Connecting Analog to H323 gateway to asterisk server

Hi all

I am recently configuring an asterisk server to work as a local PBX. I have some old 8 ports Analog to H323 Elite gateways and I want to use them to connect all the office lines to my asterisk server. I have connected these gateways to my ethernet network.

Actually I have successfully done some experiments calling analog phones connected on the same gateway and also inter-gateway calls. And actually I want to connect these gateways to my asterisk server so I can pass calls from SIP phones to the analog phones on the H323 gateways but I can’t find any tutorial to guide me.

Now I don’t know what to configure on the asterisk server to be able to forward calls concerning configured extensions to the voip gateway.

Thanks to all

Hi all,

I am replying to myself to update this topic with the last thing I have reached maybe someone can help me further.

Actually this is what I have succeeded to do :

Xlite(6000) -> Asterisk -> H323 Gateway -> Analog phone (550)

this is what I have set in my *Now extensions.conf :
[defaults]
exten => 6050,1,VoiceMailMain
exten = 7000,1,Goto(voicemenu-custom-1|s|1)
exten => 550,1,Dial(H323/550@192.168.39.5,10)

actually this makes the analog phone ring and then the call is finished and CLI shows me :
– Executing [550@my-dialplan:1] Dial(“SIP/6001-081e5008”, “H323/550@192.168.39.5|10”) in new stack
– Requested transfer capability: 0x00 - SPEECH
– Called 550@192.168.39.5
– H323/192.168.39.5-3 answered SIP/6001-081e5008
== Spawn extension (my-dialplan, 550, 1) exited non-zero on ‘SIP/6001-081e5008’

Have someone an idea why does the call get disconnected after the first ring ?

I will update the status

Thanks to all

Hi all,

still replying on my self.

Actually I have succeeded to pass a call from Xlite(SIP) -> Asterisk -> H323 Gateway -> Analog phone. The voice is passing through the call. A little annoying thing is that when I pick up the analog phone I hear beeps as if the number was composed on the keypad.

The main problem is that when I am calling the inverse way, my xlite client rings and when I pick up the softphone, asterisk crashes.

Here is my extensions.conf :
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no

[globals]
TRUNKMSD=1

[default]
exten => 1000,1,Dial(SIP/1000)
exten => 2000,1,Dial(SIP/2000)

exten => _5XX,1,Dial(H323/${EXTEN}@Gateway5); H323 Gateway Extension

my h323.conf :
[general]
port = 1720
bindaddr = 192.168.39.2
disallow=all
allow=alaw
dtmfmode=inband
gatekeeper = DISABLE
context=default
progress_setup = 8
progress_alert = 8
h245tunneling=yes

[Gateway5]
type=friend
context=default
host=192.168.39.5
port=1720
disallow=all
allow=alaw,g729,gsm,slinear

My sip.conf :

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
recordhistory=yes
dumphistory=yes

[1000]
type=friend
regexten=1000
context=default
secret=1000
username=1000
callerid=“User1” <1000>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
dtmfmode=inband

[2000]
type=friend
regexten=2000
context=default
secret=2000
username=2000
callerid=“User2” <2000>
host=dynamic
nat=yes
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
dtmfmode=inband

I have tried to debug the core dump and here is the stack trace :

#0 ast_rtp_new_source (rtp=0x0) at rtp.c:2004
#1 0x077c7c2e in oh323_indicate (c=0xa074ee0, condition=20, data=0x0, datalen=0) at chan_h323.c:919
#2 0x08082902 in ast_indicate_data (chan=0xa074ee0, condition=20, data=0x0, datalen=0) at channel.c:2348
#3 0x08087354 in ast_channel_bridge (c0=0xa074ee0, c1=0xa06bd60, config=0xb781adcc, fo=0xb7819f18, rc=0xb7819f14) at channel.c:2334
#4 0x00b7d1ad in ast_bridge_call (chan=0xa074ee0, peer=0xa06bd60, config=0xb781adcc) at res_features.c:1422
#5 0x00d8f8c8 in dial_exec_full (chan=0xa074ee0, data=, peerflags=0xb781ae94, continue_exec=0x0) at app_dial.c:1693
#6 0x00d90052 in dial_exec (chan=0xa074ee0, data=0xb781cf08) at app_dial.c:1747
#7 0x080c9a6b in pbx_extension_helper (c=0xa074ee0, con=0x0, context=0xa075060 “default”, exten=0xa0750b0 “2000”, priority=1, label=0x0,
callerid=0xa074c88 “9999209”, action=E_SPAWN) at pbx.c:537
#8 0x080cc0e4 in __ast_pbx_run (c=0xa074ee0) at pbx.c:2317
#9 0x080cd17e in pbx_thread (data=0xa074ee0) at pbx.c:2634
#10 0x080fb58b in dummy_start (data=0xa074bf8) at utils.c:865
#11 0x00d302db in start_thread () from /lib/libpthread.so.0
#12 0x00c8a12e in clone () from /lib/libc.so.6

Hope that I was clear if someone detects a defect please guide me

Thank To all

Nacef