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Routing inbound/ outbound calls from Ip500v1 ATM to IPO Srv Ed 1

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OhashiManoel

IS-IT--Management
Nov 21, 2009
35
BR
Scenario:
IP500 v1, Preferred Ed, 10 SIP channels Lic, vcm32 (R 9.0)
Avaya IPO Server Edition (R 11.1.1 Subscription)

All I need is to have the Ip500v1 inbound/ outbound calls routed from one of its ATM ports to/ from IPO Srv Ed. Is it possible to connect them through SIP trunk?
 
yes that is possible

You have to make the SIP trunk work and then send the call via a shortcode to the Server Edition.

For outgoing calls you will send the calls dialed to the SIP trunk via shortcode/ARS

If you try and make calls from the SE to the IP500V1 via SIP trunk and then out the analog lines then that will also work but will add delay to the calls but again it will work.

Step1
Make the SIP trunk work for calls in both directions then the rest is just shortcode programming.

Joe
FHandw, ACSS, ACIS
 
Thank you, Westi!
I will work on it!
Oh. Both devices are located on the same network.
 
Does the sip trunk need be registered on both ends?
Because if I put them on register state, they go out of service.
When I untick registration required, Registration goes blank on Call Status, but lines go Idle on both ends.
 
I recall if you are going between 2 IPOs on the same network you don't need registration or credentials.
 
ipohead is correct
just add the IP address of the opposite IPO as target just like you would on a trunk that has no registration then add a URI and let the ARS do the extn@xxx.xx.xxx.xxx information and if it is not working start with that to adjust and make it work.
I have a Server 11.1 talk to my old SOE on 3.2 via SIP trunk. Not that I use either one for phone calls but just for fun.

These are my settings on the SIP trunk that worked for my system

ITSP domain and proxy is the IP address of the opposite system
URI is default with simply the incoming and outgoing line ID changed
Codec adjusted to g.711 as main then go down from there
Re-invite supported (not sure if it is needed but it works)
no credentials obviously
SIP advanced I started with defaults and works

Joe
FHandw, ACSS, ACIS
 
Thank you IPO head, thank you Westi.
How about ICR? Where should I point it to?
 
Hello all!
It worked. Now, I'm able to dial users from/ to both systems (IP500/ SE).
But I'm curious though. If I simply added the ARS rule. It caused errors, when dialing the last 4 digits from SE to Ip500 and vice versa. On both devices, the call always ended up catching the carrier SIP trunk ARS config.
So, I created short codes on both devices, delimitating the extension length.
On the ICR I added a (.) for destination and for fallback, empty. That procedure ignored ARS rules and ended up establishing the call initiation.
There only 2 things that are making me curious:
1) At the moment, both I/O calls are correctly identified on the SE (user name on the top, user extension number on the botton), but I cannot call back from the registered missed/ received calls.
2) On the IP500, however, only User name is displayed on the incoming Calls from the SE. If I check for received/ missed calls I get User name/ User name on the phone screen. For outgoing calls, I get "unknown" tag on the botton of the phone screen.

On the Ip500 I use one H323 9611 phone. I have just CRAFTed it and changed its SIG to SIP, but yet nothing changes.
On the SE, on both h323 9611 and on SIP J179 phones, I'm able to see User number/ name for I/O calls.
 
If your ARS goes to the carrier SIP trunk then your line ID was set wrong in the ARS and pointed to the wrong trunk
Otherwise it would just not have worked but never been gone to the regular trunk


Joe
FHandw, ACSS, ACIS
 
Do I need to create VLAN on the ip500 and on SE Subscription?
 
Here's a guide for R8.1.

There were some changes between then and R11.0 but I am not sure when this came about. Best to check which is closest to your system.
 
Tha k you, IPO head. I'm reading it just now!
That's very interesting!
 
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