here is the working configuration & tips for oh323

Thank you for this post. I have followed your instructions and literaly copied and pasted you ooh323.conf file accept for the "bindaddr=10.1.23.101" where i have put in my servers IP. Now during a test call from my xlite sip softphone to the h323 GW the pstn/Mobile phone rings but as soon as it is answered within 2 secs the line gets cut, This is the debug msg during the line being dropped => asterisk1*CLI> -- OOH323/216.206.188.216-417b answered SIP/1979-efbf
asterisk1*CLI> -- Attempting native bridge of SIP/1979-efbf and OOH323/216.206.188.216-417b
asterisk1*CLI> --- onCallCleared ooh323c_o_2
--- find_call
+++ find_call

But my xlite is still on call and asterisk also thinks the call is active. Please help.

this is what is executed during the dial
-- Executing Dial("SIP/1979-efbf", "OOH323/3338801713005569@216.206.188.216|120|r") in new stack

Thank you for the reply. I have checked the translations this is what it shows :-

g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc
g723 - - - - - - - - - - -
gsm - - 2 2 4 2 1 5 - - 15
ulaw - 4 - 1 4 2 1 5 - - 15
alaw - 4 1 - 4 2 1 5 - - 15
g726 - 6 4 4 - 4 3 7 - - 17
adpcm - 4 2 2 4 - 1 5 - - 15
slin - 3 1 1 3 1 - 4 - - 14
lpc10 - 5 3 3 5 3 2 - - - 16
g729 - - - - - - - - - - -
speex - - - - - - - - - - -
ilbc - 5 3 3 5 3 2 6 - - -

g723 and g729 are blank. What do i do? And which conf file do i edit to check the regions?

how did you decide it does not listen?
our system is working for months&months and it seems listening,i listed upper what i did...
just follow steps..ok?
i hope you manage it.

thats a same problem. 1 second after cut the line. how can fixed this problem?

ok but we need some logs,there could be many reasons from configuration to codec problems.

after "asterisk -r"
do "ooh323 debug" and send the logs...ok?
bys.

ok,first of all could you please try

"trixbox 1.2.3" and "rpm -i asterisk-addons-1.2.3-1.219.i386.rpm"

because I ALSO COULD NOT achieve to run ooh323 in a stable way on trixbox 2 as i mentioned before.

there seems some bugs in trixbox 2.0 in ooh323 wise.
try with those couple,ok?

Hi Ogulcan,
I'm wondering which is actually your network configuration. You talk about the Cisco Call Manager, so I suppose you have a network of H323 phone terminals connected to and managed by the Call Manager; at that point you attached the Trixbox to the Cisco Call Manager and therefore you add the custom Trunk and the custom route in the Trixbox (and you accept them on the Call Manager side).

I'm actually interested in something dfferent: I'd like to manage connections between different H323 terminals exclusively by using Trixbox (and a software gatekeeper, if needed, like gnugk).

I'm trying to follow your configuraion setup and adapt it to my situation, where I have a trixbox and several PC's connected on a LAN. Therefore I'm setting up a custom Trunk to each h323 softphone from the trixbox; every softphone already is listening, so nothing needs to be set up at the client side (is this sensible?)

I'm confused by the need of custom Trunks: in your scheme you need to connect to the Call Manager, but in my situation there's none ... so, is it sensible or superfluous to set up a trunk to each client?

Secondly: I add the custom trunk and route through freePBX; is it ok? or do you directly edit the .conf files? if so, what do you add and where?

So far I could only register h323 activity on asterisk after doing 'ooh323 debug', but I never got further than
" == Starting OOH323/Anneke Praagman-b82b at default,601,1 failed so falling back to exten 's' "
this while the custom route added through freePBX was

Dial rules:601
OOH323/$OUTNUM$@129.125.71.173:1720

By this I meant to have Trixbox to connect to my h323 client at 129.125.71.173 while dialing in from a softphone with the string h323:601@ip-of-the-trixbox1.2.3 ... does this make sense? :-?

I would appreciate any hint & tips.

Sincerly,
Fabio

Thanks a lot, I'll try to follow your hints.

