Asterisk H323 trunk to ACM 3.1.2 1

[henningwentzel]

I am trying to establish a h.323 trunk between my Asterisk Trixbox 2.2 and an Avaya CM 3.1.2. I then realized I do not have the 00h323 addons installed. I have downloaded the addon package however when running the ./configure command I get the following error:
configure: error: No asterisk installation found
When I do a show channels or channeltypes i see no information associated to H.323 at all. Can the asterisk gurus please point the old telephone man in the right direction?

thread690-1163320

 

[henningwentzel]

out of the box trixbox already has the addons installed. after logging in:

cp /etc/asterisk-1.2.7.1_samples/ooh323.conf /etc/asterisk
amportal stop
amportal start

now to set up a SIP trunk and a H.323 trunk to S8710 cm 3.1.2.

If anyone has successfully achieved this please can you share your configurations?

Thanks

 

[flavigne]

Hi,
I suggest you to use YATE (running as service on Win32 platforms)
It's easy, powerfull, and i use it as sip gateway.
I'm able to call avaya ip phone from my sip soft phone from everywhere.

If you want, i can send you my avaya configuration (sig, trung, network region) and my Yate configuration

 

[henningwentzel]

Hi Flavigne,

Thankyou very much, how and where do I get YATE and add as a service to run on the Win32 platform?

If you can send me your Avaya configurations I will appreciate. Thanks in advance.

Henningwentzel

 

[flavigne]

yate-1.2.0-setup+gtk2.exe :

http://yate.null.ro/pmwiki/index.php?n=Main.Download

1) IP NODE NAMES :
- Yate ip_of_your_yate_server

2) SIGNALING GROUP :
- goup type h.323
- Supplementary Service Protocol: a

- Near-end Node Name: clan-xxxx
- Near-end Listen Port: 1720

- Far-end Node Name: Yate
- Far-end Listen Port: 1720
- Far-end Network Region: 31 (for exemple)

other values set to n or 0

3) IP NETWORK REGION :
- create IP NETWORK REGION 31 (for exemple) with default parameters and the ability to communicate with your system and your phone IP NETWORK REGION
- add the possibility to your system and phone ipnetwork region the ability to communicate with the 31 (for exemple)

4) TRUNK :
- create a new trunk
- Service Type: tie
- Two-way
- carrier Medium: IP

- in group member, Port : IP / Sig Grp : 83

5) ROUTE :
add new route using trunk 84, frl 0 and bothunr

6) TOLL :
Give your system the ability to call extension 8xxxxx

8) PARTITION GROUP NUMBER :
choose an empty route index and for all PGN assign the route you create in 5)

7) ARS :
-Change ARS ANA 8 and min:6 / max:6 / route Pattern(index of route in 8)) / Call type pubu

That's all for Avaya.

For Yate :
1) H323Chan.conf :
[general]
external_rtp=yes
passtrough_rtp=yes
needmedia=no

maxcleaning=100

[codecs]
default=no
alaw=yes
g711=on


[ep]
faststart=on
h245tunneling=on

2) regexroute.conf (you must use your clan ip address):

[default]
^0\(.*\)$=h323/\1@192.168.112.200:1720

3) regfile.conf (2 sip account)
[800000]
password=800000

[800001]
password=800001

3) yrtpchan.conf:
[general]
; This section sets global network level variables

; minport: int: Minimum port range to allocate
minport=5060

; maxport: int: Maximum port range to allocate
maxport=5070

; tos: keyword: Type Of Service to set in outgoing UDP packets
; numeric TOS value or: lowdelay, throughput, reliability, mincost
tos=0

; buffer: int: Maximum buffer size - used to fragment octet (G.711) audio streams
buffer=240

; autoaddr: bool: Auto change outgoing RTP addressort to match incoming
;autoaddr=enable

; rtcp: bool: Allocate socket for the RTCP protocol by default
;rtcp=disable

; drillhole: bool: Attempt to drill a hole through a firewall or NAT
;drillhole=enable

; defsleep: int: Default in-loop sleep time for new RTP sessions in milliseconds
defsleep=5

; minsleep: int: Minimum allowed in-loop sleep time in milliseconds
minsleep=1

4) ysipchan.conf:
[general]
; This section sets global variables of the implementation

; port: int: SIP UDP port
port=5060

; addr: ipaddress: IP address to bind to
addr=192.168.12.19 (address of your yate server)

; useragent: string: String to set in User-Agent or Server headers
;useragent=YATE/1.1.0

; realm: string: Authentication realm to offer in authentication requests
realm=Nom de ma societe

; transfer: bool: Allow handling the REFER message to perform transfers
transfer=enable

; registrar: bool: Allow the SIP module to receive registration requests
registrar=enable

; options: bool: Build and send a default 200 answer to OPTIONS requests
options=enable

; prack: bool: Enable acknowledging provisional 1xx answers (RFC 3262)
prack=disable

; info: bool: Accept incoming INFO messages
;info=enable

; fork: bool: Follow first forked 2xx answer on early dialogs
;fork=enable

; progress: bool: Send an "183 Session Progress" just after successfull routing
progress=disable

; generate: bool: Allow Yate messages to send arbitrary SIP client transactions
generate=disable

; nat: bool: Enable automatic NAT support
nat=enable

; ignorevia: bool: Ignore Via headers and send answer back to the source
ignorevia=enable

; dtmfinband: bool: Generate DTMF inband by default
dtmfinband=no

; dtmfinfo: bool: Generate INFO messages to send keypad tones
dtmfinfo=no

; privacy: bool: Process and generate privacy related SIP headers
privacy=disable

; forward_sdp: bool: Include the raw SDP body to be used as-is for forwarding RTP
forward_sdp=disable


[registrar]
; Controls the behaviour when acting as registrar

; expires_min: int: Minimum allowed expiration time in seconds
expires_min=60

; expires_def: int: Default expiration time if not present in REGISTER request
expires_def=600

; expires_max: int: Value used to limit the expiration time to something sane
expires_max=3600

; auth_required: bool: Automatically challenge all clients for authentication
auth_required=enable

[codecs]
default=off
;mulaw=yes
alaw=yes
g711=yes
;g729=yes

--------------------------------------------
And now, you can use the sip phone Yate to call an extension of your avaya system an you can call 800000 or 800001 from your avaya phone !