxx.xxx.xxx:1720', channel 'OH323/99998xxxxxx@xxx.xxx.xxx.xxx:1720-3ee0f73'.
   — Called 99998xxxxxx@xxx.xxx.xxx.xxx:1720

Then nothing happened. Even with a reason 0, 5,8,...until I hungup

   — Hungup 'OH323/99998xxxxxx@xxx.xxx.xxx.xxx:1720'
 == Spawn extension (from-internal, 99998xxxxxx, 1) exited non-zero on 'SIP/9091-094537a8'

Anyone got the correct oh323.conf to be able to link my asterisk to send calls to my Quintum DX gateway?

im trying to let my sip asterisk users to make calles from the asterisk and route the calls to the quintum that will originate the call

Any Help???

Rushan@mec.com.jo

Anyone got the correct oh323.conf to be able to link my asterisk to send calls to my Quintum DX gateway?

im trying to let my sip asterisk users to make calles from the asterisk and route the calls to the quintum that will originate the call

Any Help???

Rushan@mec.com.jo

it only installs 3 files:
$(DESTDIR)/etc/asterisk/oh323.conf
$(DESTDIR)/usr/local/lib/liboh323wrap_s.a
$(DESTDIR)/usr/lib/asterisk/modules/chan_oh323.so

I made a patch for my custom settings that included the following:

---

+++../Makefile 2005-02-01 16:20:02.142037782 -0800 @@ -38,9 +38,9 @@
#
# Install everything under this directory. If this is empty, then everything
# will be installed under /
#
-DESTDIR=
+DESTDIR=/opt/asterisk-oh323-0.6.5  
#
# Set PWLIBDIR variable to the directory containing the sources of
# PWlib library.

---

But then I had to add these symbolic links for Asterisk.
lrwxrwxrwx 1 root root 49 Feb 1 16:41 /etc/asterisk/oh323.conf -> /opt/asterisk-oh323-0.6.5/etc/asterisk/oh323.conf
lrwxrwxrwx 1 root root 64 Feb 1 16:43 /usr/lib/asterisk/modules/chan_oh323.so -> /opt/asterisk-oh323-0.6.5/usr/lib/asterisk/modules/chan_oh323.so

note, it doesn't seem to need a link to liboh323wrap_s.a

I have had problems with asterisk-oh323. I've tryed to uninstall it but it seems like some traces of the module persist on asterisk and creates problems with chan_h323.

Anyone knows how to get rid of oh323?