I replyed to a post of yours here but also at another point, so this could have been confusing, sorry for that.

About all those points:

1) Your configuration.
Well, this doesn't look a small-size system ... my compliments for getting such a network with three Trixboxes, a SER and two CCM's ... :-P

2) Actually I'm not yet using gnuGK, but along my quest for h323 information along the web I got the impression that this is a rather important piece of a H323 network.
But then again, I'm not yet using it; actually it is interesting for my project to have TrixBox to manage directly the h323 (soft)phones; would this be practical?

by the way, you seem to make difference among softphones and ipphones; why is that? As far as I can understand, you could see a "softphone" (a software phone, right?) as an ipphone (a standalone phone terminal, right?), but then emulated by a computer.

3)Trunks
Ok, in the case a gatekeeper is present in the network it is reasonable to make a trunk to it from the TrixBox and to have all the H323 terminals to conect to the GateKeeper.

My errors/problems take place in another situation, where two different softphones directly connect to The TrixBox; when one is calling, I'd like to have TrixBox to route the connection to the other softphone, without an additional gatekeeper (after all, it is optional, right?). That's why I st up two Trunks: to enable connections to the different softphones.

4) Routing
Ok, now it's clear, the Trunk and the Route should be set up through freePBX.
I realized that the configuration files are hard to manage; I also started a little reverse-engineering about the changes freePBX makes in those files ... I started digging into those ugly conf files because of the (incomplete) information around teh fora and the information within ooh323.conf. Of course, if it's not necessary' I'll avoid these time-consuming things.
I'll check the contexts, this seems reasonable.

5)my problem
Well, as I already said, I'm not using a software gatekeeper yet, so the problems do raise when a softphone connects directly to a trixbox by h323 and wants to connect to another h323 softphone. Does this make my problem more clear?

6)Dialing
ok, I agree in the case of a gatekeeper: a gk cam resolve "601" and set up a communication. But can Trixbox do anything similar without a GK?

Sorry for repeating parts of my questions...

I hope you will be able to help me a little bit more! :lol:
Thanks,
/Fabio

2) Well, I could leave without GnuGK as well, the problem here is that I'm looking for the possibilities to implement a Hybrid SIP/H323 network based on Asterisk; it is not my purpose to "outsource" H323 networking to some hardware. Any free software solution would do.

3)hmmm ... how do you register a h323 phone, say with extension '101', to the trixbox? this is still vague to me...

1.in ooh323.conf check;

h245tunneling=no
faststart=no

2.in sip.conf under the line "tos=0x68" check;

notransfer=yes
canreinvite=no

3.Be sure using asterisk-addons-1.2.3-1.219.i386.rpm

4.Be sure both your sip extension and h323 extension are in the same context.

5.Be sure they have both same codec enabled.

6.Be sure to be sure of all of those above.(joke:)

hope it helps.

Hi,

1) I had h245tunneling=yes, so I changet it; the line faststart=no was already there.

2) I addedto sip.conf both lines
notransfer=yes
canreinvite=no

3) I used the asterisk-addons-1.2.3-1.219.i386.rpm which yopu put on your website

4) I' putting the incoming call on h323 in context=ext-local in the ooh323.conf; this is because the .conf files generated by freePBX put the SIP extensions (501 and 502) within this context. See this snippet of extensions_additional.conf:

[ext-local]
include => ext-local-custom
exten => 501,1,Macro(exten-vm,novm,501)
exten => 501,hint,SIP/501
exten => 502,1,Macro(exten-vm,novm,502)
exten => 502,hint,SIP/502
exten => 601,1,Macro(exten-vm,novm,601)
exten => 601,hint,OOH323/601
exten => 602,1,Macro(exten-vm,novm,602)
exten => 602,hint,OOH323/602

here 601 and 602 are our custom extensions.
Could it be that the declaration of the 601 and 602 extensions interfere with the calls coming through the ooh323 channel? Otherwise I don't see what's wrong with this.
It could be that the ooh323 channel puts the incominc calls in another context, but this means ignoring the context=ext-local parameter, right?

