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
gatekeeper = DISABLE
disallow=all ;Note order of disallow/allow is important.
faxdetect = cng
ip=10.100.10.17 ; UPDATE with appropriate ip address
port=1720 ; UPDATE with appropriate port
The SIP Trunk in the Asterisk system has the following configuration
******@sipgw0401:/var/log/asterisk# vim /etc/asterisk/sip.conf
port = 5060
bindaddr = 0.0.0.0
context=from-sip-external ; Default context for incoming calls. Defaults to ‘default’
Thank you very much, greetings