IPOffice 500 via SIP Trunk to external Switch 1

[NeutronIC]

Hi!

I'm trying to integrate an Avaya IPOffice 500 via SIP Trunking to another switch. For simplicities sake, let's say the other switch is Asterisk (if we can make that work then it will work with our product).

I've found this:

http://www.tek-tips.com/faqs.cfm?fid=7248

This seems to me to imply that it is possible. I can't find an originator for the article above to find out more details about what was actually possible with the integration unfortunately.

Has anyone any experience with this?

The call flow scenarios that I am looking to replicate are:

1. Inbound call from PSTN goes via Q.931 to IPOffice. It matches a certain DDI via Incoming Call Route and should be delivered via SIP Trunk to remote switch for it to accept/reject etc.

2. Inbound call from Asterisk via SIP Trunk to IPOffice. Match via Short Codes such that (for example) 9N goes out via the PSTN Q.931 link and anything else is routed internally such as to other VM services on the IPO or to extensions.

3. Call from IPO Extension matching certain short code (e.g. *8N) should be delivered to the Asterisk switch.

Does the tek-tip FAQ article describe a configuration that can effect this kind of behaviour?

Many thanks in advance

Regards
Matt Peddlesden
Macfarlane Telesystems Ltd.

 

[hairlessupportmonkey]

why dont you just make it an H323 trunk? you will need SIP trunk licences on the IPO where as an H323 trunk you dont, and just use shortcodes to dial over it.

 

[NeutronIC]

Hi,

Thanks for the reply.

The other switch is actually not Asterisk in reality, it's another system that we produce here that doesn't talk H323 (only SIP). The IP Office switch in question already has the necessary SIP licenses.

Thanks

Matt.

 

[mattKnight]

>where as an H323 trunk you dont

are you sure about that?

Take Care

Matt
I have always wished that my computer would be as easy to use as my telephone.
My wish has come true. I no longer know how to use my telephone.

 

[Bas1234]

For incomming calls try to use a * on Local URI/Contact/Display, this will match anything from the Incomming Call Route. In the ICR just enter the dialed "incomming number" (without IP addr) and its destination.

for the ShortCode;

Create a SC;
Code; *8N
Feat; Dial
TelN; *8N
LGid; 5X (a new ARS)

Create a new ARS

Code; *8N
Feat; Dial
TelN; *8N"@sip.voipax.com" (with or without *8 depends on how you softpbx want's it)
LGid; X (Sip outgoing line id, don't use the same group id on an Alog/ISDN line)

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged
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[NeutronIC]

Thanks Bas,

I presume in my case for the ARS telN I use "*8N"@a.b.c.d" and refer it to the IP address of the other SIP peer?

I haven't looked at using ARS for this so I'll take a look at that.

Do you know if this configuration is one that is "supported" by Avaya (i.e. SIP to another SIP peer that isn't an ITSP) or just simply that it works?

Presumably it'll cover the cases that I mentioned in the initial post and short codes are used on calls coming in from the SIP trunks?

Many thanks indeed, very useful!

Regards
Matt Peddlesden

 

[Bas1234]

They have a list of supported SIP trunk providers, but I have a lot more working then the ones from the list.

The most are dutch providers, and some are German/US based.

If your Softpbx supports the most common SIP trunk codes then mostly you'll get it working.

Sip Extensions is different thing, transferring a call is the hardest. A Nokia E-series gets rejected but a Siemens Gigaset IP or a Linksys ATA doesn't.

___________________________________________
It works! Now if only I could remember what I did...

Dain Bramaged
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