5) Now I use Xlite and Ekiga; when setting up Ekiga with a SIP account it is possible to call and speak to the Xlite softphone, so there should be common codecs.
When switching Ekiga to H323, we manage to let ring a h323 softphone and to have Ekiga ring for a incoming connection from a SIP softphone; whenever somebody wants to accept the connection, the connection is dropped. I'll attach the combined debug output of ooh323c and sip.

6)I'm sure everything is unsure right now ... :-?

Thanx,
/Fabio

--- onNewCallCreated ooh323c_1
+++ onNewCallCreated ooh323c_1
--- ooh323_onReceivedSetup ooh323c_1
--- find_user
+++ find_user
Adding capabilities to call(incoming, ooh323c_1)
--- configure_local_rtp
+++ configure_local_rtp
+++ ooh323_onReceivedSetup - Determined context ext-local, extension 501
--- onAlerting ooh323c_1
--- find_call
+++ find_call
+++ onAlerting ooh323c_1
-- Executing Macro("OOH323/Fabio Bracci-582b", "exten-vm|novm|501") in new stack
-- Executing Macro("OOH323/Fabio Bracci-582b", "user-callerid") in new stack
-- Executing GotoIf("OOH323/Fabio Bracci-582b", "0?report") in new stack
-- Executing GotoIf("OOH323/Fabio Bracci-582b", "0?start") in new stack
-- Executing Set("OOH323/Fabio Bracci-582b", "REALCALLERIDNUM=602") in new stack
-- Executing NoOp("OOH323/Fabio Bracci-582b", "REALCALLERIDNUM is 602") in new stack
-- Executing Set("OOH323/Fabio Bracci-582b", "AMPUSER=602") in new stack
-- Executing Set("OOH323/Fabio Bracci-582b", "AMPUSERCIDNAME=Fabio h323 freePBX") in new stack
-- Executing GotoIf("OOH323/Fabio Bracci-582b", "0?report") in new stack
-- Executing Set("OOH323/Fabio Bracci-582b", "CALLERID(all)=Fabio h323 freePBX <602>") in new stack
-- Executing NoOp("OOH323/Fabio Bracci-582b", "Using CallerID "Fabio h323 freePBX" <602>") in new stack
-- Executing Set("OOH323/Fabio Bracci-582b", "FROMCONTEXT=exten-vm") in new stack
-- Executing Set("OOH323/Fabio Bracci-582b", "VMBOX=novm") in new stack
-- Executing Set("OOH323/Fabio Bracci-582b", "EXTTOCALL=501") in new stack
-- Executing Set("OOH323/Fabio Bracci-582b", "CFUEXT=") in new stack
-- Executing Set("OOH323/Fabio Bracci-582b", "RT=") in new stack
-- Executing Macro("OOH323/Fabio Bracci-582b", "record-enable|501|IN") in new stack
-- Executing GotoIf("OOH323/Fabio Bracci-582b", "0 > 0?2:4") in new stack
-- Goto (macro-record-enable,s,4)
-- Executing AGI("OOH323/Fabio Bracci-582b", "recordingcheck|20070301-162645|1172762805.8") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck
PHP Warning: Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file: No such file or directory in Unknown on line 0
recordingcheck|20070301-162645|1172762805.8: Inbound recording not enabled
-- AGI Script recordingcheck completed, returning 0
-- Executing NoOp("OOH323/Fabio Bracci-582b", "No recording needed") in new stack
-- Executing GotoIf("OOH323/Fabio Bracci-582b", "0?dolocaldial|1") in new stack
-- Executing Macro("OOH323/Fabio Bracci-582b", "dial||tr|501") in new stack
-- Executing AGI("OOH323/Fabio Bracci-582b", "dialparties.agi") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/dialparties.agi
PHP Warning: Unknown(): Unable to load dynamic library '/usr/lib/php4/imap.so' - libc-client.so.0: cannot open shared object file: No such file or directory in Unknown on line 0
dialparties.agi: Starting New Dialparties.agi
-- dialparties.agi: priority is 1
dialparties.agi: Caller ID name is 'Fabio h323 freePBX' number is '602'
dialparties.agi: Methodology of ring is 'none'
-- dialparties.agi: Added extension 501 to extension map
-- dialparties.agi: Extension 501 cf is disabled
-- dialparties.agi: Extension 501 do not disturb is disabled
== Parsing '/etc/asterisk/manager.conf': Found
== Parsing '/etc/asterisk/manager_custom.conf': Found
== Manager 'admin' logged on from 127.0.0.1
== Manager 'admin' logged off from 127.0.0.1

