Transfer user to user data from ooh323 to SIP

Asterisk Asterisk Support
1

Hello

I’m a VoIP technician, and I use Avaya mostly, but we’ve been introducing Asterisk systems for a while now.

We have an Altitude IVR, to which calls arrive from an Avaya R4 device, this switchboard transfers the call to an Avaya R6 switchboard through H323, and this connects with an Altitude IVR system that we have, through SIP.

The CRM application scripts transfer data to the IVR campaign through the external data, which we believe travels in the User-To-User field of the SIP frame.

We need to eliminate the Avaya R6 switchboard, and we have tried to put an Asterisk in the middle to make the H323 to SIP conversion, necessary for this communication, since the Avaya R4 does not speak SIP, but we did not get it to work correctly, since we did not get it to travel this data

I have seen that the information travels in the SIP User-To-User frame, if I am not mistaken (User-to-User: 04363436313033;encoding=hex ).

With Asterisk we cannot see this information, even though the Avaya R4 sends it in the same way, and the Trunk has exactly the same settings.

If someone could please help me to see how I could do it, I would greatly appreciate it, since I can’t find the question.

My H323 Trunk in Asterisk system has the following configuration.

******@sipgw0401:/var/log/asterisk# vim /etc/asterisk/ooh323.conf
[general]
bindaddr=0.0.0.0
h323id=ObjSysAsterisk
e164=100
callerid=asterisk
aniasdni=no
gatekeeper = DISABLE
context=default
disallow=all ;Note order of disallow/allow is important.
allow=gsm
allow=ulaw
allow=alaw
dtmfmode=rfc2833
faxdetect = cng
directmedia=no
directrtpsetup=no

[AVAYA]
type=friend
context=default
ip=10.100.10.17 ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
disallow=all
;allow=g729
allow=alaw
allow=ulaw
e164=12345
rtptimeout=3600
dtmfmode=rfc2833
h245tunneling=yes
outgoinglimit=300
canreinvite=no

The SIP Trunk in the Asterisk system has the following configuration

******@sipgw0401:/var/log/asterisk# vim /etc/asterisk/sip.conf

[general]

port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
allow=alaw
context=from-sip-external ; Default context for incoming calls. Defaults to ‘default’
;callerid=unknown
recordall=no
alwaysauthreject=yes

localnet=10.100.10.0/24
localnet=10.103.2.0/24

[TR_IVR0101]
type=peer
host=10.103.2.31
port=5060
fromdomain=10.100.10.218
context=from-IVR0101
dtmfmode=rfc2833
canreinvite=yes
nat=never
disallow=all
allow=ulaw
allow=alaw
rtpkeepalive=1
insecure=port,invite
qualify=yes

Thank you very much, greetings

2

User-to-User only seems to have a draft RFC.

Asterisk is a back to back user agent, so is the user!

chan_sip is no longer supported, although I doubt that chan_pjsip has any knowledge of the draft RFC, either.

You would have to use the relevant SIP header manipulation mechanisms to accss and set this information.

3

Thank yo David, I am trying to find where can I read this information from the H323 call(the call arrives with the ooh323)

I don’t know how to read this data, but if this can be done with my AvayaR6, it is because the data travels with the AvayaR4 call in H323.

4

Hi again

I checked the H323 package and the information arrives to the Asterisk, but I can’t see it in the SIP package after conversion

image

The UUI data is the number 646994.

Is there any way to capture this data.

Thank you.

5

I don’t understand what you mean by conversion. Asterisk is a a back to back user agent, not a gateway.

Also I’m pretty sure that H.323 is even less well supported than chan_sip.

6

Thanks David

I am trying to make asterisk works as a gateway, because is what I know

Do you know any free h323 gateway?

7

Yate is probably an option, but I have no personal experience with it myself:
http://docs.yate.ro/wiki/Yate_as_H323_GateKeeper_and_YateClient_as_H323_client

8

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