SIP Line showing extension as caller ID

[DQuaN]

Hi all,

We've all seen this before. The answer is simple, make sure the SIP URI is set to use internal data and then put the details in the SIP tab for the user. I have, it's not working. I've tried putting the number in the URI in the line, it still shows the extension. I've contacted our Avaya supplier and sent them a trace and config, they cannot figure out why the system is sending the extension instead of the ddi. They say they may have to raise it with Avaya directly.

A snippet of the trace is below. This is of the customer calling me.

12:46:40 2152474mS SIP Call Tx: 17
INVITE sip:0208232XXXX@sip.node4.co.uk SIP/2.0
Via: SIP/2.0/UDP 185.41.26.66:5060;rport;branch=z9hG4bK062b84aa54c9c44d98bcfc08c8ee187b
From: "Gina FXXXX" <sip:201@sip.node4.co.uk>;tag=fe9fc1846cd4763d
To: <sip:0208232XXXX@sip.node4.co.uk>

You can see that the From field is showing sip:201@sip.node4.co.uk. It should be showing sip:linenumber@sip.node4.co.uk

The caller ID I get when she calls me is 0201. It always puts a 0 in front of the extension.

If I tick "anonymous" in the SIP tab, the number is withheld.

Any thoughts?

 

[DQuaN]

Anyone?

 

[hairlessupportmonkey]

What does your ARS look like?

ACSS - SME
General Geek

 

[DQuaN]

Standard dial shortcode. 0N; and 0N"@sipprovider.com

 

[hairlessupportmonkey]

So, if you create a dial short code to present a specific DDI does that work correctly?

or amend your code to be 0Ns0208232xxxx

later versions of IPO you can drop the "@sipprovider.com" or route via a specific URI

ACSS - SME
General Geek

 

[DQuaN]

I've tried doing .s0208xxxxx Didn't work.

I'm not familiar with the 0Ns0208232... I could give that a try.

Yeah I found that out recently. I added it back in for this build as I cannot think of anything else to change.

 

[IPGuru]

What version of IPO firmware are you running

1) the @'sipprovider' suffix in the telephone number field has not been required for some time
2) The cli settings should be set in the users SIP TAB, if not then it is possible a bug
3) find a better SIP provider, they should not be accepting any old cli & passing it on, they should only accept numbers valid on the circuit.


A Maintenance contract is essential, not a Luxury.
Do things on the cheap & it will cost you dear

 

[DQuaN]

1, Interesting. I wasn't aware of that.
2, It is set in the SIP tab but makes no difference. The provider have said that they only accept P asserted ID. I've enabled PAI but it made no difference.
3, I agree. It was the customers decision.

 

[hairlessupportmonkey]

3) I dont think this area is regulated. In fact, its no longer illegal in the UK to forward on a callers CLI information out to a forwarded or twinned handset.

Anyway, perhaps you can pay for someone on here from the UK to take a look for you. (I'm in the UK )

ACSS - SME
General Geek

 

[DQuaN]

I've logged a case direct with Node4. One of the issues was having is that our customer purchased it through a reseller who get it from a reseller who resell from Node4. Getting support is a nightmare!

 

[DQuaN]

Welp. In the end I deleted and recreated the SIP line. Now it works. I'm sure the config was exactly the same but perhaps there was a typo that I was just not seeing.