<-- SIP read from 129.125.71.172:5060:

--- (0 headers 0 lines) Nat keepalive ---
-- dialparties.agi: Checking CW and CFB status for extension 501
-- dialparties.agi: DbSet CALLTRACE/501 to 602
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial("OOH323/Fabio Bracci-582b", "SIP/501||tr") in new stack
We're at 129.125.21.241 port 17704
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (no NAT) to 129.125.71.172:5060:
INVITE sip:501@129.125.71.172:5060 SIP/2.0
Via: SIP/2.0/UDP 129.125.21.241:5060;branch=z9hG4bK061f6d81
From: "Fabio h323 freePBX" ;tag=as0a17cbbe
To:
Contact:
Call-ID: 3bb3e2b827b1134e683cf76b18789f8c@129.125.21.241
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 01 Mar 2007 15:26:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 244

v=0
o=root 11555 11555 IN IP4 129.125.21.241
s=session
c=IN IP4 129.125.21.241
t=0 0
m=audio 17704 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 501
----- ooh323_indicate 3 on call ooh323c_1
++++ ooh323_indicate 3 on ooh323c_1

<-- SIP read from 129.125.71.172:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 129.125.21.241:5060;branch=z9hG4bK061f6d81
From: "Fabio h323 freePBX" ;tag=as0a17cbbe
To: ;tag=1851193428
Contact:
Call-ID: 3bb3e2b827b1134e683cf76b18789f8c@129.125.21.241
CSeq: 102 INVITE
Server: X-Lite release 1105d
Content-Length: 0

--- (9 headers 0 lines)---

<-- SIP read from 129.125.71.172:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 129.125.21.241:5060;branch=z9hG4bK061f6d81
From: "Fabio h323 freePBX" ;tag=as0a17cbbe
To: ;tag=1851193428
Contact:
Call-ID: 3bb3e2b827b1134e683cf76b18789f8c@129.125.21.241
CSeq: 102 INVITE
Server: X-Lite release 1105d
Content-Length: 0

--- (9 headers 0 lines)---
-- SIP/501-09b4a380 is ringing

<-- SIP read from 129.125.71.172:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 129.125.21.241:5060;branch=z9hG4bK061f6d81
From: "Fabio h323 freePBX" ;tag=as0a17cbbe
To: ;tag=1851193428
Contact:
Call-ID: 3bb3e2b827b1134e683cf76b18789f8c@129.125.21.241
CSeq: 102 INVITE
Content-Type: application/sdp
Server: X-Lite release 1105d
Content-Length: 310

v=0
o=501 234564338 234567788 IN IP4 129.125.71.172
s=X-Lite
c=IN IP4 129.125.71.172
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

--- (10 headers 14 lines)---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 129.125.71.172:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop:
set_destination: Parsing for address/port to send to
set_destination: set destination to 129.125.71.172, port 5060
Transmitting (no NAT) to 129.125.71.172:5060:
ACK sip:501@129.125.71.172:5060 SIP/2.0
Via: SIP/2.0/UDP 129.125.21.241:5060;branch=z9hG4bK65a49af2
From: "Fabio h323 freePBX" ;tag=as0a17cbbe
To: ;tag=1851193428
Contact:
Call-ID: 3bb3e2b827b1134e683cf76b18789f8c@129.125.21.241
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
-- SIP/501-09b4a380 answered OOH323/Fabio Bracci-582b
----- ooh323_indicate -1 on call ooh323c_1
++++ ooh323_indicate -1 on ooh323c_1
--- ooh323_answer
+++ ooh323_answer
--- onCallEstablished ooh323c_1
--- find_call
+++ find_call
+++ onCallEstablished ooh323c_1
--- onCallCleared ooh323c_1
--- find_call
+++ find_call
Scheduling destruction of call '3bb3e2b827b1134e683cf76b18789f8c@129.125.21.241' in 32000 ms
set_destination: Parsing for address/port to send to
set_destination: set destination to 129.125.71.172, port 5060
Reliably Transmitting (no NAT) to 129.125.71.172:5060:
BYE sip:501@129.125.71.172:5060 SIP/2.0
Via: SIP/2.0/UDP 129.125.21.241:5060;branch=z9hG4bK6bc559da
From: "Fabio h323 freePBX" ;tag=as0a17cbbe
To: ;tag=1851193428
Contact:
Call-ID: 3bb3e2b827b1134e683cf76b18789f8c@129.125.21.241
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

---
== Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Fabio Bracci-582b' in macro 'dial'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Fabio Bracci-582b' in macro 'exten-vm'
== Spawn extension (macro-dial, s, 10) exited non-zero on 'OOH323/Fabio Bracci-582b'

<-- SIP read from 129.125.71.172:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 129.125.21.241:5060;branch=z9hG4bK6bc559da
From: "Fabio h323 freePBX" ;tag=as0a17cbbe
To: ;tag=1851193428
Contact:
Call-ID: 3bb3e2b827b1134e683cf76b18789f8c@129.125.21.241
CSeq: 103 BYE
Server: X-Lite release 1105d
Content-Length: 0

--- (9 headers 0 lines)---
Destroying call '3bb3e2b827b1134e683cf76b18789f8c@129.125.21.241'
--- ooh323_hangup
hanging Fabio Bracci
+++ ooh323_hangup
--- ooh323_destroy
Destroying Fabio Bracci
+++ ooh323_destroy

I don't know either why Asterisk says "Scheduling destruction of call '3bb3e2b827b1134e683cf76b18789f8c@129.125.21.241' in 32000 ms" ...

1) I did it, but it doesn't help. But why anyhow? extension_addional.conf gets included in extensions.conf, so the same definitions are fed into Asterisk twice, right?

2)I discovered that Ekiga and Xlite and Gnomemeeting have a codec in common, PCMU aka g711u, so I use everywhere only that, but it didn't help.

3)I don't get how to follow your configs ... In asterisk I have two conventional SIP extensions (501,502) generated through freePBX.
Then I made also two custom extensions (601/602) with custom dial string (OOH323/601, OOH323/602); as I mentioned before everything boils down to

[ext-local]
include => ext-local-custom
exten => 501,1,Macro(exten-vm,novm,501)
exten => 501,hint,SIP/501
exten => 502,1,Macro(exten-vm,novm,502)
exten => 502,hint,SIP/502
exten => 601,1,Macro(exten-vm,novm,601)
exten => 601,hint,OOH323/601
exten => 602,1,Macro(exten-vm,novm,602)
exten => 602,hint,OOH323/602

which gets included in extensions.conf which gets included somewhere else ...

but I notice that in extensions.conf you do something different:
exten => _60xx,1,Macro(dialout-trunk,2,${EXTEN},,)
What is this, a kind of route?
Sincerly,
/Fabio

I already configure a trunk between an Avaya and Asterisk. Some curiosities, I can't configure a trunk via FreePBX, I had to do it manually.
Second when I make a call Avaya => Asterisk I have good audio (Both sides), but when I beggin the call from Asterisk => Avaya I have one way audio, I can here people from Avaya but they can't here me.
When I went to the h323_log y see this:

14:33:39:313 ERROR: Logical Channel 1004 not found, fast start. (outgoing, ooh323c_o_2)
14:33:43:304 H.225 Connect message received (outgoing, ooh323c_o_2)
14:33:43:304 ERROR: Logical Channel 1004 not found, fasts start answered. (outgoing, ooh323c_o_2)

Also in my ooh323.conf I put the next line

faststart=yes

But nothing yet. I keep having the one way audio.

Anyone know something THANKS

problem solved itself. thanks